Repo created

TMessagesProj/jni/voip/webrtc/modules/audio_device/OWNERS

@@ -0,0 +1,2 @@
+henrika@webrtc.org
+tkchin@webrtc.org

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_player.cc

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+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_player.h"
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AAudioPlayer::AAudioPlayer(AudioManager* audio_manager)
+    : main_thread_(TaskQueueBase::Current()),
+      aaudio_(audio_manager, AAUDIO_DIRECTION_OUTPUT, this) {
+  RTC_LOG(LS_INFO) << "ctor";
+  thread_checker_aaudio_.Detach();
+}
+
+AAudioPlayer::~AAudioPlayer() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  Terminate();
+  RTC_LOG(LS_INFO) << "#detected underruns: " << underrun_count_;
+}
+
+int AAudioPlayer::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (aaudio_.audio_parameters().channels() == 2) {
+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
+  }
+  return 0;
+}
+
+int AAudioPlayer::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  StopPlayout();
+  return 0;
+}
+
+int AAudioPlayer::InitPlayout() {
+  RTC_LOG(LS_INFO) << "InitPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!aaudio_.Init()) {
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+bool AAudioPlayer::PlayoutIsInitialized() const {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  return initialized_;
+}
+
+int AAudioPlayer::StartPlayout() {
+  RTC_LOG(LS_INFO) << "StartPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!playing_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Playout can not start since InitPlayout must succeed first";
+    return 0;
+  }
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  if (!aaudio_.Start()) {
+    return -1;
+  }
+  underrun_count_ = aaudio_.xrun_count();
+  first_data_callback_ = true;
+  playing_ = true;
+  return 0;
+}
+
+int AAudioPlayer::StopPlayout() {
+  RTC_LOG(LS_INFO) << "StopPlayout";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  if (!aaudio_.Stop()) {
+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
+    return -1;
+  }
+  thread_checker_aaudio_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  return 0;
+}
+
+bool AAudioPlayer::Playing() const {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  return playing_;
+}
+
+void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_DLOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  audio_device_buffer_ = audioBuffer;
+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
+  audio_device_buffer_->SetPlayoutSampleRate(audio_parameters.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(audio_parameters.channels());
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the optimal buffer
+  // size per callback used by AAudio.
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+}
+
+int AAudioPlayer::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
+  // TODO(henrika): investigate if we can use a thread checker here. Initial
+  // tests shows that this callback can sometimes be called on a unique thread
+  // but according to the documentation it should be on the same thread as the
+  // data callback.
+  // RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+    // The stream is disconnected and any attempt to use it will return
+    // AAUDIO_ERROR_DISCONNECTED.
+    RTC_LOG(LS_WARNING) << "Output stream disconnected";
+    // AAudio documentation states: "You should not close or reopen the stream
+    // from the callback, use another thread instead". A message is therefore
+    // sent to the main thread to do the restart operation.
+    RTC_DCHECK(main_thread_);
+    main_thread_->PostTask([this] { HandleStreamDisconnected(); });
+  }
+}
+
+aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
+                                                           int32_t num_frames) {
+  RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
+  // Log device id in first data callback to ensure that a valid device is
+  // utilized.
+  if (first_data_callback_) {
+    RTC_LOG(LS_INFO) << "--- First output data callback: "
+                        "device id="
+                     << aaudio_.device_id();
+    first_data_callback_ = false;
+  }
+
+  // Check if the underrun count has increased. If it has, increase the buffer
+  // size by adding the size of a burst. It will reduce the risk of underruns
+  // at the expense of an increased latency.
+  // TODO(henrika): enable possibility to disable and/or tune the algorithm.
+  const int32_t underrun_count = aaudio_.xrun_count();
+  if (underrun_count > underrun_count_) {
+    RTC_LOG(LS_ERROR) << "Underrun detected: " << underrun_count;
+    underrun_count_ = underrun_count;
+    aaudio_.IncreaseOutputBufferSize();
+  }
+
+  // Estimate latency between writing an audio frame to the output stream and
+  // the time that same frame is played out on the output audio device.
+  latency_millis_ = aaudio_.EstimateLatencyMillis();
+  // TODO(henrika): use for development only.
+  if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
+    RTC_DLOG(LS_INFO) << "output latency: " << latency_millis_
+                      << ", num_frames: " << num_frames;
+  }
+
+  // Read audio data from the WebRTC source using the FineAudioBuffer object
+  // and write that data into `audio_data` to be played out by AAudio.
+  // Prime output with zeros during a short initial phase to avoid distortion.
+  // TODO(henrika): do more work to figure out of if the initial forced silence
+  // period is really needed.
+  if (aaudio_.frames_written() < 50 * aaudio_.frames_per_burst()) {
+    const size_t num_bytes =
+        sizeof(int16_t) * aaudio_.samples_per_frame() * num_frames;
+    memset(audio_data, 0, num_bytes);
+  } else {
+    fine_audio_buffer_->GetPlayoutData(
+        rtc::MakeArrayView(static_cast<int16_t*>(audio_data),
+                           aaudio_.samples_per_frame() * num_frames),
+        static_cast<int>(latency_millis_ + 0.5));
+  }
+
+  // TODO(henrika): possibly add trace here to be included in systrace.
+  // See https://developer.android.com/studio/profile/systrace-commandline.html.
+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+void AAudioPlayer::HandleStreamDisconnected() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DLOG(LS_INFO) << "HandleStreamDisconnected";
+  if (!initialized_ || !playing_) {
+    return;
+  }
+  // Perform a restart by first closing the disconnected stream and then start
+  // a new stream; this time using the new (preferred) audio output device.
+  StopPlayout();
+  InitPlayout();
+  StartPlayout();
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_player.h

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+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
+
+#include <aaudio/AAudio.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/aaudio_wrapper.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+class AudioManager;
+
+// Implements low-latency 16-bit mono PCM audio output support for Android
+// using the C based AAudio API.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will DCHECK if any method is called on an invalid thread. Audio buffers
+// are requested on a dedicated high-priority thread owned by AAudio.
+//
+// The existing design forces the user to call InitPlayout() after StopPlayout()
+// to be able to call StartPlayout() again. This is in line with how the Java-
+// based implementation works.
+//
+// An audio stream can be disconnected, e.g. when an audio device is removed.
+// This implementation will restart the audio stream using the new preferred
+// device if such an event happens.
+//
+// Also supports automatic buffer-size adjustment based on underrun detections
+// where the internal AAudio buffer can be increased when needed. It will
+// reduce the risk of underruns (~glitches) at the expense of an increased
+// latency.
+class AAudioPlayer final : public AAudioObserverInterface {
+ public:
+  explicit AAudioPlayer(AudioManager* audio_manager);
+  ~AAudioPlayer();
+
+  int Init();
+  int Terminate();
+
+  int InitPlayout();
+  bool PlayoutIsInitialized() const;
+
+  int StartPlayout();
+  int StopPlayout();
+  bool Playing() const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  // Not implemented in AAudio.
+  int SpeakerVolumeIsAvailable(bool& available);  // NOLINT
+  int SetSpeakerVolume(uint32_t volume) { return -1; }
+  int SpeakerVolume(uint32_t& volume) const { return -1; }        // NOLINT
+  int MaxSpeakerVolume(uint32_t& maxVolume) const { return -1; }  // NOLINT
+  int MinSpeakerVolume(uint32_t& minVolume) const { return -1; }  // NOLINT
+
+ protected:
+  // AAudioObserverInterface implementation.
+
+  // For an output stream, this function should render and write `num_frames`
+  // of data in the streams current data format to the `audio_data` buffer.
+  // Called on a real-time thread owned by AAudio.
+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                               int32_t num_frames) override;
+  // AAudio calls this functions if any error occurs on a callback thread.
+  // Called on a real-time thread owned by AAudio.
+  void OnErrorCallback(aaudio_result_t error) override;
+
+ private:
+  // Closes the existing stream and starts a new stream.
+  void HandleStreamDisconnected();
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker main_thread_checker_;
+
+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
+  // real-time thread owned by AAudio. Detached during construction of this
+  // object.
+  SequenceChecker thread_checker_aaudio_;
+
+  // The task queue on which this object is created on.
+  TaskQueueBase* main_thread_;
+
+  // Wraps all AAudio resources. Contains an output stream using the default
+  // output audio device. Can be accessed on both the main thread and the
+  // real-time thread owned by AAudio. See separate AAudio documentation about
+  // thread safety.
+  AAudioWrapper aaudio_;
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples.
+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192
+  // in each callback (once every 4th ms). This class can then ask for 192 and
+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
+  // second callback and also cache non-utilized audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Counts number of detected underrun events reported by AAudio.
+  int32_t underrun_count_ = 0;
+
+  // True only for the first data callback in each audio session.
+  bool first_data_callback_ = true;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) =
+      nullptr;
+
+  bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false;
+  bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false;
+
+  // Estimated latency between writing an audio frame to the output stream and
+  // the time that same frame is played out on the output audio device.
+  double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_recorder.cc

@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_recorder.h"
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+AAudioRecorder::AAudioRecorder(AudioManager* audio_manager)
+    : main_thread_(TaskQueueBase::Current()),
+      aaudio_(audio_manager, AAUDIO_DIRECTION_INPUT, this) {
+  RTC_LOG(LS_INFO) << "ctor";
+  thread_checker_aaudio_.Detach();
+}
+
+AAudioRecorder::~AAudioRecorder() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  RTC_LOG(LS_INFO) << "detected owerflows: " << overflow_count_;
+}
+
+int AAudioRecorder::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (aaudio_.audio_parameters().channels() == 2) {
+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
+  }
+  return 0;
+}
+
+int AAudioRecorder::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int AAudioRecorder::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  if (!aaudio_.Init()) {
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int AAudioRecorder::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  if (!aaudio_.Start()) {
+    return -1;
+  }
+  overflow_count_ = aaudio_.xrun_count();
+  first_data_callback_ = true;
+  recording_ = true;
+  return 0;
+}
+
+int AAudioRecorder::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!aaudio_.Stop()) {
+    return -1;
+  }
+  thread_checker_aaudio_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  return 0;
+}
+
+void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
+  audio_device_buffer_->SetRecordingSampleRate(audio_parameters.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(audio_parameters.channels());
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to deliver any number
+  // of samples (and not only multiples of 10ms which WebRTC uses) to match the
+  // native AAudio buffer size.
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+}
+
+int AAudioRecorder::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+int AAudioRecorder::EnableBuiltInAGC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAGC: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+int AAudioRecorder::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS: " << enable;
+  RTC_LOG(LS_ERROR) << "Not implemented";
+  return -1;
+}
+
+void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
+  // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+    // The stream is disconnected and any attempt to use it will return
+    // AAUDIO_ERROR_DISCONNECTED..
+    RTC_LOG(LS_WARNING) << "Input stream disconnected => restart is required";
+    // AAudio documentation states: "You should not close or reopen the stream
+    // from the callback, use another thread instead". A message is therefore
+    // sent to the main thread to do the restart operation.
+    RTC_DCHECK(main_thread_);
+    main_thread_->PostTask([this] { HandleStreamDisconnected(); });
+  }
+}
+
+// Read and process `num_frames` of data from the `audio_data` buffer.
+// TODO(henrika): possibly add trace here to be included in systrace.
+// See https://developer.android.com/studio/profile/systrace-commandline.html.
+aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
+    void* audio_data,
+    int32_t num_frames) {
+  // TODO(henrika): figure out why we sometimes hit this one.
+  // RTC_DCHECK(thread_checker_aaudio_.IsCurrent());
+  // RTC_LOG(LS_INFO) << "OnDataCallback: " << num_frames;
+  // Drain the input buffer at first callback to ensure that it does not
+  // contain any old data. Will also ensure that the lowest possible latency
+  // is obtained.
+  if (first_data_callback_) {
+    RTC_LOG(LS_INFO) << "--- First input data callback: "
+                        "device id="
+                     << aaudio_.device_id();
+    aaudio_.ClearInputStream(audio_data, num_frames);
+    first_data_callback_ = false;
+  }
+  // Check if the overflow counter has increased and if so log a warning.
+  // TODO(henrika): possible add UMA stat or capacity extension.
+  const int32_t overflow_count = aaudio_.xrun_count();
+  if (overflow_count > overflow_count_) {
+    RTC_LOG(LS_ERROR) << "Overflow detected: " << overflow_count;
+    overflow_count_ = overflow_count;
+  }
+  // Estimated time between an audio frame was recorded by the input device and
+  // it can read on the input stream.
+  latency_millis_ = aaudio_.EstimateLatencyMillis();
+  // TODO(henrika): use for development only.
+  if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
+    RTC_DLOG(LS_INFO) << "input latency: " << latency_millis_
+                      << ", num_frames: " << num_frames;
+  }
+  // Copy recorded audio in `audio_data` to the WebRTC sink using the
+  // FineAudioBuffer object.
+  fine_audio_buffer_->DeliverRecordedData(
+      rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
+                         aaudio_.samples_per_frame() * num_frames),
+      static_cast<int>(latency_millis_ + 0.5));
+
+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+void AAudioRecorder::HandleStreamDisconnected() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_LOG(LS_INFO) << "HandleStreamDisconnected";
+  if (!initialized_ || !recording_) {
+    return;
+  }
+  // Perform a restart by first closing the disconnected stream and then start
+  // a new stream; this time using the new (preferred) audio input device.
+  // TODO(henrika): resolve issue where a one restart attempt leads to a long
+  // sequence of new calls to OnErrorCallback().
+  // See b/73148976 for details.
+  StopRecording();
+  InitRecording();
+  StartRecording();
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_recorder.h

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
+
+#include <aaudio/AAudio.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_device/android/aaudio_wrapper.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+class FineAudioBuffer;
+class AudioManager;
+
+// Implements low-latency 16-bit mono PCM audio input support for Android
+// using the C based AAudio API.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
+// are delivered on a dedicated high-priority thread owned by AAudio.
+//
+// The existing design forces the user to call InitRecording() after
+// StopRecording() to be able to call StartRecording() again. This is in line
+// with how the Java- based implementation works.
+//
+// TODO(henrika): add comments about device changes and adaptive buffer
+// management.
+class AAudioRecorder : public AAudioObserverInterface {
+ public:
+  explicit AAudioRecorder(AudioManager* audio_manager);
+  ~AAudioRecorder();
+
+  int Init();
+  int Terminate();
+
+  int InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int StartRecording();
+  int StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  double latency_millis() const { return latency_millis_; }
+
+  // TODO(henrika): add support using AAudio APIs when available.
+  int EnableBuiltInAEC(bool enable);
+  int EnableBuiltInAGC(bool enable);
+  int EnableBuiltInNS(bool enable);
+
+ protected:
+  // AAudioObserverInterface implementation.
+
+  // For an input stream, this function should read `num_frames` of recorded
+  // data, in the stream's current data format, from the `audio_data` buffer.
+  // Called on a real-time thread owned by AAudio.
+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                               int32_t num_frames) override;
+
+  // AAudio calls this function if any error occurs on a callback thread.
+  // Called on a real-time thread owned by AAudio.
+  void OnErrorCallback(aaudio_result_t error) override;
+
+ private:
+  // Closes the existing stream and starts a new stream.
+  void HandleStreamDisconnected();
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
+  // real-time thread owned by AAudio. Detached during construction of this
+  // object.
+  SequenceChecker thread_checker_aaudio_;
+
+  // The thread on which this object is created on.
+  TaskQueueBase* main_thread_;
+
+  // Wraps all AAudio resources. Contains an input stream using the default
+  // input audio device.
+  AAudioWrapper aaudio_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
+
+  bool initialized_ = false;
+  bool recording_ = false;
+
+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
+  // chunks of audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Counts number of detected overflow events reported by AAudio.
+  int32_t overflow_count_ = 0;
+
+  // Estimated time between an audio frame was recorded by the input device and
+  // it can read on the input stream.
+  double latency_millis_ = 0;
+
+  // True only for the first data callback in each audio session.
+  bool first_data_callback_ = true;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_wrapper.cc

@@ -0,0 +1,499 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/aaudio_wrapper.h"
+
+#include "modules/audio_device/android/audio_manager.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+#define LOG_ON_ERROR(op)                                                      \
+  do {                                                                        \
+    aaudio_result_t result = (op);                                            \
+    if (result != AAUDIO_OK) {                                                \
+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
+    }                                                                         \
+  } while (0)
+
+#define RETURN_ON_ERROR(op, ...)                                              \
+  do {                                                                        \
+    aaudio_result_t result = (op);                                            \
+    if (result != AAUDIO_OK) {                                                \
+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
+      return __VA_ARGS__;                                                     \
+    }                                                                         \
+  } while (0)
+
+namespace webrtc {
+
+namespace {
+
+const char* DirectionToString(aaudio_direction_t direction) {
+  switch (direction) {
+    case AAUDIO_DIRECTION_OUTPUT:
+      return "OUTPUT";
+    case AAUDIO_DIRECTION_INPUT:
+      return "INPUT";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* SharingModeToString(aaudio_sharing_mode_t mode) {
+  switch (mode) {
+    case AAUDIO_SHARING_MODE_EXCLUSIVE:
+      return "EXCLUSIVE";
+    case AAUDIO_SHARING_MODE_SHARED:
+      return "SHARED";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
+  switch (mode) {
+    case AAUDIO_PERFORMANCE_MODE_NONE:
+      return "NONE";
+    case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
+      return "POWER_SAVING";
+    case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
+      return "LOW_LATENCY";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+const char* FormatToString(int32_t id) {
+  switch (id) {
+    case AAUDIO_FORMAT_INVALID:
+      return "INVALID";
+    case AAUDIO_FORMAT_UNSPECIFIED:
+      return "UNSPECIFIED";
+    case AAUDIO_FORMAT_PCM_I16:
+      return "PCM_I16";
+    case AAUDIO_FORMAT_PCM_FLOAT:
+      return "FLOAT";
+    default:
+      return "UNKNOWN";
+  }
+}
+
+void ErrorCallback(AAudioStream* stream,
+                   void* user_data,
+                   aaudio_result_t error) {
+  RTC_DCHECK(user_data);
+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
+  RTC_LOG(LS_WARNING) << "ErrorCallback: "
+                      << DirectionToString(aaudio_wrapper->direction());
+  RTC_DCHECK(aaudio_wrapper->observer());
+  aaudio_wrapper->observer()->OnErrorCallback(error);
+}
+
+aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
+                                           void* user_data,
+                                           void* audio_data,
+                                           int32_t num_frames) {
+  RTC_DCHECK(user_data);
+  RTC_DCHECK(audio_data);
+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
+  RTC_DCHECK(aaudio_wrapper->observer());
+  return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
+}
+
+// Wraps the stream builder object to ensure that it is released properly when
+// the stream builder goes out of scope.
+class ScopedStreamBuilder {
+ public:
+  ScopedStreamBuilder() {
+    LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
+    RTC_DCHECK(builder_);
+  }
+  ~ScopedStreamBuilder() {
+    if (builder_) {
+      LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
+    }
+  }
+
+  AAudioStreamBuilder* get() const { return builder_; }
+
+ private:
+  AAudioStreamBuilder* builder_ = nullptr;
+};
+
+}  // namespace
+
+AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
+                             aaudio_direction_t direction,
+                             AAudioObserverInterface* observer)
+    : direction_(direction), observer_(observer) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(observer_);
+  direction_ == AAUDIO_DIRECTION_OUTPUT
+      ? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
+      : audio_parameters_ = audio_manager->GetRecordAudioParameters();
+  aaudio_thread_checker_.Detach();
+  RTC_LOG(LS_INFO) << audio_parameters_.ToString();
+}
+
+AAudioWrapper::~AAudioWrapper() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!stream_);
+}
+
+bool AAudioWrapper::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Creates a stream builder which can be used to open an audio stream.
+  ScopedStreamBuilder builder;
+  // Configures the stream builder using audio parameters given at construction.
+  SetStreamConfiguration(builder.get());
+  // Opens a stream based on options in the stream builder.
+  if (!OpenStream(builder.get())) {
+    return false;
+  }
+  // Ensures that the opened stream could activate the requested settings.
+  if (!VerifyStreamConfiguration()) {
+    return false;
+  }
+  // Optimizes the buffer scheme for lowest possible latency and creates
+  // additional buffer logic to match the 10ms buffer size used in WebRTC.
+  if (!OptimizeBuffers()) {
+    return false;
+  }
+  LogStreamState();
+  return true;
+}
+
+bool AAudioWrapper::Start() {
+  RTC_LOG(LS_INFO) << "Start";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // TODO(henrika): this state check might not be needed.
+  aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
+  if (current_state != AAUDIO_STREAM_STATE_OPEN) {
+    RTC_LOG(LS_ERROR) << "Invalid state: "
+                      << AAudio_convertStreamStateToText(current_state);
+    return false;
+  }
+  // Asynchronous request for the stream to start.
+  RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
+  LogStreamState();
+  return true;
+}
+
+bool AAudioWrapper::Stop() {
+  RTC_LOG(LS_INFO) << "Stop: " << DirectionToString(direction());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Asynchronous request for the stream to stop.
+  RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
+  CloseStream();
+  aaudio_thread_checker_.Detach();
+  return true;
+}
+
+double AAudioWrapper::EstimateLatencyMillis() const {
+  RTC_DCHECK(stream_);
+  double latency_millis = 0.0;
+  if (direction() == AAUDIO_DIRECTION_INPUT) {
+    // For input streams. Best guess we can do is to use the current burst size
+    // as delay estimate.
+    latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
+                     rtc::kNumMillisecsPerSec;
+  } else {
+    int64_t existing_frame_index;
+    int64_t existing_frame_presentation_time;
+    // Get the time at which a particular frame was presented to audio hardware.
+    aaudio_result_t result = AAudioStream_getTimestamp(
+        stream_, CLOCK_MONOTONIC, &existing_frame_index,
+        &existing_frame_presentation_time);
+    // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
+    if (result == AAUDIO_OK) {
+      // Get write index for next audio frame.
+      int64_t next_frame_index = frames_written();
+      // Number of frames between next frame and the existing frame.
+      int64_t frame_index_delta = next_frame_index - existing_frame_index;
+      // Assume the next frame will be written now.
+      int64_t next_frame_write_time = rtc::TimeNanos();
+      // Calculate time when next frame will be presented to the hardware taking
+      // sample rate into account.
+      int64_t frame_time_delta =
+          (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
+      int64_t next_frame_presentation_time =
+          existing_frame_presentation_time + frame_time_delta;
+      // Derive a latency estimate given results above.
+      latency_millis = static_cast<double>(next_frame_presentation_time -
+                                           next_frame_write_time) /
+                       rtc::kNumNanosecsPerMillisec;
+    }
+  }
+  return latency_millis;
+}
+
+// Returns new buffer size or a negative error value if buffer size could not
+// be increased.
+bool AAudioWrapper::IncreaseOutputBufferSize() {
+  RTC_LOG(LS_INFO) << "IncreaseBufferSize";
+  RTC_DCHECK(stream_);
+  RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
+  aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
+  // Try to increase size of buffer with one burst to reduce risk of underrun.
+  buffer_size += frames_per_burst();
+  // Verify that the new buffer size is not larger than max capacity.
+  // TODO(henrika): keep track of case when we reach the capacity limit.
+  const int32_t max_buffer_size = buffer_capacity_in_frames();
+  if (buffer_size > max_buffer_size) {
+    RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
+                      << ") is higher than max: " << max_buffer_size;
+    return false;
+  }
+  RTC_LOG(LS_INFO) << "Updating buffer size to: " << buffer_size
+                   << " (max=" << max_buffer_size << ")";
+  buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
+  if (buffer_size < 0) {
+    RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
+                      << AAudio_convertResultToText(buffer_size);
+    return false;
+  }
+  RTC_LOG(LS_INFO) << "Buffer size changed to: " << buffer_size;
+  return true;
+}
+
+void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
+  RTC_LOG(LS_INFO) << "ClearInputStream";
+  RTC_DCHECK(stream_);
+  RTC_DCHECK(aaudio_thread_checker_.IsCurrent());
+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
+  aaudio_result_t cleared_frames = 0;
+  do {
+    cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
+  } while (cleared_frames > 0);
+}
+
+AAudioObserverInterface* AAudioWrapper::observer() const {
+  return observer_;
+}
+
+AudioParameters AAudioWrapper::audio_parameters() const {
+  return audio_parameters_;
+}
+
+int32_t AAudioWrapper::samples_per_frame() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSamplesPerFrame(stream_);
+}
+
+int32_t AAudioWrapper::buffer_size_in_frames() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getBufferSizeInFrames(stream_);
+}
+
+int32_t AAudioWrapper::buffer_capacity_in_frames() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getBufferCapacityInFrames(stream_);
+}
+
+int32_t AAudioWrapper::device_id() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getDeviceId(stream_);
+}
+
+int32_t AAudioWrapper::xrun_count() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getXRunCount(stream_);
+}
+
+int32_t AAudioWrapper::format() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFormat(stream_);
+}
+
+int32_t AAudioWrapper::sample_rate() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSampleRate(stream_);
+}
+
+int32_t AAudioWrapper::channel_count() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getChannelCount(stream_);
+}
+
+int32_t AAudioWrapper::frames_per_callback() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesPerDataCallback(stream_);
+}
+
+aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getSharingMode(stream_);
+}
+
+aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getPerformanceMode(stream_);
+}
+
+aaudio_stream_state_t AAudioWrapper::stream_state() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getState(stream_);
+}
+
+int64_t AAudioWrapper::frames_written() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesWritten(stream_);
+}
+
+int64_t AAudioWrapper::frames_read() const {
+  RTC_DCHECK(stream_);
+  return AAudioStream_getFramesRead(stream_);
+}
+
+void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
+  RTC_LOG(LS_INFO) << "SetStreamConfiguration";
+  RTC_DCHECK(builder);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Request usage of default primary output/input device.
+  // TODO(henrika): verify that default device follows Java APIs.
+  // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
+  AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
+  // Use preferred sample rate given by the audio parameters.
+  AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
+  // Use preferred channel configuration given by the audio parameters.
+  AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
+  // Always use 16-bit PCM audio sample format.
+  AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
+  // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
+  // Ask for exclusive mode since this will give us the lowest possible latency.
+  // If exclusive mode isn't available, shared mode will be used instead.
+  AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
+  // Use the direction that was given at construction.
+  AAudioStreamBuilder_setDirection(builder, direction_);
+  // TODO(henrika): investigate performance using different performance modes.
+  AAudioStreamBuilder_setPerformanceMode(builder,
+                                         AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+  // Given that WebRTC applications require low latency, our audio stream uses
+  // an asynchronous callback function to transfer data to and from the
+  // application. AAudio executes the callback in a higher-priority thread that
+  // has better performance.
+  AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
+  // Request that AAudio calls this functions if any error occurs on a callback
+  // thread.
+  AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
+}
+
+bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
+  RTC_LOG(LS_INFO) << "OpenStream";
+  RTC_DCHECK(builder);
+  AAudioStream* stream = nullptr;
+  RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
+  stream_ = stream;
+  LogStreamConfiguration();
+  return true;
+}
+
+void AAudioWrapper::CloseStream() {
+  RTC_LOG(LS_INFO) << "CloseStream";
+  RTC_DCHECK(stream_);
+  LOG_ON_ERROR(AAudioStream_close(stream_));
+  stream_ = nullptr;
+}
+
+void AAudioWrapper::LogStreamConfiguration() {
+  RTC_DCHECK(stream_);
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "Stream Configuration: ";
+  ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
+  ss << ", samples per frame=" << samples_per_frame();
+  ss << ", format=" << FormatToString(format());
+  ss << ", sharing mode=" << SharingModeToString(sharing_mode());
+  ss << ", performance mode=" << PerformanceModeToString(performance_mode());
+  ss << ", direction=" << DirectionToString(direction());
+  ss << ", device id=" << AAudioStream_getDeviceId(stream_);
+  ss << ", frames per callback=" << frames_per_callback();
+  RTC_LOG(LS_INFO) << ss.str();
+}
+
+void AAudioWrapper::LogStreamState() {
+  RTC_LOG(LS_INFO) << "AAudio stream state: "
+                   << AAudio_convertStreamStateToText(stream_state());
+}
+
+bool AAudioWrapper::VerifyStreamConfiguration() {
+  RTC_LOG(LS_INFO) << "VerifyStreamConfiguration";
+  RTC_DCHECK(stream_);
+  // TODO(henrika): should we verify device ID as well?
+  if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
+    return false;
+  }
+  if (AAudioStream_getChannelCount(stream_) !=
+      static_cast<int32_t>(audio_parameters().channels())) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
+    return false;
+  }
+  if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
+    return false;
+  }
+  if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
+    return false;
+  }
+  if (AAudioStream_getPerformanceMode(stream_) !=
+      AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
+    RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
+    return false;
+  }
+  if (AAudioStream_getDirection(stream_) != direction()) {
+    RTC_LOG(LS_ERROR) << "Stream direction could not be set";
+    return false;
+  }
+  if (AAudioStream_getSamplesPerFrame(stream_) !=
+      static_cast<int32_t>(audio_parameters().channels())) {
+    RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
+    return false;
+  }
+  return true;
+}
+
+bool AAudioWrapper::OptimizeBuffers() {
+  RTC_LOG(LS_INFO) << "OptimizeBuffers";
+  RTC_DCHECK(stream_);
+  // Maximum number of frames that can be filled without blocking.
+  RTC_LOG(LS_INFO) << "max buffer capacity in frames: "
+                   << buffer_capacity_in_frames();
+  // Query the number of frames that the application should read or write at
+  // one time for optimal performance.
+  int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
+  RTC_LOG(LS_INFO) << "frames per burst for optimal performance: "
+                   << frames_per_burst;
+  frames_per_burst_ = frames_per_burst;
+  if (direction() == AAUDIO_DIRECTION_INPUT) {
+    // There is no point in calling setBufferSizeInFrames() for input streams
+    // since it has no effect on the performance (latency in this case).
+    return true;
+  }
+  // Set buffer size to same as burst size to guarantee lowest possible latency.
+  // This size might change for output streams if underruns are detected and
+  // automatic buffer adjustment is enabled.
+  AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
+  int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
+  if (buffer_size != frames_per_burst) {
+    RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
+    return false;
+  }
+  // Maximum number of frames that can be filled without blocking.
+  RTC_LOG(LS_INFO) << "buffer burst size in frames: " << buffer_size;
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/aaudio_wrapper.h

@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
+
+#include <aaudio/AAudio.h>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioManager;
+
+// AAudio callback interface for audio transport to/from the AAudio stream.
+// The interface also contains an error callback method for notifications of
+// e.g. device changes.
+class AAudioObserverInterface {
+ public:
+  // Audio data will be passed in our out of this function dependning on the
+  // direction of the audio stream. This callback function will be called on a
+  // real-time thread owned by AAudio.
+  virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data,
+                                                       int32_t num_frames) = 0;
+  // AAudio will call this functions if any error occurs on a callback thread.
+  // In response, this function could signal or launch another thread to reopen
+  // a stream on another device. Do not reopen the stream in this callback.
+  virtual void OnErrorCallback(aaudio_result_t error) = 0;
+
+ protected:
+  virtual ~AAudioObserverInterface() {}
+};
+
+// Utility class which wraps the C-based AAudio API into a more handy C++ class
+// where the underlying resources (AAudioStreamBuilder and AAudioStream) are
+// encapsulated. User must set the direction (in or out) at construction since
+// it defines the stream type and the direction of the data flow in the
+// AAudioObserverInterface.
+//
+// AAudio is a new Android C API introduced in the Android O (26) release.
+// It is designed for high-performance audio applications that require low
+// latency. Applications communicate with AAudio by reading and writing data
+// to streams.
+//
+// Each stream is attached to a single audio device, where each audio device
+// has a unique ID. The ID can be used to bind an audio stream to a specific
+// audio device but this implementation lets AAudio choose the default primary
+// device instead (device selection takes place in Java). A stream can only
+// move data in one direction. When a stream is opened, Android checks to
+// ensure that the audio device and stream direction agree.
+class AAudioWrapper {
+ public:
+  AAudioWrapper(AudioManager* audio_manager,
+                aaudio_direction_t direction,
+                AAudioObserverInterface* observer);
+  ~AAudioWrapper();
+
+  bool Init();
+  bool Start();
+  bool Stop();
+
+  // For output streams: estimates latency between writing an audio frame to
+  // the output stream and the time that same frame is played out on the output
+  // audio device.
+  // For input streams: estimates latency between reading an audio frame from
+  // the input stream and the time that same frame was recorded on the input
+  // audio device.
+  double EstimateLatencyMillis() const;
+
+  // Increases the internal buffer size for output streams by one burst size to
+  // reduce the risk of underruns. Can be used while a stream is active.
+  bool IncreaseOutputBufferSize();
+
+  // Drains the recording stream of any existing data by reading from it until
+  // it's empty. Can be used to clear out old data before starting a new audio
+  // session.
+  void ClearInputStream(void* audio_data, int32_t num_frames);
+
+  AAudioObserverInterface* observer() const;
+  AudioParameters audio_parameters() const;
+  int32_t samples_per_frame() const;
+  int32_t buffer_size_in_frames() const;
+  int32_t buffer_capacity_in_frames() const;
+  int32_t device_id() const;
+  int32_t xrun_count() const;
+  int32_t format() const;
+  int32_t sample_rate() const;
+  int32_t channel_count() const;
+  int32_t frames_per_callback() const;
+  aaudio_sharing_mode_t sharing_mode() const;
+  aaudio_performance_mode_t performance_mode() const;
+  aaudio_stream_state_t stream_state() const;
+  int64_t frames_written() const;
+  int64_t frames_read() const;
+  aaudio_direction_t direction() const { return direction_; }
+  AAudioStream* stream() const { return stream_; }
+  int32_t frames_per_burst() const { return frames_per_burst_; }
+
+ private:
+  void SetStreamConfiguration(AAudioStreamBuilder* builder);
+  bool OpenStream(AAudioStreamBuilder* builder);
+  void CloseStream();
+  void LogStreamConfiguration();
+  void LogStreamState();
+  bool VerifyStreamConfiguration();
+  bool OptimizeBuffers();
+
+  SequenceChecker thread_checker_;
+  SequenceChecker aaudio_thread_checker_;
+  AudioParameters audio_parameters_;
+  const aaudio_direction_t direction_;
+  AAudioObserverInterface* observer_ = nullptr;
+  AAudioStream* stream_ = nullptr;
+  int32_t frames_per_burst_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_common.h

@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_
+
+namespace webrtc {
+
+const int kDefaultSampleRate = 44100;
+// Delay estimates for the two different supported modes. These values are based
+// on real-time round-trip delay estimates on a large set of devices and they
+// are lower bounds since the filter length is 128 ms, so the AEC works for
+// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
+// cases, the lowest delay estimate will not be utilized since devices that
+// support low-latency output audio often supports HW AEC as well.
+const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
+const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_COMMON_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_device_template.h

@@ -0,0 +1,441 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+// InputType/OutputType can be any class that implements the capturing/rendering
+// part of the AudioDeviceGeneric API.
+// Construction and destruction must be done on one and the same thread. Each
+// internal implementation of InputType and OutputType will RTC_DCHECK if that
+// is not the case. All implemented methods must also be called on the same
+// thread. See comments in each InputType/OutputType class for more info.
+// It is possible to call the two static methods (SetAndroidAudioDeviceObjects
+// and ClearAndroidAudioDeviceObjects) from a different thread but both will
+// RTC_CHECK that the calling thread is attached to a Java VM.
+
+template <class InputType, class OutputType>
+class AudioDeviceTemplate : public AudioDeviceGeneric {
+ public:
+  AudioDeviceTemplate(AudioDeviceModule::AudioLayer audio_layer,
+                      AudioManager* audio_manager)
+      : audio_layer_(audio_layer),
+        audio_manager_(audio_manager),
+        output_(audio_manager_),
+        input_(audio_manager_),
+        initialized_(false) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_CHECK(audio_manager);
+    audio_manager_->SetActiveAudioLayer(audio_layer);
+  }
+
+  virtual ~AudioDeviceTemplate() { RTC_LOG(LS_INFO) << __FUNCTION__; }
+
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    audioLayer = audio_layer_;
+    return 0;
+  }
+
+  InitStatus Init() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    RTC_DCHECK(!initialized_);
+    if (!audio_manager_->Init()) {
+      return InitStatus::OTHER_ERROR;
+    }
+    if (output_.Init() != 0) {
+      audio_manager_->Close();
+      return InitStatus::PLAYOUT_ERROR;
+    }
+    if (input_.Init() != 0) {
+      output_.Terminate();
+      audio_manager_->Close();
+      return InitStatus::RECORDING_ERROR;
+    }
+    initialized_ = true;
+    return InitStatus::OK;
+  }
+
+  int32_t Terminate() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    int32_t err = input_.Terminate();
+    err |= output_.Terminate();
+    err |= !audio_manager_->Close();
+    initialized_ = false;
+    RTC_DCHECK_EQ(err, 0);
+    return err;
+  }
+
+  bool Initialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    return initialized_;
+  }
+
+  int16_t PlayoutDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 1;
+  }
+
+  int16_t RecordingDevices() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 1;
+  }
+
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+
+  int32_t SetPlayoutDevice(uint16_t index) override {
+    // OK to use but it has no effect currently since device selection is
+    // done using Andoid APIs instead.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+      // OK to use but it has no effect currently since device selection is
+      // done using Andoid APIs instead.
+      RTC_DLOG(LS_INFO) << __FUNCTION__;
+      return 0;
+  }
+
+  int32_t SetRecordingDevice(uint16_t index) override {
+    // OK to use but it has no effect currently since device selection is
+    // done using Andoid APIs instead.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+      // OK to use but it has no effect currently since device selection is
+      // done using Andoid APIs instead.
+      RTC_DLOG(LS_INFO) << __FUNCTION__;
+      return 0;
+  }
+
+  int32_t PlayoutIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = true;
+    return 0;
+  }
+
+  int32_t InitPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.InitPlayout();
+  }
+
+  bool PlayoutIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.PlayoutIsInitialized();
+  }
+
+  int32_t RecordingIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = true;
+    return 0;
+  }
+
+  int32_t InitRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return input_.InitRecording();
+  }
+
+  bool RecordingIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return input_.RecordingIsInitialized();
+  }
+
+  int32_t StartPlayout() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The application should use MODE_IN_COMMUNICATION audio mode!";
+    }
+    return output_.StartPlayout();
+  }
+
+  int32_t StopPlayout() override {
+    // Avoid using audio manger (JNI/Java cost) if playout was inactive.
+    if (!Playing())
+      return 0;
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int32_t err = output_.StopPlayout();
+    return err;
+  }
+
+  bool Playing() const override {
+    RTC_LOG(LS_INFO) << __FUNCTION__;
+    return output_.Playing();
+  }
+
+  int32_t StartRecording() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!audio_manager_->IsCommunicationModeEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The application should use MODE_IN_COMMUNICATION audio mode!";
+    }
+    return input_.StartRecording();
+  }
+
+  int32_t StopRecording() override {
+    // Avoid using audio manger (JNI/Java cost) if recording was inactive.
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!Recording())
+      return 0;
+    int32_t err = input_.StopRecording();
+    return err;
+  }
+
+  bool Recording() const override { return input_.Recording(); }
+
+  int32_t InitSpeaker() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  bool SpeakerIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return true;
+  }
+
+  int32_t InitMicrophone() override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return 0;
+  }
+
+  bool MicrophoneIsInitialized() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return true;
+  }
+
+  int32_t SpeakerVolumeIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SpeakerVolumeIsAvailable(available);
+  }
+
+  int32_t SetSpeakerVolume(uint32_t volume) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SetSpeakerVolume(volume);
+  }
+
+  int32_t SpeakerVolume(uint32_t& volume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.SpeakerVolume(volume);
+  }
+
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.MaxSpeakerVolume(maxVolume);
+  }
+
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return output_.MinSpeakerVolume(minVolume);
+  }
+
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override {
+    available = false;
+    return -1;
+  }
+
+  int32_t SetMicrophoneVolume(uint32_t volume) override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t MicrophoneVolume(uint32_t& volume) const override {
+    RTC_CHECK_NOTREACHED();
+    return -1;
+  }
+
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  int32_t SpeakerMuteIsAvailable(bool& available) override {
+    return false;
+  }
+
+  int32_t SetSpeakerMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t SpeakerMute(bool& enabled) const override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t MicrophoneMuteIsAvailable(bool& available) override {
+    return false;
+  }
+
+  int32_t SetMicrophoneMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
+
+  int32_t MicrophoneMute(bool& enabled) const override {
+    RTC_CHECK_NOTREACHED();
+  }
+
+  // Returns true if the audio manager has been configured to support stereo
+  // and false otherwised. Default is mono.
+  int32_t StereoPlayoutIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = audio_manager_->IsStereoPlayoutSupported();
+    return 0;
+  }
+
+  int32_t SetStereoPlayout(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    bool available = audio_manager_->IsStereoPlayoutSupported();
+    // Android does not support changes between mono and stero on the fly.
+    // Instead, the native audio layer is configured via the audio manager
+    // to either support mono or stereo. It is allowed to call this method
+    // if that same state is not modified.
+    return (enable == available) ? 0 : -1;
+  }
+
+  int32_t StereoPlayout(bool& enabled) const override {
+    enabled = audio_manager_->IsStereoPlayoutSupported();
+    return 0;
+  }
+
+  int32_t StereoRecordingIsAvailable(bool& available) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    available = audio_manager_->IsStereoRecordSupported();
+    return 0;
+  }
+
+  int32_t SetStereoRecording(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    bool available = audio_manager_->IsStereoRecordSupported();
+    // Android does not support changes between mono and stero on the fly.
+    // Instead, the native audio layer is configured via the audio manager
+    // to either support mono or stereo. It is allowed to call this method
+    // if that same state is not modified.
+    return (enable == available) ? 0 : -1;
+  }
+
+  int32_t StereoRecording(bool& enabled) const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    enabled = audio_manager_->IsStereoRecordSupported();
+    return 0;
+  }
+
+  int32_t PlayoutDelay(uint16_t& delay_ms) const override {
+    // Best guess we can do is to use half of the estimated total delay.
+    delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2;
+    RTC_DCHECK_GT(delay_ms, 0);
+    return 0;
+  }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    output_.AttachAudioBuffer(audioBuffer);
+    input_.AttachAudioBuffer(audioBuffer);
+  }
+
+  // Returns true if the device both supports built in AEC and the device
+  // is not blacklisted.
+  // Currently, if OpenSL ES is used in both directions, this method will still
+  // report the correct value and it has the correct effect. As an example:
+  // a device supports built in AEC and this method returns true. Libjingle
+  // will then disable the WebRTC based AEC and that will work for all devices
+  // (mainly Nexus) even when OpenSL ES is used for input since our current
+  // implementation will enable built-in AEC by default also for OpenSL ES.
+  // The only "bad" thing that happens today is that when Libjingle calls
+  // OpenSLESRecorder::EnableBuiltInAEC() it will not have any real effect and
+  // a "Not Implemented" log will be filed. This non-perfect state will remain
+  // until I have added full support for audio effects based on OpenSL ES APIs.
+  bool BuiltInAECIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsAcousticEchoCancelerSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInAEC(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
+    return input_.EnableBuiltInAEC(enable);
+  }
+
+  // Returns true if the device both supports built in AGC and the device
+  // is not blacklisted.
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  // In addition, see comments for BuiltInAECIsAvailable().
+  bool BuiltInAGCIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsAutomaticGainControlSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInAGC(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInAGCIsAvailable()) << "HW AGC is not available";
+    return input_.EnableBuiltInAGC(enable);
+  }
+
+  // Returns true if the device both supports built in NS and the device
+  // is not blacklisted.
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  // In addition, see comments for BuiltInAECIsAvailable().
+  bool BuiltInNSIsAvailable() const override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_manager_->IsNoiseSuppressorSupported();
+  }
+
+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
+  int32_t EnableBuiltInNS(bool enable) override {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
+    return input_.EnableBuiltInNS(enable);
+  }
+
+ private:
+  SequenceChecker thread_checker_;
+
+  // Local copy of the audio layer set during construction of the
+  // AudioDeviceModuleImpl instance. Read only value.
+  const AudioDeviceModule::AudioLayer audio_layer_;
+
+  // Non-owning raw pointer to AudioManager instance given to use at
+  // construction. The real object is owned by AudioDeviceModuleImpl and the
+  // life time is the same as that of the AudioDeviceModuleImpl, hence there
+  // is no risk of reading a NULL pointer at any time in this class.
+  AudioManager* const audio_manager_;
+
+  OutputType output_;
+
+  InputType input_;
+
+  bool initialized_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_DEVICE_TEMPLATE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_manager.cc

@@ -0,0 +1,318 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_manager.h"
+
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/utility/include/helpers_android.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+
+namespace webrtc {
+
+// AudioManager::JavaAudioManager implementation
+AudioManager::JavaAudioManager::JavaAudioManager(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_manager)
+    : audio_manager_(std::move(audio_manager)),
+      init_(native_reg->GetMethodId("init", "()Z")),
+      dispose_(native_reg->GetMethodId("dispose", "()V")),
+      is_communication_mode_enabled_(
+          native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
+      is_device_blacklisted_for_open_sles_usage_(
+          native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
+                                  "()Z")) {
+  RTC_LOG(LS_INFO) << "JavaAudioManager::ctor";
+}
+
+AudioManager::JavaAudioManager::~JavaAudioManager() {
+  RTC_LOG(LS_INFO) << "JavaAudioManager::~dtor";
+}
+
+bool AudioManager::JavaAudioManager::Init() {
+  return audio_manager_->CallBooleanMethod(init_);
+}
+
+void AudioManager::JavaAudioManager::Close() {
+  audio_manager_->CallVoidMethod(dispose_);
+}
+
+bool AudioManager::JavaAudioManager::IsCommunicationModeEnabled() {
+  return audio_manager_->CallBooleanMethod(is_communication_mode_enabled_);
+}
+
+bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
+  return audio_manager_->CallBooleanMethod(
+      is_device_blacklisted_for_open_sles_usage_);
+}
+
+// AudioManager implementation
+AudioManager::AudioManager()
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
+      initialized_(false),
+      hardware_aec_(false),
+      hardware_agc_(false),
+      hardware_ns_(false),
+      low_latency_playout_(false),
+      low_latency_record_(false),
+      delay_estimate_in_milliseconds_(0) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
+      arraysize(native_methods));
+  j_audio_manager_.reset(
+      new JavaAudioManager(j_native_registration_.get(),
+                           j_native_registration_->NewObject(
+                               "<init>", "(J)V", PointerTojlong(this))));
+}
+
+AudioManager::~AudioManager() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Close();
+}
+
+void AudioManager::SetActiveAudioLayer(
+    AudioDeviceModule::AudioLayer audio_layer) {
+  RTC_LOG(LS_INFO) << "SetActiveAudioLayer: " << audio_layer;
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  // Store the currently utilized audio layer.
+  audio_layer_ = audio_layer;
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not. However, it is also possible
+  // that the user explicitly selects the high-latency audio path, hence we use
+  // the selected `audio_layer` here to set the delay estimate.
+  delay_estimate_in_milliseconds_ =
+      (audio_layer == AudioDeviceModule::kAndroidJavaAudio)
+          ? kHighLatencyModeDelayEstimateInMilliseconds
+          : kLowLatencyModeDelayEstimateInMilliseconds;
+  RTC_LOG(LS_INFO) << "delay_estimate_in_milliseconds: "
+                   << delay_estimate_in_milliseconds_;
+}
+
+SLObjectItf AudioManager::GetOpenSLEngine() {
+  RTC_LOG(LS_INFO) << "GetOpenSLEngine";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Only allow usage of OpenSL ES if such an audio layer has been specified.
+  if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
+      audio_layer_ !=
+          AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
+    RTC_LOG(LS_INFO)
+        << "Unable to create OpenSL engine for the current audio layer: "
+        << audio_layer_;
+    return nullptr;
+  }
+  // OpenSL ES for Android only supports a single engine per application.
+  // If one already has been created, return existing object instead of
+  // creating a new.
+  if (engine_object_.Get() != nullptr) {
+    RTC_LOG(LS_WARNING)
+        << "The OpenSL ES engine object has already been created";
+    return engine_object_.Get();
+  }
+  // Create the engine object in thread safe mode.
+  const SLEngineOption option[] = {
+      {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
+  SLresult result =
+      slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
+  if (result != SL_RESULT_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "slCreateEngine() failed: "
+                      << GetSLErrorString(result);
+    engine_object_.Reset();
+    return nullptr;
+  }
+  // Realize the SL Engine in synchronous mode.
+  result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
+  if (result != SL_RESULT_SUCCESS) {
+    RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result);
+    engine_object_.Reset();
+    return nullptr;
+  }
+  // Finally return the SLObjectItf interface of the engine object.
+  return engine_object_.Get();
+}
+
+bool AudioManager::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
+  if (!j_audio_manager_->Init()) {
+    RTC_LOG(LS_ERROR) << "Init() failed";
+    return false;
+  }
+  initialized_ = true;
+  return true;
+}
+
+bool AudioManager::Close() {
+  RTC_LOG(LS_INFO) << "Close";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_)
+    return true;
+  j_audio_manager_->Close();
+  initialized_ = false;
+  return true;
+}
+
+bool AudioManager::IsCommunicationModeEnabled() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_manager_->IsCommunicationModeEnabled();
+}
+
+bool AudioManager::IsAcousticEchoCancelerSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_aec_;
+}
+
+bool AudioManager::IsAutomaticGainControlSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_agc_;
+}
+
+bool AudioManager::IsNoiseSuppressorSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return hardware_ns_;
+}
+
+bool AudioManager::IsLowLatencyPlayoutSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // Some devices are blacklisted for usage of OpenSL ES even if they report
+  // that low-latency playout is supported. See b/21485703 for details.
+  return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage()
+             ? false
+             : low_latency_playout_;
+}
+
+bool AudioManager::IsLowLatencyRecordSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return low_latency_record_;
+}
+
+bool AudioManager::IsProAudioSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  // TODO(henrika): return the state independently of if OpenSL ES is
+  // blacklisted or not for now. We could use the same approach as in
+  // IsLowLatencyPlayoutSupported() but I can't see the need for it yet.
+  return pro_audio_;
+}
+
+// TODO(henrika): improve comments...
+bool AudioManager::IsAAudioSupported() const {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+  return a_audio_;
+#else
+  return false;
+#endif
+}
+
+bool AudioManager::IsStereoPlayoutSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (playout_parameters_.channels() == 2);
+}
+
+bool AudioManager::IsStereoRecordSupported() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return (record_parameters_.channels() == 2);
+}
+
+int AudioManager::GetDelayEstimateInMilliseconds() const {
+  return delay_estimate_in_milliseconds_;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
+                                                jobject obj,
+                                                jint sample_rate,
+                                                jint output_channels,
+                                                jint input_channels,
+                                                jboolean hardware_aec,
+                                                jboolean hardware_agc,
+                                                jboolean hardware_ns,
+                                                jboolean low_latency_output,
+                                                jboolean low_latency_input,
+                                                jboolean pro_audio,
+                                                jboolean a_audio,
+                                                jint output_buffer_size,
+                                                jint input_buffer_size,
+                                                jlong native_audio_manager) {
+  webrtc::AudioManager* this_object =
+      reinterpret_cast<webrtc::AudioManager*>(native_audio_manager);
+  this_object->OnCacheAudioParameters(
+      env, sample_rate, output_channels, input_channels, hardware_aec,
+      hardware_agc, hardware_ns, low_latency_output, low_latency_input,
+      pro_audio, a_audio, output_buffer_size, input_buffer_size);
+}
+
+void AudioManager::OnCacheAudioParameters(JNIEnv* env,
+                                          jint sample_rate,
+                                          jint output_channels,
+                                          jint input_channels,
+                                          jboolean hardware_aec,
+                                          jboolean hardware_agc,
+                                          jboolean hardware_ns,
+                                          jboolean low_latency_output,
+                                          jboolean low_latency_input,
+                                          jboolean pro_audio,
+                                          jboolean a_audio,
+                                          jint output_buffer_size,
+                                          jint input_buffer_size) {
+  RTC_LOG(LS_INFO)
+      << "OnCacheAudioParameters: "
+         "hardware_aec: "
+      << static_cast<bool>(hardware_aec)
+      << ", hardware_agc: " << static_cast<bool>(hardware_agc)
+      << ", hardware_ns: " << static_cast<bool>(hardware_ns)
+      << ", low_latency_output: " << static_cast<bool>(low_latency_output)
+      << ", low_latency_input: " << static_cast<bool>(low_latency_input)
+      << ", pro_audio: " << static_cast<bool>(pro_audio)
+      << ", a_audio: " << static_cast<bool>(a_audio)
+      << ", sample_rate: " << static_cast<int>(sample_rate)
+      << ", output_channels: " << static_cast<int>(output_channels)
+      << ", input_channels: " << static_cast<int>(input_channels)
+      << ", output_buffer_size: " << static_cast<int>(output_buffer_size)
+      << ", input_buffer_size: " << static_cast<int>(input_buffer_size);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  hardware_aec_ = hardware_aec;
+  hardware_agc_ = hardware_agc;
+  hardware_ns_ = hardware_ns;
+  low_latency_playout_ = low_latency_output;
+  low_latency_record_ = low_latency_input;
+  pro_audio_ = pro_audio;
+  a_audio_ = a_audio;
+  playout_parameters_.reset(sample_rate, static_cast<size_t>(output_channels),
+                            static_cast<size_t>(output_buffer_size));
+  record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels),
+                           static_cast<size_t>(input_buffer_size));
+}
+
+const AudioParameters& AudioManager::GetPlayoutAudioParameters() {
+  RTC_CHECK(playout_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return playout_parameters_;
+}
+
+const AudioParameters& AudioManager::GetRecordAudioParameters() {
+  RTC_CHECK(record_parameters_.is_valid());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return record_parameters_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_manager.h

@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
+
+#include <SLES/OpenSLES.h>
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_config.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements support for functions in the WebRTC audio stack for Android that
+// relies on the AudioManager in android.media. It also populates an
+// AudioParameter structure with native audio parameters detected at
+// construction. This class does not make any audio-related modifications
+// unless Init() is called. Caching audio parameters makes no changes but only
+// reads data from the Java side.
+class AudioManager {
+ public:
+  // Wraps the Java specific parts of the AudioManager into one helper class.
+  // Stores method IDs for all supported methods at construction and then
+  // allows calls like JavaAudioManager::Close() while hiding the Java/JNI
+  // parts that are associated with this call.
+  class JavaAudioManager {
+   public:
+    JavaAudioManager(NativeRegistration* native_registration,
+                     std::unique_ptr<GlobalRef> audio_manager);
+    ~JavaAudioManager();
+
+    bool Init();
+    void Close();
+    bool IsCommunicationModeEnabled();
+    bool IsDeviceBlacklistedForOpenSLESUsage();
+
+   private:
+    std::unique_ptr<GlobalRef> audio_manager_;
+    jmethodID init_;
+    jmethodID dispose_;
+    jmethodID is_communication_mode_enabled_;
+    jmethodID is_device_blacklisted_for_open_sles_usage_;
+  };
+
+  AudioManager();
+  ~AudioManager();
+
+  // Sets the currently active audio layer combination. Must be called before
+  // Init().
+  void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer);
+
+  // Creates and realizes the main (global) Open SL engine object and returns
+  // a reference to it. The engine object is only created at the first call
+  // since OpenSL ES for Android only supports a single engine per application.
+  // Subsequent calls returns the already created engine. The SL engine object
+  // is destroyed when the AudioManager object is deleted. It means that the
+  // engine object will be the first OpenSL ES object to be created and last
+  // object to be destroyed.
+  // Note that NULL will be returned unless the audio layer is specified as
+  // AudioDeviceModule::kAndroidOpenSLESAudio or
+  // AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio.
+  SLObjectItf GetOpenSLEngine();
+
+  // Initializes the audio manager and stores the current audio mode.
+  bool Init();
+  // Revert any setting done by Init().
+  bool Close();
+
+  // Returns true if current audio mode is AudioManager.MODE_IN_COMMUNICATION.
+  bool IsCommunicationModeEnabled() const;
+
+  // Native audio parameters stored during construction.
+  const AudioParameters& GetPlayoutAudioParameters();
+  const AudioParameters& GetRecordAudioParameters();
+
+  // Returns true if the device supports built-in audio effects for AEC, AGC
+  // and NS. Some devices can also be blacklisted for use in combination with
+  // platform effects and these devices will return false.
+  // Can currently only be used in combination with a Java based audio backend
+  // for the recoring side (i.e. using the android.media.AudioRecord API).
+  bool IsAcousticEchoCancelerSupported() const;
+  bool IsAutomaticGainControlSupported() const;
+  bool IsNoiseSuppressorSupported() const;
+
+  // Returns true if the device supports the low-latency audio paths in
+  // combination with OpenSL ES.
+  bool IsLowLatencyPlayoutSupported() const;
+  bool IsLowLatencyRecordSupported() const;
+
+  // Returns true if the device supports (and has been configured for) stereo.
+  // Call the Java API WebRtcAudioManager.setStereoOutput/Input() with true as
+  // paramter to enable stereo. Default is mono in both directions and the
+  // setting is set once and for all when the audio manager object is created.
+  // TODO(henrika): stereo is not supported in combination with OpenSL ES.
+  bool IsStereoPlayoutSupported() const;
+  bool IsStereoRecordSupported() const;
+
+  // Returns true if the device supports pro-audio features in combination with
+  // OpenSL ES.
+  bool IsProAudioSupported() const;
+
+  // Returns true if the device supports AAudio.
+  bool IsAAudioSupported() const;
+
+  // Returns the estimated total delay of this device. Unit is in milliseconds.
+  // The vaule is set once at construction and never changes after that.
+  // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
+  // webrtc::kHighLatencyModeDelayEstimateInMilliseconds.
+  int GetDelayEstimateInMilliseconds() const;
+
+ private:
+  // Called from Java side so we can cache the native audio parameters.
+  // This method will be called by the WebRtcAudioManager constructor, i.e.
+  // on the same thread that this object is created on.
+  static void JNICALL CacheAudioParameters(JNIEnv* env,
+                                           jobject obj,
+                                           jint sample_rate,
+                                           jint output_channels,
+                                           jint input_channels,
+                                           jboolean hardware_aec,
+                                           jboolean hardware_agc,
+                                           jboolean hardware_ns,
+                                           jboolean low_latency_output,
+                                           jboolean low_latency_input,
+                                           jboolean pro_audio,
+                                           jboolean a_audio,
+                                           jint output_buffer_size,
+                                           jint input_buffer_size,
+                                           jlong native_audio_manager);
+  void OnCacheAudioParameters(JNIEnv* env,
+                              jint sample_rate,
+                              jint output_channels,
+                              jint input_channels,
+                              jboolean hardware_aec,
+                              jboolean hardware_agc,
+                              jboolean hardware_ns,
+                              jboolean low_latency_output,
+                              jboolean low_latency_input,
+                              jboolean pro_audio,
+                              jboolean a_audio,
+                              jint output_buffer_size,
+                              jint input_buffer_size);
+
+  // Stores thread ID in the constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  SequenceChecker thread_checker_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioManager.
+  std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
+
+  // Contains the selected audio layer specified by the AudioLayer enumerator
+  // in the AudioDeviceModule class.
+  AudioDeviceModule::AudioLayer audio_layer_;
+
+  // This object is the global entry point of the OpenSL ES API.
+  // After creating the engine object, the application can obtain this object‘s
+  // SLEngineItf interface. This interface contains creation methods for all
+  // the other object types in the API. None of these interface are realized
+  // by this class. It only provides access to the global engine object.
+  webrtc::ScopedSLObjectItf engine_object_;
+
+  // Set to true by Init() and false by Close().
+  bool initialized_;
+
+  // True if device supports hardware (or built-in) AEC.
+  bool hardware_aec_;
+  // True if device supports hardware (or built-in) AGC.
+  bool hardware_agc_;
+  // True if device supports hardware (or built-in) NS.
+  bool hardware_ns_;
+
+  // True if device supports the low-latency OpenSL ES audio path for output.
+  bool low_latency_playout_;
+
+  // True if device supports the low-latency OpenSL ES audio path for input.
+  bool low_latency_record_;
+
+  // True if device supports the low-latency OpenSL ES pro-audio path.
+  bool pro_audio_;
+
+  // True if device supports the low-latency AAudio audio path.
+  bool a_audio_;
+
+  // The delay estimate can take one of two fixed values depending on if the
+  // device supports low-latency output or not.
+  int delay_estimate_in_milliseconds_;
+
+  // Contains native parameters (e.g. sample rate, channel configuration).
+  // Set at construction in OnCacheAudioParameters() which is called from
+  // Java on the same thread as this object is created on.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_merged_screen_record_jni.cc

@@ -0,0 +1,283 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_merged_screen_record_jni.h"
+
+#include <string>
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Scoped class which logs its time of life as a UMA statistic. It generates
+// a histogram which measures the time it takes for a method/scope to execute.
+class ScopedHistogramTimer {
+ public:
+  explicit ScopedHistogramTimer(const std::string& name)
+      : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
+  ~ScopedHistogramTimer() {
+    const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
+    RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
+    RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
+  }
+
+ private:
+  const std::string histogram_name_;
+  int64_t start_time_ms_;
+};
+}  // namespace
+
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioMergedScreenRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")),
+      on_destroy_(native_reg->GetMethodId("onDestroy", "()V")) {}
+
+AudioMergedScreenRecordJni::JavaAudioRecord::~JavaAudioRecord() {
+  audio_record_->CallVoidMethod(on_destroy_);
+}
+
+int AudioMergedScreenRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
+                                                   size_t channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
+}
+
+bool AudioMergedScreenRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
+}
+
+bool AudioMergedScreenRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioMergedScreenRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioMergedScreenRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioMergedScreenRecordJni::AudioMergedScreenRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      recording_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioMergedScreenRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioMergedScreenRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
+      arraysize(native_methods));
+  j_audio_record_.reset(
+      new JavaAudioRecord(j_native_registration_.get(),
+                          j_native_registration_->NewObject(
+                              "<init>", "(JI)V", PointerTojlong(this), 2)));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioMergedScreenRecordJni::~AudioMergedScreenRecordJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioMergedScreenRecordJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioMergedScreenRecordJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int32_t AudioMergedScreenRecordJni::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (frames_per_buffer < 0) {
+    direct_buffer_address_ = nullptr;
+    RTC_LOG(LS_ERROR) << "InitRecording failed";
+    return -1;
+  }
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * bytes_per_frame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioMergedScreenRecordJni::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!recording_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Recording can not start since InitRecording must succeed first";
+    return 0;
+  }
+  ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
+  if (!j_audio_record_->StartRecording()) {
+    RTC_LOG(LS_ERROR) << "StartRecording failed";
+    return -1;
+  }
+  recording_ = true;
+  return 0;
+}
+
+int32_t AudioMergedScreenRecordJni::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!j_audio_record_->StopRecording()) {
+    RTC_LOG(LS_ERROR) << "StopRecording failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+void AudioMergedScreenRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  audio_device_buffer_->SetRecordingChannels(channels);
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
+                << total_delay_in_milliseconds_;
+}
+
+int32_t AudioMergedScreenRecordJni::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioMergedScreenRecordJni::EnableBuiltInAGC(bool enable) {
+  // TODO(henrika): possibly remove when no longer used by any client.
+  RTC_CHECK_NOTREACHED();
+}
+
+int32_t AudioMergedScreenRecordJni::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioMergedScreenRecordJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                      jobject obj,
+                                                      jobject byte_buffer,
+                                                      jlong nativeAudioRecord) {
+  webrtc::AudioMergedScreenRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioMergedScreenRecordJni*>(nativeAudioRecord);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioMergedScreenRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                                jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioMergedScreenRecordJni::DataIsRecorded(JNIEnv* env,
+                                            jobject obj,
+                                            jint length,
+                                            jlong nativeAudioRecord) {
+  webrtc::AudioMergedScreenRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioMergedScreenRecordJni*>(nativeAudioRecord);
+  this_object->OnDataIsRecorded(length);
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordThread'.
+void AudioMergedScreenRecordJni::OnDataIsRecorded(int length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
+                                          frames_per_buffer_);
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // |playDelayMs| parameter only. Components like the AEC only sees the sum
+  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
+    RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_merged_screen_record_jni.h

@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MERGED_SCREEN_RECORD_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MERGED_SCREEN_RECORD_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio input support for Android using the Java
+// AudioRecord interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioRecord.java. This class is created and lives on a thread in
+// C++-land, but recorded audio buffers are delivered on a high-priority
+// thread managed by the Java class.
+//
+// The Java class makes use of AudioEffect features (mainly AEC) which are
+// first available in Jelly Bean. If it is instantiated running against earlier
+// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
+// separately instead.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioMergedScreenRecordJni {
+ public:
+  // Wraps the Java specific parts of the AudioRecordJni into one helper class.
+  class JavaAudioRecord {
+   public:
+    JavaAudioRecord(NativeRegistration* native_registration,
+                    std::unique_ptr<GlobalRef> audio_track);
+    ~JavaAudioRecord();
+
+    int InitRecording(int sample_rate, size_t channels);
+    bool StartRecording();
+    bool StopRecording();
+    bool EnableBuiltInAEC(bool enable);
+    bool EnableBuiltInNS(bool enable);
+
+   private:
+    std::unique_ptr<GlobalRef> audio_record_;
+    jmethodID init_recording_;
+    jmethodID start_recording_;
+    jmethodID stop_recording_;
+    jmethodID enable_built_in_aec_;
+    jmethodID enable_built_in_ns_;
+    jmethodID on_destroy_;
+  };
+
+  explicit AudioMergedScreenRecordJni(AudioManager* audio_manager);
+  ~AudioMergedScreenRecordJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int32_t StartRecording();
+  int32_t StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  int32_t EnableBuiltInAEC(bool enable);
+  int32_t EnableBuiltInAGC(bool enable);
+  int32_t EnableBuiltInNS(bool enable);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
+  // is also stored in |direct_buffer_capacity_in_bytes_|.
+  // This method will be called by the WebRtcAudioRecord constructor, i.e.,
+  // on the same thread that this object is created on.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioRecord);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioRecord object when
+  // recording has started. Each call indicates that there are |length| new
+  // bytes recorded in the memory area |direct_buffer_address_| and it is
+  // now time to send these to the consumer.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioRecordThread'.
+  static void JNICALL DataIsRecorded(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioRecord);
+  void OnDataIsRecorded(int length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnDataIsRecorded() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioRecordJni class.
+  std::unique_ptr<AudioMergedScreenRecordJni::JavaAudioRecord> j_audio_record_;
+
+  // Raw pointer to the audio manger.
+  const AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Delay estimate of the total round-trip delay (input + output).
+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
+  // possible values. See audio_common.h for details.
+  int total_delay_in_milliseconds_;
+
+  // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool recording_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_record_jni.cc

@@ -0,0 +1,280 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_record_jni.h"
+
+#include <string>
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Scoped class which logs its time of life as a UMA statistic. It generates
+// a histogram which measures the time it takes for a method/scope to execute.
+class ScopedHistogramTimer {
+ public:
+  explicit ScopedHistogramTimer(const std::string& name)
+      : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
+  ~ScopedHistogramTimer() {
+    const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
+    RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
+    RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
+  }
+
+ private:
+  const std::string histogram_name_;
+  int64_t start_time_ms_;
+};
+}  // namespace
+
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
+
+AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
+
+int AudioRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
+                                                   size_t channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
+}
+
+bool AudioRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      recording_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
+      arraysize(native_methods));
+  j_audio_record_.reset(
+      new JavaAudioRecord(j_native_registration_.get(),
+                          j_native_registration_->NewObject(
+                              "<init>", "(JI)V", PointerTojlong(this), 0)));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioRecordJni::~AudioRecordJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioRecordJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioRecordJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int32_t AudioRecordJni::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (frames_per_buffer < 0) {
+    direct_buffer_address_ = nullptr;
+    RTC_LOG(LS_ERROR) << "InitRecording failed";
+    return -1;
+  }
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * bytes_per_frame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!recording_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Recording can not start since InitRecording must succeed first";
+    return 0;
+  }
+  ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
+  if (!j_audio_record_->StartRecording()) {
+    RTC_LOG(LS_ERROR) << "StartRecording failed";
+    return -1;
+  }
+  recording_ = true;
+  return 0;
+}
+
+int32_t AudioRecordJni::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!j_audio_record_->StopRecording()) {
+    RTC_LOG(LS_ERROR) << "StopRecording failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  audio_device_buffer_->SetRecordingChannels(channels);
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
+                   << total_delay_in_milliseconds_;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
+  // TODO(henrika): possibly remove when no longer used by any client.
+  RTC_CHECK_NOTREACHED();
+}
+
+int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioRecordJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                      jobject obj,
+                                                      jobject byte_buffer,
+                                                      jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                                jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioRecordJni::DataIsRecorded(JNIEnv* env,
+                                            jobject obj,
+                                            jint length,
+                                            jlong nativeAudioRecord) {
+  webrtc::AudioRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioRecordJni*>(nativeAudioRecord);
+  this_object->OnDataIsRecorded(length);
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordThread'.
+void AudioRecordJni::OnDataIsRecorded(int length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
+                                          frames_per_buffer_);
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // `playDelayMs` parameter only. Components like the AEC only sees the sum
+  // of `playDelayMs` and `recDelayMs`, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
+    RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_record_jni.h

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio input support for Android using the Java
+// AudioRecord interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioRecord.java. This class is created and lives on a thread in
+// C++-land, but recorded audio buffers are delivered on a high-priority
+// thread managed by the Java class.
+//
+// The Java class makes use of AudioEffect features (mainly AEC) which are
+// first available in Jelly Bean. If it is instantiated running against earlier
+// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
+// separately instead.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioRecordJni {
+ public:
+  // Wraps the Java specific parts of the AudioRecordJni into one helper class.
+  class JavaAudioRecord {
+   public:
+    JavaAudioRecord(NativeRegistration* native_registration,
+                    std::unique_ptr<GlobalRef> audio_track);
+    ~JavaAudioRecord();
+
+    int InitRecording(int sample_rate, size_t channels);
+    bool StartRecording();
+    bool StopRecording();
+    bool EnableBuiltInAEC(bool enable);
+    bool EnableBuiltInNS(bool enable);
+
+   private:
+    std::unique_ptr<GlobalRef> audio_record_;
+    jmethodID init_recording_;
+    jmethodID start_recording_;
+    jmethodID stop_recording_;
+    jmethodID enable_built_in_aec_;
+    jmethodID enable_built_in_ns_;
+  };
+
+  explicit AudioRecordJni(AudioManager* audio_manager);
+  ~AudioRecordJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int32_t StartRecording();
+  int32_t StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  int32_t EnableBuiltInAEC(bool enable);
+  int32_t EnableBuiltInAGC(bool enable);
+  int32_t EnableBuiltInNS(bool enable);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
+  // This method will be called by the WebRtcAudioRecord constructor, i.e.,
+  // on the same thread that this object is created on.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioRecord);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioRecord object when
+  // recording has started. Each call indicates that there are `length` new
+  // bytes recorded in the memory area `direct_buffer_address_` and it is
+  // now time to send these to the consumer.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioRecordThread'.
+  static void JNICALL DataIsRecorded(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioRecord);
+  void OnDataIsRecorded(int length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnDataIsRecorded() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioRecordJni class.
+  std::unique_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
+
+  // Raw pointer to the audio manger.
+  const AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Delay estimate of the total round-trip delay (input + output).
+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
+  // possible values. See audio_common.h for details.
+  int total_delay_in_milliseconds_;
+
+  // Cached copy of address to direct audio buffer owned by `j_audio_record_`.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by `j_audio_record_`.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool recording_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_screen_record_jni.cc

@@ -0,0 +1,278 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_screen_record_jni.h"
+
+#include <string>
+#include <utility>
+
+#include "modules/audio_device/android/audio_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Scoped class which logs its time of life as a UMA statistic. It generates
+// a histogram which measures the time it takes for a method/scope to execute.
+class ScopedHistogramTimer {
+ public:
+  explicit ScopedHistogramTimer(const std::string& name)
+      : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
+  ~ScopedHistogramTimer() {
+    const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
+    RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
+    RTC_LOG(LS_INFO) << histogram_name_ << ": " << life_time_ms;
+  }
+
+ private:
+  const std::string histogram_name_;
+  int64_t start_time_ms_;
+};
+}  // namespace
+
+// AudioRecordJni::JavaAudioRecord implementation.
+AudioScreenRecordJni::JavaAudioRecord::JavaAudioRecord(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_record)
+    : audio_record_(std::move(audio_record)),
+      init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
+      start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
+      stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
+      enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+      enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
+
+int AudioScreenRecordJni::JavaAudioRecord::InitRecording(int sample_rate,
+                                                   size_t channels) {
+  return audio_record_->CallIntMethod(init_recording_,
+                                      static_cast<jint>(sample_rate),
+                                      static_cast<jint>(channels));
+}
+
+bool AudioScreenRecordJni::JavaAudioRecord::StartRecording() {
+  return audio_record_->CallBooleanMethod(start_recording_);
+}
+
+bool AudioScreenRecordJni::JavaAudioRecord::StopRecording() {
+  return audio_record_->CallBooleanMethod(stop_recording_);
+}
+
+bool AudioScreenRecordJni::JavaAudioRecord::EnableBuiltInAEC(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_aec_,
+                                          static_cast<jboolean>(enable));
+}
+
+bool AudioScreenRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
+  return audio_record_->CallBooleanMethod(enable_built_in_ns_,
+                                          static_cast<jboolean>(enable));
+}
+
+// AudioRecordJni implementation.
+AudioScreenRecordJni::AudioScreenRecordJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      total_delay_in_milliseconds_(0),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      recording_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioScreenRecordJni::CacheDirectBufferAddress)},
+      {"nativeDataIsRecorded", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioScreenRecordJni::DataIsRecorded)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioRecord", native_methods,
+      arraysize(native_methods));
+  j_audio_record_.reset(
+      new JavaAudioRecord(j_native_registration_.get(),
+                          j_native_registration_->NewObject(
+                              "<init>", "(JI)V", PointerTojlong(this), 1)));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioScreenRecordJni::~AudioScreenRecordJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioScreenRecordJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioScreenRecordJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int32_t AudioScreenRecordJni::InitRecording() {
+  RTC_LOG(LS_INFO) << "InitRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
+  int frames_per_buffer = j_audio_record_->InitRecording(
+      audio_parameters_.sample_rate(), audio_parameters_.channels());
+  if (frames_per_buffer < 0) {
+    direct_buffer_address_ = nullptr;
+    RTC_LOG(LS_ERROR) << "InitRecording failed";
+    return -1;
+  }
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
+               frames_per_buffer_ * bytes_per_frame);
+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioScreenRecordJni::StartRecording() {
+  RTC_LOG(LS_INFO) << "StartRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!recording_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Recording can not start since InitRecording must succeed first";
+    return 0;
+  }
+  ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
+  if (!j_audio_record_->StartRecording()) {
+    RTC_LOG(LS_ERROR) << "StartRecording failed";
+    return -1;
+  }
+  recording_ = true;
+  return 0;
+}
+
+int32_t AudioScreenRecordJni::StopRecording() {
+  RTC_LOG(LS_INFO) << "StopRecording";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  if (!j_audio_record_->StopRecording()) {
+    RTC_LOG(LS_ERROR) << "StopRecording failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+void AudioScreenRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  audio_device_buffer_->SetRecordingChannels(channels);
+  total_delay_in_milliseconds_ =
+      audio_manager_->GetDelayEstimateInMilliseconds();
+  RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
+  RTC_LOG(LS_INFO) << "total_delay_in_milliseconds: "
+                << total_delay_in_milliseconds_;
+}
+
+int32_t AudioScreenRecordJni::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInAEC(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInAEC(enable) ? 0 : -1;
+}
+
+int32_t AudioScreenRecordJni::EnableBuiltInAGC(bool enable) {
+  // TODO(henrika): possibly remove when no longer used by any client.
+  RTC_CHECK_NOTREACHED();
+}
+
+int32_t AudioScreenRecordJni::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << "EnableBuiltInNS(" << enable << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioScreenRecordJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                      jobject obj,
+                                                      jobject byte_buffer,
+                                                      jlong nativeAudioRecord) {
+  webrtc::AudioScreenRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioScreenRecordJni*>(nativeAudioRecord);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioScreenRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                                jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioScreenRecordJni::DataIsRecorded(JNIEnv* env,
+                                            jobject obj,
+                                            jint length,
+                                            jlong nativeAudioRecord) {
+  webrtc::AudioScreenRecordJni* this_object =
+      reinterpret_cast<webrtc::AudioScreenRecordJni*>(nativeAudioRecord);
+  this_object->OnDataIsRecorded(length);
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordThread'.
+void AudioScreenRecordJni::OnDataIsRecorded(int length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
+                                          frames_per_buffer_);
+  // We provide one (combined) fixed delay estimate for the APM and use the
+  // |playDelayMs| parameter only. Components like the AEC only sees the sum
+  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
+  audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_, 0);
+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
+    RTC_LOG(LS_INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_screen_record_jni.h

@@ -0,0 +1,167 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_SCREEN_RECORD_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_SCREEN_RECORD_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio input support for Android using the Java
+// AudioRecord interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioRecord.java. This class is created and lives on a thread in
+// C++-land, but recorded audio buffers are delivered on a high-priority
+// thread managed by the Java class.
+//
+// The Java class makes use of AudioEffect features (mainly AEC) which are
+// first available in Jelly Bean. If it is instantiated running against earlier
+// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
+// separately instead.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioScreenRecordJni {
+ public:
+  // Wraps the Java specific parts of the AudioRecordJni into one helper class.
+  class JavaAudioRecord {
+   public:
+    JavaAudioRecord(NativeRegistration* native_registration,
+                    std::unique_ptr<GlobalRef> audio_track);
+
+    int InitRecording(int sample_rate, size_t channels);
+    bool StartRecording();
+    bool StopRecording();
+    bool EnableBuiltInAEC(bool enable);
+    bool EnableBuiltInNS(bool enable);
+
+   private:
+    std::unique_ptr<GlobalRef> audio_record_;
+    jmethodID init_recording_;
+    jmethodID start_recording_;
+    jmethodID stop_recording_;
+    jmethodID enable_built_in_aec_;
+    jmethodID enable_built_in_ns_;
+  };
+
+  explicit AudioScreenRecordJni(AudioManager* audio_manager);
+  ~AudioScreenRecordJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int32_t StartRecording();
+  int32_t StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+  int32_t EnableBuiltInAEC(bool enable);
+  int32_t EnableBuiltInAGC(bool enable);
+  int32_t EnableBuiltInNS(bool enable);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
+  // is also stored in |direct_buffer_capacity_in_bytes_|.
+  // This method will be called by the WebRtcAudioRecord constructor, i.e.,
+  // on the same thread that this object is created on.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioRecord);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioRecord object when
+  // recording has started. Each call indicates that there are |length| new
+  // bytes recorded in the memory area |direct_buffer_address_| and it is
+  // now time to send these to the consumer.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioRecordThread'.
+  static void JNICALL DataIsRecorded(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioRecord);
+  void OnDataIsRecorded(int length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnDataIsRecorded() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioRecordJni class.
+  std::unique_ptr<AudioScreenRecordJni::JavaAudioRecord> j_audio_record_;
+
+  // Raw pointer to the audio manger.
+  const AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Delay estimate of the total round-trip delay (input + output).
+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
+  // possible values. See audio_common.h for details.
+  int total_delay_in_milliseconds_;
+
+  // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool recording_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_track_jni.cc

@@ -0,0 +1,296 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/audio_track_jni.h"
+
+#include <utility>
+
+#include "modules/audio_device/android/audio_manager.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+// AudioTrackJni::JavaAudioTrack implementation.
+AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
+    NativeRegistration* native_reg,
+    std::unique_ptr<GlobalRef> audio_track)
+    : audio_track_(std::move(audio_track)),
+      init_playout_(native_reg->GetMethodId("initPlayout", "(IID)I")),
+      start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
+      stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
+      set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
+      get_stream_max_volume_(
+          native_reg->GetMethodId("getStreamMaxVolume", "()I")),
+      get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")),
+      get_buffer_size_in_frames_(
+          native_reg->GetMethodId("getBufferSizeInFrames", "()I")) {}
+
+AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
+
+bool AudioTrackJni::JavaAudioTrack::InitPlayout(int sample_rate, int channels) {
+  double buffer_size_factor =
+      strtod(webrtc::field_trial::FindFullName(
+                 "WebRTC-AudioDevicePlayoutBufferSizeFactor")
+                 .c_str(),
+             nullptr);
+  if (buffer_size_factor == 0)
+    buffer_size_factor = 1.0;
+  int requested_buffer_size_bytes = audio_track_->CallIntMethod(
+      init_playout_, sample_rate, channels, buffer_size_factor);
+  // Update UMA histograms for both the requested and actual buffer size.
+  if (requested_buffer_size_bytes >= 0) {
+    // To avoid division by zero, we assume the sample rate is 48k if an invalid
+    // value is found.
+    sample_rate = sample_rate <= 0 ? 48000 : sample_rate;
+    // This calculation assumes that audio is mono.
+    const int requested_buffer_size_ms =
+        (requested_buffer_size_bytes * 1000) / (2 * sample_rate);
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeRequestedAudioBufferSizeMs",
+                         requested_buffer_size_ms, 0, 1000, 100);
+    int actual_buffer_size_frames =
+        audio_track_->CallIntMethod(get_buffer_size_in_frames_);
+    if (actual_buffer_size_frames >= 0) {
+      const int actual_buffer_size_ms =
+          actual_buffer_size_frames * 1000 / sample_rate;
+      RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AndroidNativeAudioBufferSizeMs",
+                           actual_buffer_size_ms, 0, 1000, 100);
+    }
+    return true;
+  }
+  return false;
+}
+
+bool AudioTrackJni::JavaAudioTrack::StartPlayout() {
+  return audio_track_->CallBooleanMethod(start_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::StopPlayout() {
+  return audio_track_->CallBooleanMethod(stop_playout_);
+}
+
+bool AudioTrackJni::JavaAudioTrack::SetStreamVolume(int volume) {
+  return audio_track_->CallBooleanMethod(set_stream_volume_, volume);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() {
+  return audio_track_->CallIntMethod(get_stream_max_volume_);
+}
+
+int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
+  return audio_track_->CallIntMethod(get_stream_volume_);
+}
+
+// TODO(henrika): possible extend usage of AudioManager and add it as member.
+AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
+    : j_environment_(JVM::GetInstance()->environment()),
+      audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      direct_buffer_address_(nullptr),
+      direct_buffer_capacity_in_bytes_(0),
+      frames_per_buffer_(0),
+      initialized_(false),
+      playing_(false),
+      audio_device_buffer_(nullptr) {
+  RTC_LOG(LS_INFO) << "ctor";
+  RTC_DCHECK(audio_parameters_.is_valid());
+  RTC_CHECK(j_environment_);
+  JNINativeMethod native_methods[] = {
+      {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
+       reinterpret_cast<void*>(
+           &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
+      {"nativeGetPlayoutData", "(IJ)V",
+       reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
+  j_native_registration_ = j_environment_->RegisterNatives(
+      "org/webrtc/voiceengine/WebRtcAudioTrack", native_methods,
+      arraysize(native_methods));
+  j_audio_track_.reset(
+      new JavaAudioTrack(j_native_registration_.get(),
+                         j_native_registration_->NewObject(
+                             "<init>", "(J)V", PointerTojlong(this))));
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the Java based audio thread.
+  thread_checker_java_.Detach();
+}
+
+AudioTrackJni::~AudioTrackJni() {
+  RTC_LOG(LS_INFO) << "dtor";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+}
+
+int32_t AudioTrackJni::Init() {
+  RTC_LOG(LS_INFO) << "Init";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return 0;
+}
+
+int32_t AudioTrackJni::Terminate() {
+  RTC_LOG(LS_INFO) << "Terminate";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopPlayout();
+  return 0;
+}
+
+int32_t AudioTrackJni::InitPlayout() {
+  RTC_LOG(LS_INFO) << "InitPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!j_audio_track_->InitPlayout(audio_parameters_.sample_rate(),
+                                   audio_parameters_.channels())) {
+    RTC_LOG(LS_ERROR) << "InitPlayout failed";
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioTrackJni::StartPlayout() {
+  RTC_LOG(LS_INFO) << "StartPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!playing_);
+  if (!initialized_) {
+    RTC_DLOG(LS_WARNING)
+        << "Playout can not start since InitPlayout must succeed first";
+    return 0;
+  }
+  if (!j_audio_track_->StartPlayout()) {
+    RTC_LOG(LS_ERROR) << "StartPlayout failed";
+    return -1;
+  }
+  playing_ = true;
+  return 0;
+}
+
+int32_t AudioTrackJni::StopPlayout() {
+  RTC_LOG(LS_INFO) << "StopPlayout";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !playing_ || j_audio_track_ == nullptr) {
+    return 0;
+  }
+  if (!j_audio_track_->StopPlayout()) {
+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
+    return -1;
+  }
+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
+  // next time StartRecording() is called since it will create a new Java
+  // thread.
+  thread_checker_java_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  direct_buffer_address_ = nullptr;
+  return 0;
+}
+
+int AudioTrackJni::SpeakerVolumeIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << "SetSpeakerVolume(" << volume << ")";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  return j_audio_track_->SetStreamVolume(volume) ? 0 : -1;
+}
+
+int AudioTrackJni::MaxSpeakerVolume(uint32_t& max_volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  max_volume = j_audio_track_->GetStreamMaxVolume();
+  return 0;
+}
+
+int AudioTrackJni::MinSpeakerVolume(uint32_t& min_volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  min_volume = 0;
+  return 0;
+}
+
+int AudioTrackJni::SpeakerVolume(uint32_t& volume) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  volume = j_audio_track_->GetStreamVolume();
+  RTC_LOG(LS_INFO) << "SpeakerVolume: " << volume;
+  return 0;
+}
+
+// TODO(henrika): possibly add stereo support.
+void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  RTC_LOG(LS_INFO) << "AttachAudioBuffer";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
+  audio_device_buffer_->SetPlayoutChannels(channels);
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env,
+                                                     jobject obj,
+                                                     jobject byte_buffer,
+                                                     jlong nativeAudioTrack) {
+  webrtc::AudioTrackJni* this_object =
+      reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
+  this_object->OnCacheDirectBufferAddress(env, byte_buffer);
+}
+
+void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
+                                               jobject byte_buffer) {
+  RTC_LOG(LS_INFO) << "OnCacheDirectBufferAddress";
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!direct_buffer_address_);
+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
+  RTC_LOG(LS_INFO) << "direct buffer capacity: " << capacity;
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
+  RTC_LOG(LS_INFO) << "frames_per_buffer: " << frames_per_buffer_;
+}
+
+JNI_FUNCTION_ALIGN
+void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env,
+                                           jobject obj,
+                                           jint length,
+                                           jlong nativeAudioTrack) {
+  webrtc::AudioTrackJni* this_object =
+      reinterpret_cast<webrtc::AudioTrackJni*>(nativeAudioTrack);
+  this_object->OnGetPlayoutData(static_cast<size_t>(length));
+}
+
+// This method is called on a high-priority thread from Java. The name of
+// the thread is 'AudioRecordTrack'.
+void AudioTrackJni::OnGetPlayoutData(size_t length) {
+  RTC_DCHECK(thread_checker_java_.IsCurrent());
+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
+  RTC_DCHECK_EQ(frames_per_buffer_, length / bytes_per_frame);
+  if (!audio_device_buffer_) {
+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
+    return;
+  }
+  // Pull decoded data (in 16-bit PCM format) from jitter buffer.
+  int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
+  if (samples <= 0) {
+    RTC_LOG(LS_ERROR) << "AudioDeviceBuffer::RequestPlayoutData failed";
+    return;
+  }
+  RTC_DCHECK_EQ(samples, frames_per_buffer_);
+  // Copy decoded data into common byte buffer to ensure that it can be
+  // written to the Java based audio track.
+  samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
+  RTC_DCHECK_EQ(length, bytes_per_frame * samples);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/audio_track_jni.h

@@ -0,0 +1,161 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
+
+#include <jni.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// Implements 16-bit mono PCM audio output support for Android using the Java
+// AudioTrack interface. Most of the work is done by its Java counterpart in
+// WebRtcAudioTrack.java. This class is created and lives on a thread in
+// C++-land, but decoded audio buffers are requested on a high-priority
+// thread managed by the Java class.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread.
+//
+// This class uses JvmThreadConnector to attach to a Java VM if needed
+// and detach when the object goes out of scope. Additional thread checking
+// guarantees that no other (possibly non attached) thread is used.
+class AudioTrackJni {
+ public:
+  // Wraps the Java specific parts of the AudioTrackJni into one helper class.
+  class JavaAudioTrack {
+   public:
+    JavaAudioTrack(NativeRegistration* native_registration,
+                   std::unique_ptr<GlobalRef> audio_track);
+    ~JavaAudioTrack();
+
+    bool InitPlayout(int sample_rate, int channels);
+    bool StartPlayout();
+    bool StopPlayout();
+    bool SetStreamVolume(int volume);
+    int GetStreamMaxVolume();
+    int GetStreamVolume();
+
+   private:
+    std::unique_ptr<GlobalRef> audio_track_;
+    jmethodID init_playout_;
+    jmethodID start_playout_;
+    jmethodID stop_playout_;
+    jmethodID set_stream_volume_;
+    jmethodID get_stream_max_volume_;
+    jmethodID get_stream_volume_;
+    jmethodID get_buffer_size_in_frames_;
+  };
+
+  explicit AudioTrackJni(AudioManager* audio_manager);
+  ~AudioTrackJni();
+
+  int32_t Init();
+  int32_t Terminate();
+
+  int32_t InitPlayout();
+  bool PlayoutIsInitialized() const { return initialized_; }
+
+  int32_t StartPlayout();
+  int32_t StopPlayout();
+  bool Playing() const { return playing_; }
+
+  int SpeakerVolumeIsAvailable(bool& available);
+  int SetSpeakerVolume(uint32_t volume);
+  int SpeakerVolume(uint32_t& volume) const;
+  int MaxSpeakerVolume(uint32_t& max_volume) const;
+  int MinSpeakerVolume(uint32_t& min_volume) const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  // Called from Java side so we can cache the address of the Java-manged
+  // `byte_buffer` in `direct_buffer_address_`. The size of the buffer
+  // is also stored in `direct_buffer_capacity_in_bytes_`.
+  // Called on the same thread as the creating thread.
+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
+                                               jobject obj,
+                                               jobject byte_buffer,
+                                               jlong nativeAudioTrack);
+  void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
+
+  // Called periodically by the Java based WebRtcAudioTrack object when
+  // playout has started. Each call indicates that `length` new bytes should
+  // be written to the memory area `direct_buffer_address_` for playout.
+  // This method is called on a high-priority thread from Java. The name of
+  // the thread is 'AudioTrackThread'.
+  static void JNICALL GetPlayoutData(JNIEnv* env,
+                                     jobject obj,
+                                     jint length,
+                                     jlong nativeAudioTrack);
+  void OnGetPlayoutData(size_t length);
+
+  // Stores thread ID in constructor.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to OnGetPlayoutData() from high-priority
+  // thread in Java. Detached during construction of this object.
+  SequenceChecker thread_checker_java_;
+
+  // Calls JavaVM::AttachCurrentThread() if this thread is not attached at
+  // construction.
+  // Also ensures that DetachCurrentThread() is called at destruction.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Wraps the JNI interface pointer and methods associated with it.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Contains factory method for creating the Java object.
+  std::unique_ptr<NativeRegistration> j_native_registration_;
+
+  // Wraps the Java specific parts of the AudioTrackJni class.
+  std::unique_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Cached copy of address to direct audio buffer owned by `j_audio_track_`.
+  void* direct_buffer_address_;
+
+  // Number of bytes in the direct audio buffer owned by `j_audio_track_`.
+  size_t direct_buffer_capacity_in_bytes_;
+
+  // Number of audio frames per audio buffer. Each audio frame corresponds to
+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
+  // frame contains 2 bytes (given that the Java layer only supports mono).
+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
+  size_t frames_per_buffer_;
+
+  bool initialized_;
+
+  bool playing_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
+  // and therefore outlives this object.
+  AudioDeviceBuffer* audio_device_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/build_info.cc

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/build_info.h"
+
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+BuildInfo::BuildInfo()
+    : j_environment_(JVM::GetInstance()->environment()),
+      j_build_info_(
+          JVM::GetInstance()->GetClass("org/webrtc/voiceengine/BuildInfo")) {}
+
+std::string BuildInfo::GetStringFromJava(const char* name) {
+  jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
+  jstring j_string =
+      static_cast<jstring>(j_build_info_.CallStaticObjectMethod(id));
+  return j_environment_->JavaToStdString(j_string);
+}
+
+std::string BuildInfo::GetDeviceModel() {
+  return GetStringFromJava("getDeviceModel");
+}
+
+std::string BuildInfo::GetBrand() {
+  return GetStringFromJava("getBrand");
+}
+
+std::string BuildInfo::GetDeviceManufacturer() {
+  return GetStringFromJava("getDeviceManufacturer");
+}
+
+std::string BuildInfo::GetAndroidBuildId() {
+  return GetStringFromJava("getAndroidBuildId");
+}
+
+std::string BuildInfo::GetBuildType() {
+  return GetStringFromJava("getBuildType");
+}
+
+std::string BuildInfo::GetBuildRelease() {
+  return GetStringFromJava("getBuildRelease");
+}
+
+SdkCode BuildInfo::GetSdkVersion() {
+  jmethodID id = j_build_info_.GetStaticMethodId("getSdkVersion", "()I");
+  jint j_version = j_build_info_.CallStaticIntMethod(id);
+  return static_cast<SdkCode>(j_version);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/build_info.h

@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
+
+#include <jni.h>
+
+#include <memory>
+#include <string>
+
+#include "modules/utility/include/jvm_android.h"
+
+namespace webrtc {
+
+// This enumeration maps to the values returned by BuildInfo::GetSdkVersion(),
+// indicating the Android release associated with a given SDK version.
+// See https://developer.android.com/guide/topics/manifest/uses-sdk-element.html
+// for details.
+enum SdkCode {
+  SDK_CODE_JELLY_BEAN = 16,      // Android 4.1
+  SDK_CODE_JELLY_BEAN_MR1 = 17,  // Android 4.2
+  SDK_CODE_JELLY_BEAN_MR2 = 18,  // Android 4.3
+  SDK_CODE_KITKAT = 19,          // Android 4.4
+  SDK_CODE_WATCH = 20,           // Android 4.4W
+  SDK_CODE_LOLLIPOP = 21,        // Android 5.0
+  SDK_CODE_LOLLIPOP_MR1 = 22,    // Android 5.1
+  SDK_CODE_MARSHMALLOW = 23,     // Android 6.0
+  SDK_CODE_N = 24,
+};
+
+// Utility class used to query the Java class (org/webrtc/voiceengine/BuildInfo)
+// for device and Android build information.
+// The calling thread is attached to the JVM at construction if needed and a
+// valid Java environment object is also created.
+// All Get methods must be called on the creating thread. If not, the code will
+// hit RTC_DCHECKs when calling JNIEnvironment::JavaToStdString().
+class BuildInfo {
+ public:
+  BuildInfo();
+  ~BuildInfo() {}
+
+  // End-user-visible name for the end product (e.g. "Nexus 6").
+  std::string GetDeviceModel();
+  // Consumer-visible brand (e.g. "google").
+  std::string GetBrand();
+  // Manufacturer of the product/hardware (e.g. "motorola").
+  std::string GetDeviceManufacturer();
+  // Android build ID (e.g. LMY47D).
+  std::string GetAndroidBuildId();
+  // The type of build (e.g. "user" or "eng").
+  std::string GetBuildType();
+  // The user-visible version string (e.g. "5.1").
+  std::string GetBuildRelease();
+  // The user-visible SDK version of the framework (e.g. 21). See SdkCode enum
+  // for translation.
+  SdkCode GetSdkVersion();
+
+ private:
+  // Helper method which calls a static getter method with `name` and returns
+  // a string from Java.
+  std::string GetStringFromJava(const char* name);
+
+  // Ensures that this class can access a valid JNI interface pointer even
+  // if the creating thread was not attached to the JVM.
+  JvmThreadConnector attach_thread_if_needed_;
+
+  // Provides access to the JNIEnv interface pointer and the JavaToStdString()
+  // method which is used to translate Java strings to std strings.
+  std::unique_ptr<JNIEnvironment> j_environment_;
+
+  // Holds the jclass object and provides access to CallStaticObjectMethod().
+  // Used by GetStringFromJava() during construction only.
+  JavaClass j_build_info_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/ensure_initialized.cc

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/ensure_initialized.h"
+
+#include <jni.h>
+#include <pthread.h>
+#include <stddef.h>
+
+#include "modules/utility/include/jvm_android.h"
+#include "rtc_base/checks.h"
+#include "sdk/android/src/jni/jvm.h"
+
+namespace webrtc {
+namespace audiodevicemodule {
+
+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
+
+void EnsureInitializedOnce() {
+  RTC_CHECK(::webrtc::jni::GetJVM() != nullptr);
+
+  JNIEnv* jni = ::webrtc::jni::AttachCurrentThreadIfNeeded();
+  JavaVM* jvm = NULL;
+  RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
+
+  // Initialize the Java environment (currently only used by the audio manager).
+  webrtc::JVM::Initialize(jvm);
+}
+
+void EnsureInitialized() {
+  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+}
+
+}  // namespace audiodevicemodule
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/ensure_initialized.h

@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+namespace webrtc {
+namespace audiodevicemodule {
+
+void EnsureInitialized();
+
+}  // namespace audiodevicemodule
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_common.cc

@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_common.h"
+
+#include <SLES/OpenSLES.h>
+
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Returns a string representation given an integer SL_RESULT_XXX code.
+// The mapping can be found in <SLES/OpenSLES.h>.
+const char* GetSLErrorString(size_t code) {
+  static const char* sl_error_strings[] = {
+      "SL_RESULT_SUCCESS",                 // 0
+      "SL_RESULT_PRECONDITIONS_VIOLATED",  // 1
+      "SL_RESULT_PARAMETER_INVALID",       // 2
+      "SL_RESULT_MEMORY_FAILURE",          // 3
+      "SL_RESULT_RESOURCE_ERROR",          // 4
+      "SL_RESULT_RESOURCE_LOST",           // 5
+      "SL_RESULT_IO_ERROR",                // 6
+      "SL_RESULT_BUFFER_INSUFFICIENT",     // 7
+      "SL_RESULT_CONTENT_CORRUPTED",       // 8
+      "SL_RESULT_CONTENT_UNSUPPORTED",     // 9
+      "SL_RESULT_CONTENT_NOT_FOUND",       // 10
+      "SL_RESULT_PERMISSION_DENIED",       // 11
+      "SL_RESULT_FEATURE_UNSUPPORTED",     // 12
+      "SL_RESULT_INTERNAL_ERROR",          // 13
+      "SL_RESULT_UNKNOWN_ERROR",           // 14
+      "SL_RESULT_OPERATION_ABORTED",       // 15
+      "SL_RESULT_CONTROL_LOST",            // 16
+  };
+
+  if (code >= arraysize(sl_error_strings)) {
+    return "SL_RESULT_UNKNOWN_ERROR";
+  }
+  return sl_error_strings[code];
+}
+
+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                        int sample_rate,
+                                        size_t bits_per_sample) {
+  RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
+  SLDataFormat_PCM format;
+  format.formatType = SL_DATAFORMAT_PCM;
+  format.numChannels = static_cast<SLuint32>(channels);
+  // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
+  switch (sample_rate) {
+    case 8000:
+      format.samplesPerSec = SL_SAMPLINGRATE_8;
+      break;
+    case 16000:
+      format.samplesPerSec = SL_SAMPLINGRATE_16;
+      break;
+    case 22050:
+      format.samplesPerSec = SL_SAMPLINGRATE_22_05;
+      break;
+    case 32000:
+      format.samplesPerSec = SL_SAMPLINGRATE_32;
+      break;
+    case 44100:
+      format.samplesPerSec = SL_SAMPLINGRATE_44_1;
+      break;
+    case 48000:
+      format.samplesPerSec = SL_SAMPLINGRATE_48;
+      break;
+    case 64000:
+      format.samplesPerSec = SL_SAMPLINGRATE_64;
+      break;
+    case 88200:
+      format.samplesPerSec = SL_SAMPLINGRATE_88_2;
+      break;
+    case 96000:
+      format.samplesPerSec = SL_SAMPLINGRATE_96;
+      break;
+    default:
+      RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
+      break;
+  }
+  format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+  format.endianness = SL_BYTEORDER_LITTLEENDIAN;
+  if (format.numChannels == 1) {
+    format.channelMask = SL_SPEAKER_FRONT_CENTER;
+  } else if (format.numChannels == 2) {
+    format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+  } else {
+    RTC_CHECK(false) << "Unsupported number of channels: "
+                     << format.numChannels;
+  }
+  return format;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_common.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
+
+#include <SLES/OpenSLES.h>
+#include <stddef.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Returns a string representation given an integer SL_RESULT_XXX code.
+// The mapping can be found in <SLES/OpenSLES.h>.
+const char* GetSLErrorString(size_t code);
+
+// Configures an SL_DATAFORMAT_PCM structure based on native audio parameters.
+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
+                                        int sample_rate,
+                                        size_t bits_per_sample);
+
+// Helper class for using SLObjectItf interfaces.
+template <typename SLType, typename SLDerefType>
+class ScopedSLObject {
+ public:
+  ScopedSLObject() : obj_(nullptr) {}
+
+  ~ScopedSLObject() { Reset(); }
+
+  SLType* Receive() {
+    RTC_DCHECK(!obj_);
+    return &obj_;
+  }
+
+  SLDerefType operator->() { return *obj_; }
+
+  SLType Get() const { return obj_; }
+
+  void Reset() {
+    if (obj_) {
+      (*obj_)->Destroy(obj_);
+      obj_ = nullptr;
+    }
+  }
+
+ private:
+  SLType obj_;
+};
+
+typedef ScopedSLObject<SLObjectItf, const SLObjectItf_*> ScopedSLObjectItf;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_player.cc

@@ -0,0 +1,434 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_player.h"
+
+#include <android/log.h>
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+
+#define TAG "OpenSLESPlayer"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+#define RETURN_ON_ERROR(op, ...)                          \
+  do {                                                    \
+    SLresult err = (op);                                  \
+    if (err != SL_RESULT_SUCCESS) {                       \
+      ALOGE("%s failed: %s", #op, GetSLErrorString(err)); \
+      return __VA_ARGS__;                                 \
+    }                                                     \
+  } while (0)
+
+namespace webrtc {
+
+OpenSLESPlayer::OpenSLESPlayer(AudioManager* audio_manager)
+    : audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
+      audio_device_buffer_(nullptr),
+      initialized_(false),
+      playing_(false),
+      buffer_index_(0),
+      engine_(nullptr),
+      player_(nullptr),
+      simple_buffer_queue_(nullptr),
+      volume_(nullptr),
+      last_play_time_(0) {
+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
+  // Use native audio output parameters provided by the audio manager and
+  // define the PCM format structure.
+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
+                                       audio_parameters_.sample_rate(),
+                                       audio_parameters_.bits_per_sample());
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the internal  audio thread.
+  thread_checker_opensles_.Detach();
+}
+
+OpenSLESPlayer::~OpenSLESPlayer() {
+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  DestroyAudioPlayer();
+  DestroyMix();
+  engine_ = nullptr;
+  RTC_DCHECK(!engine_);
+  RTC_DCHECK(!output_mix_.Get());
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+}
+
+int OpenSLESPlayer::Init() {
+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (audio_parameters_.channels() == 2) {
+    ALOGW("Stereo mode is enabled");
+  }
+  return 0;
+}
+
+int OpenSLESPlayer::Terminate() {
+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopPlayout();
+  return 0;
+}
+
+int OpenSLESPlayer::InitPlayout() {
+  ALOGD("InitPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!playing_);
+  if (!ObtainEngineInterface()) {
+    ALOGE("Failed to obtain SL Engine interface");
+    return -1;
+  }
+  CreateMix();
+  initialized_ = true;
+  buffer_index_ = 0;
+  return 0;
+}
+
+int OpenSLESPlayer::StartPlayout() {
+  ALOGD("StartPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!playing_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  CreateAudioPlayer();
+  // Fill up audio buffers to avoid initial glitch and to ensure that playback
+  // starts when mode is later changed to SL_PLAYSTATE_PLAYING.
+  // TODO(henrika): we can save some delay by only making one call to
+  // EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
+  last_play_time_ = rtc::Time();
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    EnqueuePlayoutData(true);
+  }
+  // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
+  // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
+  // state, adding buffers will implicitly start playback.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
+  playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
+  RTC_DCHECK(playing_);
+  return 0;
+}
+
+int OpenSLESPlayer::StopPlayout() {
+  ALOGD("StopPlayout[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !playing_) {
+    return 0;
+  }
+  // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
+  // Clear the buffer queue to flush out any remaining data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
+#if RTC_DCHECK_IS_ON
+  // Verify that the buffer queue is in fact cleared as it should.
+  SLAndroidSimpleBufferQueueState buffer_queue_state;
+  (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
+  RTC_DCHECK_EQ(0, buffer_queue_state.count);
+  RTC_DCHECK_EQ(0, buffer_queue_state.index);
+#endif
+  // The number of lower latency audio players is limited, hence we create the
+  // audio player in Start() and destroy it in Stop().
+  DestroyAudioPlayer();
+  thread_checker_opensles_.Detach();
+  initialized_ = false;
+  playing_ = false;
+  return 0;
+}
+
+int OpenSLESPlayer::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int OpenSLESPlayer::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int OpenSLESPlayer::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_device_buffer_ = audioBuffer;
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%zu)", channels);
+  audio_device_buffer_->SetPlayoutChannels(channels);
+  RTC_CHECK(audio_device_buffer_);
+  AllocateDataBuffers();
+}
+
+void OpenSLESPlayer::AllocateDataBuffers() {
+  ALOGD("AllocateDataBuffers");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to ask for any number
+  // of samples (and not only multiple of 10ms) to match the native OpenSL ES
+  // buffer size. The native buffer size corresponds to the
+  // PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
+  // frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
+  // recommended to construct audio buffers so that they contain an exact
+  // multiple of this number. If so, callbacks will occur at regular intervals,
+  // which reduces jitter.
+  const size_t buffer_size_in_samples =
+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
+  ALOGD("native buffer size: %zu", buffer_size_in_samples);
+  ALOGD("native buffer size in ms: %.2f",
+        audio_parameters_.GetBufferSizeInMilliseconds());
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+  // Allocated memory for audio buffers.
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);
+  }
+}
+
+bool OpenSLESPlayer::ObtainEngineInterface() {
+  ALOGD("ObtainEngineInterface");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (engine_)
+    return true;
+  // Get access to (or create if not already existing) the global OpenSL Engine
+  // object.
+  SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
+  if (engine_object == nullptr) {
+    ALOGE("Failed to access the global OpenSL engine");
+    return false;
+  }
+  // Get the SL Engine Interface which is implicit.
+  RETURN_ON_ERROR(
+      (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine_),
+      false);
+  return true;
+}
+
+bool OpenSLESPlayer::CreateMix() {
+  ALOGD("CreateMix");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(engine_);
+  if (output_mix_.Get())
+    return true;
+
+  // Create the ouput mix on the engine object. No interfaces will be used.
+  RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
+                                              nullptr, nullptr),
+                  false);
+  RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
+                  false);
+  return true;
+}
+
+void OpenSLESPlayer::DestroyMix() {
+  ALOGD("DestroyMix");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!output_mix_.Get())
+    return;
+  output_mix_.Reset();
+}
+
+bool OpenSLESPlayer::CreateAudioPlayer() {
+  ALOGD("CreateAudioPlayer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(output_mix_.Get());
+  if (player_object_.Get())
+    return true;
+  RTC_DCHECK(!player_);
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_DCHECK(!volume_);
+
+  // source: Android Simple Buffer Queue Data Locator is source.
+  SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
+  SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
+
+  // sink: OutputMix-based data is sink.
+  SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
+                                                output_mix_.Get()};
+  SLDataSink audio_sink = {&locator_output_mix, nullptr};
+
+  // Define interfaces that we indend to use and realize.
+  const SLInterfaceID interface_ids[] = {SL_IID_ANDROIDCONFIGURATION,
+                                         SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
+                                          SL_BOOLEAN_TRUE};
+
+  // Create the audio player on the engine interface.
+  RETURN_ON_ERROR(
+      (*engine_)->CreateAudioPlayer(
+          engine_, player_object_.Receive(), &audio_source, &audio_sink,
+          arraysize(interface_ids), interface_ids, interface_required),
+      false);
+
+  // Use the Android configuration interface to set platform-specific
+  // parameters. Should be done before player is realized.
+  SLAndroidConfigurationItf player_config;
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(),
+                                   SL_IID_ANDROIDCONFIGURATION, &player_config),
+      false);
+  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
+  // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
+  RETURN_ON_ERROR(
+      (*player_config)
+          ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
+                             &stream_type, sizeof(SLint32)),
+      false);
+
+  // Realize the audio player object after configuration has been set.
+  RETURN_ON_ERROR(
+      player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
+
+  // Get the SLPlayItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY, &player_),
+      false);
+
+  // Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
+  RETURN_ON_ERROR(
+      player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE,
+                                   &simple_buffer_queue_),
+      false);
+
+  // Register callback method for the Android Simple Buffer Queue interface.
+  // This method will be called when the native audio layer needs audio data.
+  RETURN_ON_ERROR((*simple_buffer_queue_)
+                      ->RegisterCallback(simple_buffer_queue_,
+                                         SimpleBufferQueueCallback, this),
+                  false);
+
+  // Get the SLVolumeItf interface on the audio player.
+  RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
+                                               SL_IID_VOLUME, &volume_),
+                  false);
+
+  // TODO(henrika): might not be required to set volume to max here since it
+  // seems to be default on most devices. Might be required for unit tests.
+  // RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
+
+  return true;
+}
+
+void OpenSLESPlayer::DestroyAudioPlayer() {
+  ALOGD("DestroyAudioPlayer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!player_object_.Get())
+    return;
+  (*simple_buffer_queue_)
+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
+  player_object_.Reset();
+  player_ = nullptr;
+  simple_buffer_queue_ = nullptr;
+  volume_ = nullptr;
+}
+
+// static
+void OpenSLESPlayer::SimpleBufferQueueCallback(
+    SLAndroidSimpleBufferQueueItf caller,
+    void* context) {
+  OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
+  stream->FillBufferQueue();
+}
+
+void OpenSLESPlayer::FillBufferQueue() {
+  RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+  SLuint32 state = GetPlayState();
+  if (state != SL_PLAYSTATE_PLAYING) {
+    ALOGW("Buffer callback in non-playing state!");
+    return;
+  }
+  EnqueuePlayoutData(false);
+}
+
+void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_play_time_;
+  if (diff > 150) {
+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
+  }
+  last_play_time_ = current_time;
+  SLint8* audio_ptr8 =
+      reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get());
+  if (silence) {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    // Avoid acquiring real audio data from WebRTC and fill the buffer with
+    // zeros instead. Used to prime the buffer with silence and to avoid asking
+    // for audio data from two different threads.
+    memset(audio_ptr8, 0, audio_parameters_.GetBytesPerBuffer());
+  } else {
+    RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+    // Read audio data from the WebRTC source using the FineAudioBuffer object
+    // to adjust for differences in buffer size between WebRTC (10ms) and native
+    // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
+    // delay estimation.
+    fine_audio_buffer_->GetPlayoutData(
+        rtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
+                                audio_parameters_.frames_per_buffer() *
+                                    audio_parameters_.channels()),
+        25);
+  }
+  // Enqueue the decoded audio buffer for playback.
+  SLresult err = (*simple_buffer_queue_)
+                     ->Enqueue(simple_buffer_queue_, audio_ptr8,
+                               audio_parameters_.GetBytesPerBuffer());
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("Enqueue failed: %d", err);
+  }
+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
+}
+
+SLuint32 OpenSLESPlayer::GetPlayState() const {
+  RTC_DCHECK(player_);
+  SLuint32 state;
+  SLresult err = (*player_)->GetPlayState(player_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetPlayState failed: %d", err);
+  }
+  return state;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_player.h

@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+// Implements 16-bit mono PCM audio output support for Android using the
+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Decoded audio
+// buffers are requested on a dedicated internal thread managed by the OpenSL
+// ES layer.
+//
+// The existing design forces the user to call InitPlayout() after Stoplayout()
+// to be able to call StartPlayout() again. This is inline with how the Java-
+// based implementation works.
+//
+// OpenSL ES is a native C API which have no Dalvik-related overhead such as
+// garbage collection pauses and it supports reduced audio output latency.
+// If the device doesn't claim this feature but supports API level 9 (Android
+// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
+// the output latency may be higher.
+class OpenSLESPlayer {
+ public:
+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
+  // size and sample rate must be compatible with the device's native output
+  // configuration provided via the audio manager at construction.
+  // TODO(henrika): perhaps set this value dynamically based on OS version.
+  static const int kNumOfOpenSLESBuffers = 2;
+
+  explicit OpenSLESPlayer(AudioManager* audio_manager);
+  ~OpenSLESPlayer();
+
+  int Init();
+  int Terminate();
+
+  int InitPlayout();
+  bool PlayoutIsInitialized() const { return initialized_; }
+
+  int StartPlayout();
+  int StopPlayout();
+  bool Playing() const { return playing_; }
+
+  int SpeakerVolumeIsAvailable(bool& available);
+  int SetSpeakerVolume(uint32_t volume);
+  int SpeakerVolume(uint32_t& volume) const;
+  int MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int MinSpeakerVolume(uint32_t& minVolume) const;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
+
+ private:
+  // These callback methods are called when data is required for playout.
+  // They are both called from an internal "OpenSL ES thread" which is not
+  // attached to the Dalvik VM.
+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
+                                        void* context);
+  void FillBufferQueue();
+  // Reads audio data in PCM format using the AudioDeviceBuffer.
+  // Can be called both on the main thread (during Start()) and from the
+  // internal audio thread while output streaming is active.
+  // If the `silence` flag is set, the audio is filled with zeros instead of
+  // asking the WebRTC layer for real audio data. This procedure is also known
+  // as audio priming.
+  void EnqueuePlayoutData(bool silence);
+
+  // Allocate memory for audio buffers which will be used to render audio
+  // via the SLAndroidSimpleBufferQueueItf interface.
+  void AllocateDataBuffers();
+
+  // Obtaines the SL Engine Interface from the existing global Engine object.
+  // The interface exposes creation methods of all the OpenSL ES object types.
+  // This method defines the `engine_` member variable.
+  bool ObtainEngineInterface();
+
+  // Creates/destroys the output mix object.
+  bool CreateMix();
+  void DestroyMix();
+
+  // Creates/destroys the audio player and the simple-buffer object.
+  // Also creates the volume object.
+  bool CreateAudioPlayer();
+  void DestroyAudioPlayer();
+
+  SLuint32 GetPlayState() const;
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
+  // non-application thread which is not attached to the Dalvik JVM.
+  // Detached during construction of this object.
+  SequenceChecker thread_checker_opensles_;
+
+  // Raw pointer to the audio manager injected at construction. Used to cache
+  // audio parameters and to access the global SL engine object needed by the
+  // ObtainEngineInterface() method. The audio manager outlives any instance of
+  // this class.
+  AudioManager* audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  bool initialized_;
+  bool playing_;
+
+  // PCM-type format definition.
+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
+  // 32-bit float representation is needed.
+  SLDataFormat_PCM pcm_format_;
+
+  // Queue of audio buffers to be used by the player object for rendering
+  // audio.
+  std::unique_ptr<SLint16[]> audio_buffers_[kNumOfOpenSLESBuffers];
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples.
+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 192
+  // in each callback (one every 4th ms). This class can then ask for 192 and
+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
+  // second callback and also cache non-utilized audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
+  int buffer_index_;
+
+  // This interface exposes creation methods for all the OpenSL ES object types.
+  // It is the OpenSL ES API entry point.
+  SLEngineItf engine_;
+
+  // Output mix object to be used by the player object.
+  webrtc::ScopedSLObjectItf output_mix_;
+
+  // The audio player media object plays out audio to the speakers. It also
+  // supports volume control.
+  webrtc::ScopedSLObjectItf player_object_;
+
+  // This interface is supported on the audio player and it controls the state
+  // of the audio player.
+  SLPlayItf player_;
+
+  // The Android Simple Buffer Queue interface is supported on the audio player
+  // and it provides methods to send audio data from the source to the audio
+  // player for rendering.
+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
+
+  // This interface exposes controls for manipulating the object’s audio volume
+  // properties. This interface is supported on the Audio Player object.
+  SLVolumeItf volume_;
+
+  // Last time the OpenSL ES layer asked for audio data to play out.
+  uint32_t last_play_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_recorder.cc

@@ -0,0 +1,431 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/android/opensles_recorder.h"
+
+#include <android/log.h>
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+
+#define TAG "OpenSLESRecorder"
+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
+
+#define LOG_ON_ERROR(op)                                    \
+  [](SLresult err) {                                        \
+    if (err != SL_RESULT_SUCCESS) {                         \
+      ALOGE("%s:%d %s failed: %s", __FILE__, __LINE__, #op, \
+            GetSLErrorString(err));                         \
+      return true;                                          \
+    }                                                       \
+    return false;                                           \
+  }(op)
+
+namespace webrtc {
+
+OpenSLESRecorder::OpenSLESRecorder(AudioManager* audio_manager)
+    : audio_manager_(audio_manager),
+      audio_parameters_(audio_manager->GetRecordAudioParameters()),
+      audio_device_buffer_(nullptr),
+      initialized_(false),
+      recording_(false),
+      engine_(nullptr),
+      recorder_(nullptr),
+      simple_buffer_queue_(nullptr),
+      buffer_index_(0),
+      last_rec_time_(0) {
+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
+  // Detach from this thread since we want to use the checker to verify calls
+  // from the internal  audio thread.
+  thread_checker_opensles_.Detach();
+  // Use native audio output parameters provided by the audio manager and
+  // define the PCM format structure.
+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
+                                       audio_parameters_.sample_rate(),
+                                       audio_parameters_.bits_per_sample());
+}
+
+OpenSLESRecorder::~OpenSLESRecorder() {
+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  Terminate();
+  DestroyAudioRecorder();
+  engine_ = nullptr;
+  RTC_DCHECK(!engine_);
+  RTC_DCHECK(!recorder_);
+  RTC_DCHECK(!simple_buffer_queue_);
+}
+
+int OpenSLESRecorder::Init() {
+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (audio_parameters_.channels() == 2) {
+    ALOGD("Stereo mode is enabled");
+  }
+  return 0;
+}
+
+int OpenSLESRecorder::Terminate() {
+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  StopRecording();
+  return 0;
+}
+
+int OpenSLESRecorder::InitRecording() {
+  ALOGD("InitRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!initialized_);
+  RTC_DCHECK(!recording_);
+  if (!ObtainEngineInterface()) {
+    ALOGE("Failed to obtain SL Engine interface");
+    return -1;
+  }
+  CreateAudioRecorder();
+  initialized_ = true;
+  buffer_index_ = 0;
+  return 0;
+}
+
+int OpenSLESRecorder::StartRecording() {
+  ALOGD("StartRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!recording_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetRecord();
+  }
+  // Add buffers to the queue before changing state to SL_RECORDSTATE_RECORDING
+  // to ensure that recording starts as soon as the state is modified. On some
+  // devices, SLAndroidSimpleBufferQueue::Clear() used in Stop() does not flush
+  // the buffers as intended and we therefore check the number of buffers
+  // already queued first. Enqueue() can return SL_RESULT_BUFFER_INSUFFICIENT
+  // otherwise.
+  int num_buffers_in_queue = GetBufferCount();
+  for (int i = 0; i < kNumOfOpenSLESBuffers - num_buffers_in_queue; ++i) {
+    if (!EnqueueAudioBuffer()) {
+      recording_ = false;
+      return -1;
+    }
+  }
+  num_buffers_in_queue = GetBufferCount();
+  RTC_DCHECK_EQ(num_buffers_in_queue, kNumOfOpenSLESBuffers);
+  LogBufferState();
+  // Start audio recording by changing the state to SL_RECORDSTATE_RECORDING.
+  // Given that buffers are already enqueued, recording should start at once.
+  // The macro returns -1 if recording fails to start.
+  last_rec_time_ = rtc::Time();
+  if (LOG_ON_ERROR(
+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_RECORDING))) {
+    return -1;
+  }
+  recording_ = (GetRecordState() == SL_RECORDSTATE_RECORDING);
+  RTC_DCHECK(recording_);
+  return 0;
+}
+
+int OpenSLESRecorder::StopRecording() {
+  ALOGD("StopRecording[tid=%d]", rtc::CurrentThreadId());
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!initialized_ || !recording_) {
+    return 0;
+  }
+  // Stop recording by setting the record state to SL_RECORDSTATE_STOPPED.
+  if (LOG_ON_ERROR(
+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_STOPPED))) {
+    return -1;
+  }
+  // Clear the buffer queue to get rid of old data when resuming recording.
+  if (LOG_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_))) {
+    return -1;
+  }
+  thread_checker_opensles_.Detach();
+  initialized_ = false;
+  recording_ = false;
+  return 0;
+}
+
+void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
+  ALOGD("AttachAudioBuffer");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_CHECK(audio_buffer);
+  audio_device_buffer_ = audio_buffer;
+  // Ensure that the audio device buffer is informed about the native sample
+  // rate used on the recording side.
+  const int sample_rate_hz = audio_parameters_.sample_rate();
+  ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
+  // Ensure that the audio device buffer is informed about the number of
+  // channels preferred by the OS on the recording side.
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%zu)", channels);
+  audio_device_buffer_->SetRecordingChannels(channels);
+  // Allocated memory for internal data buffers given existing audio parameters.
+  AllocateDataBuffers();
+}
+
+int OpenSLESRecorder::EnableBuiltInAEC(bool enable) {
+  ALOGD("EnableBuiltInAEC(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+int OpenSLESRecorder::EnableBuiltInAGC(bool enable) {
+  ALOGD("EnableBuiltInAGC(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+int OpenSLESRecorder::EnableBuiltInNS(bool enable) {
+  ALOGD("EnableBuiltInNS(%d)", enable);
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  ALOGE("Not implemented");
+  return 0;
+}
+
+bool OpenSLESRecorder::ObtainEngineInterface() {
+  ALOGD("ObtainEngineInterface");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (engine_)
+    return true;
+  // Get access to (or create if not already existing) the global OpenSL Engine
+  // object.
+  SLObjectItf engine_object = audio_manager_->GetOpenSLEngine();
+  if (engine_object == nullptr) {
+    ALOGE("Failed to access the global OpenSL engine");
+    return false;
+  }
+  // Get the SL Engine Interface which is implicit.
+  if (LOG_ON_ERROR(
+          (*engine_object)
+              ->GetInterface(engine_object, SL_IID_ENGINE, &engine_))) {
+    return false;
+  }
+  return true;
+}
+
+bool OpenSLESRecorder::CreateAudioRecorder() {
+  ALOGD("CreateAudioRecorder");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (recorder_object_.Get())
+    return true;
+  RTC_DCHECK(!recorder_);
+  RTC_DCHECK(!simple_buffer_queue_);
+
+  // Audio source configuration.
+  SLDataLocator_IODevice mic_locator = {SL_DATALOCATOR_IODEVICE,
+                                        SL_IODEVICE_AUDIOINPUT,
+                                        SL_DEFAULTDEVICEID_AUDIOINPUT, NULL};
+  SLDataSource audio_source = {&mic_locator, NULL};
+
+  // Audio sink configuration.
+  SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {
+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
+  SLDataSink audio_sink = {&buffer_queue, &pcm_format_};
+
+  // Create the audio recorder object (requires the RECORD_AUDIO permission).
+  // Do not realize the recorder yet. Set the configuration first.
+  const SLInterfaceID interface_id[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+                                        SL_IID_ANDROIDCONFIGURATION};
+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
+  if (LOG_ON_ERROR((*engine_)->CreateAudioRecorder(
+          engine_, recorder_object_.Receive(), &audio_source, &audio_sink,
+          arraysize(interface_id), interface_id, interface_required))) {
+    return false;
+  }
+
+  // Configure the audio recorder (before it is realized).
+  SLAndroidConfigurationItf recorder_config;
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(recorder_object_.Get(),
+                                                   SL_IID_ANDROIDCONFIGURATION,
+                                                   &recorder_config)))) {
+    return false;
+  }
+
+  // Uses the default microphone tuned for audio communication.
+  // Note that, SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION leads to a fast
+  // track but also excludes usage of required effects like AEC, AGC and NS.
+  // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION
+  SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
+  if (LOG_ON_ERROR(((*recorder_config)
+                        ->SetConfiguration(recorder_config,
+                                           SL_ANDROID_KEY_RECORDING_PRESET,
+                                           &stream_type, sizeof(SLint32))))) {
+    return false;
+  }
+
+  // The audio recorder can now be realized (in synchronous mode).
+  if (LOG_ON_ERROR((recorder_object_->Realize(recorder_object_.Get(),
+                                              SL_BOOLEAN_FALSE)))) {
+    return false;
+  }
+
+  // Get the implicit recorder interface (SL_IID_RECORD).
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
+          recorder_object_.Get(), SL_IID_RECORD, &recorder_)))) {
+    return false;
+  }
+
+  // Get the simple buffer queue interface (SL_IID_ANDROIDSIMPLEBUFFERQUEUE).
+  // It was explicitly requested.
+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
+          recorder_object_.Get(), SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+          &simple_buffer_queue_)))) {
+    return false;
+  }
+
+  // Register the input callback for the simple buffer queue.
+  // This callback will be called when receiving new data from the device.
+  if (LOG_ON_ERROR(((*simple_buffer_queue_)
+                        ->RegisterCallback(simple_buffer_queue_,
+                                           SimpleBufferQueueCallback, this)))) {
+    return false;
+  }
+  return true;
+}
+
+void OpenSLESRecorder::DestroyAudioRecorder() {
+  ALOGD("DestroyAudioRecorder");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (!recorder_object_.Get())
+    return;
+  (*simple_buffer_queue_)
+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
+  recorder_object_.Reset();
+  recorder_ = nullptr;
+  simple_buffer_queue_ = nullptr;
+}
+
+void OpenSLESRecorder::SimpleBufferQueueCallback(
+    SLAndroidSimpleBufferQueueItf buffer_queue,
+    void* context) {
+  OpenSLESRecorder* stream = static_cast<OpenSLESRecorder*>(context);
+  stream->ReadBufferQueue();
+}
+
+void OpenSLESRecorder::AllocateDataBuffers() {
+  ALOGD("AllocateDataBuffers");
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DCHECK(!simple_buffer_queue_);
+  RTC_CHECK(audio_device_buffer_);
+  // Create a modified audio buffer class which allows us to deliver any number
+  // of samples (and not only multiple of 10ms) to match the native audio unit
+  // buffer size.
+  ALOGD("frames per native buffer: %zu", audio_parameters_.frames_per_buffer());
+  ALOGD("frames per 10ms buffer: %zu",
+        audio_parameters_.frames_per_10ms_buffer());
+  ALOGD("bytes per native buffer: %zu", audio_parameters_.GetBytesPerBuffer());
+  ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
+  RTC_DCHECK(audio_device_buffer_);
+  fine_audio_buffer_ = std::make_unique<FineAudioBuffer>(audio_device_buffer_);
+  // Allocate queue of audio buffers that stores recorded audio samples.
+  const int buffer_size_samples =
+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
+  audio_buffers_.reset(new std::unique_ptr<SLint16[]>[kNumOfOpenSLESBuffers]);
+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
+    audio_buffers_[i].reset(new SLint16[buffer_size_samples]);
+  }
+}
+
+void OpenSLESRecorder::ReadBufferQueue() {
+  RTC_DCHECK(thread_checker_opensles_.IsCurrent());
+  SLuint32 state = GetRecordState();
+  if (state != SL_RECORDSTATE_RECORDING) {
+    ALOGW("Buffer callback in non-recording state!");
+    return;
+  }
+  // Check delta time between two successive callbacks and provide a warning
+  // if it becomes very large.
+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
+  const uint32_t current_time = rtc::Time();
+  const uint32_t diff = current_time - last_rec_time_;
+  if (diff > 150) {
+    ALOGW("Bad OpenSL ES record timing, dT=%u [ms]", diff);
+  }
+  last_rec_time_ = current_time;
+  // Send recorded audio data to the WebRTC sink.
+  // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
+  // since there is no support to turn off built-in EC in combination with
+  // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
+  // these estimates) will never be active.
+  fine_audio_buffer_->DeliverRecordedData(
+      rtc::ArrayView<const int16_t>(
+          audio_buffers_[buffer_index_].get(),
+          audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
+      25);
+  // Enqueue the utilized audio buffer and use if for recording again.
+  EnqueueAudioBuffer();
+}
+
+bool OpenSLESRecorder::EnqueueAudioBuffer() {
+  SLresult err =
+      (*simple_buffer_queue_)
+          ->Enqueue(
+              simple_buffer_queue_,
+              reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get()),
+              audio_parameters_.GetBytesPerBuffer());
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("Enqueue failed: %s", GetSLErrorString(err));
+    return false;
+  }
+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
+  return true;
+}
+
+SLuint32 OpenSLESRecorder::GetRecordState() const {
+  RTC_DCHECK(recorder_);
+  SLuint32 state;
+  SLresult err = (*recorder_)->GetRecordState(recorder_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetRecordState failed: %s", GetSLErrorString(err));
+  }
+  return state;
+}
+
+SLAndroidSimpleBufferQueueState OpenSLESRecorder::GetBufferQueueState() const {
+  RTC_DCHECK(simple_buffer_queue_);
+  // state.count: Number of buffers currently in the queue.
+  // state.index: Index of the currently filling buffer. This is a linear index
+  // that keeps a cumulative count of the number of buffers recorded.
+  SLAndroidSimpleBufferQueueState state;
+  SLresult err =
+      (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &state);
+  if (SL_RESULT_SUCCESS != err) {
+    ALOGE("GetState failed: %s", GetSLErrorString(err));
+  }
+  return state;
+}
+
+void OpenSLESRecorder::LogBufferState() const {
+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
+  ALOGD("state.count:%d state.index:%d", state.count, state.index);
+}
+
+SLuint32 OpenSLESRecorder::GetBufferCount() {
+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
+  return state.count;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/android/opensles_recorder.h

@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
+#define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <SLES/OpenSLES_AndroidConfiguration.h>
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/android/audio_common.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/opensles_common.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+// Implements 16-bit mono PCM audio input support for Android using the
+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All public methods must also be called on the same thread. A thread checker
+// will RTC_DCHECK if any method is called on an invalid thread. Recorded audio
+// buffers are provided on a dedicated internal thread managed by the OpenSL
+// ES layer.
+//
+// The existing design forces the user to call InitRecording() after
+// StopRecording() to be able to call StartRecording() again. This is inline
+// with how the Java-based implementation works.
+//
+// As of API level 21, lower latency audio input is supported on select devices.
+// To take advantage of this feature, first confirm that lower latency output is
+// available. The capability for lower latency output is a prerequisite for the
+// lower latency input feature. Then, create an AudioRecorder with the same
+// sample rate and buffer size as would be used for output. OpenSL ES interfaces
+// for input effects preclude the lower latency path.
+// See https://developer.android.com/ndk/guides/audio/opensl-prog-notes.html
+// for more details.
+class OpenSLESRecorder {
+ public:
+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
+  // size and sample rate must be compatible with the device's native input
+  // configuration provided via the audio manager at construction.
+  // TODO(henrika): perhaps set this value dynamically based on OS version.
+  static const int kNumOfOpenSLESBuffers = 2;
+
+  explicit OpenSLESRecorder(AudioManager* audio_manager);
+  ~OpenSLESRecorder();
+
+  int Init();
+  int Terminate();
+
+  int InitRecording();
+  bool RecordingIsInitialized() const { return initialized_; }
+
+  int StartRecording();
+  int StopRecording();
+  bool Recording() const { return recording_; }
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer);
+
+  // TODO(henrika): add support using OpenSL ES APIs when available.
+  int EnableBuiltInAEC(bool enable);
+  int EnableBuiltInAGC(bool enable);
+  int EnableBuiltInNS(bool enable);
+
+ private:
+  // Obtaines the SL Engine Interface from the existing global Engine object.
+  // The interface exposes creation methods of all the OpenSL ES object types.
+  // This method defines the `engine_` member variable.
+  bool ObtainEngineInterface();
+
+  // Creates/destroys the audio recorder and the simple-buffer queue object.
+  bool CreateAudioRecorder();
+  void DestroyAudioRecorder();
+
+  // Allocate memory for audio buffers which will be used to capture audio
+  // via the SLAndroidSimpleBufferQueueItf interface.
+  void AllocateDataBuffers();
+
+  // These callback methods are called when data has been written to the input
+  // buffer queue. They are both called from an internal "OpenSL ES thread"
+  // which is not attached to the Dalvik VM.
+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
+                                        void* context);
+  void ReadBufferQueue();
+
+  // Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
+  // called both on the main thread (but before recording has started) and from
+  // the internal audio thread while input streaming is active. It uses
+  // `simple_buffer_queue_` but no lock is needed since the initial calls from
+  // the main thread and the native callback thread are mutually exclusive.
+  bool EnqueueAudioBuffer();
+
+  // Returns the current recorder state.
+  SLuint32 GetRecordState() const;
+
+  // Returns the current buffer queue state.
+  SLAndroidSimpleBufferQueueState GetBufferQueueState() const;
+
+  // Number of buffers currently in the queue.
+  SLuint32 GetBufferCount();
+
+  // Prints a log message of the current queue state. Can be used for debugging
+  // purposes.
+  void LogBufferState() const;
+
+  // Ensures that methods are called from the same thread as this object is
+  // created on.
+  SequenceChecker thread_checker_;
+
+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
+  // non-application thread which is not attached to the Dalvik JVM.
+  // Detached during construction of this object.
+  SequenceChecker thread_checker_opensles_;
+
+  // Raw pointer to the audio manager injected at construction. Used to cache
+  // audio parameters and to access the global SL engine object needed by the
+  // ObtainEngineInterface() method. The audio manager outlives any instance of
+  // this class.
+  AudioManager* const audio_manager_;
+
+  // Contains audio parameters provided to this class at construction by the
+  // AudioManager.
+  const AudioParameters audio_parameters_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  // PCM-type format definition.
+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
+  // 32-bit float representation is needed.
+  SLDataFormat_PCM pcm_format_;
+
+  bool initialized_;
+  bool recording_;
+
+  // This interface exposes creation methods for all the OpenSL ES object types.
+  // It is the OpenSL ES API entry point.
+  SLEngineItf engine_;
+
+  // The audio recorder media object records audio to the destination specified
+  // by the data sink capturing it from the input specified by the data source.
+  webrtc::ScopedSLObjectItf recorder_object_;
+
+  // This interface is supported on the audio recorder object and it controls
+  // the state of the audio recorder.
+  SLRecordItf recorder_;
+
+  // The Android Simple Buffer Queue interface is supported on the audio
+  // recorder. For recording, an app should enqueue empty buffers. When a
+  // registered callback sends notification that the system has finished writing
+  // data to the buffer, the app can read the buffer.
+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
+
+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
+  // chunks of audio.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Queue of audio buffers to be used by the recorder object for capturing
+  // audio. They will be used in a Round-robin way and the size of each buffer
+  // is given by AudioParameters::frames_per_buffer(), i.e., it corresponds to
+  // the native OpenSL ES buffer size.
+  std::unique_ptr<std::unique_ptr<SLint16[]>[]> audio_buffers_;
+
+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
+  int buffer_index_;
+
+  // Last time the OpenSL ES layer delivered recorded audio data.
+  uint32_t last_rec_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_RECORDER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_buffer.cc

@@ -0,0 +1,539 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_buffer.h"
+
+#include <string.h>
+
+#include <cmath>
+#include <cstddef>
+#include <cstdint>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+static const char kTimerQueueName[] = "AudioDeviceBufferTimer";
+
+// Time between two sucessive calls to LogStats().
+static const size_t kTimerIntervalInSeconds = 10;
+static const size_t kTimerIntervalInMilliseconds =
+    kTimerIntervalInSeconds * rtc::kNumMillisecsPerSec;
+// Min time required to qualify an audio session as a "call". If playout or
+// recording has been active for less than this time we will not store any
+// logs or UMA stats but instead consider the call as too short.
+static const size_t kMinValidCallTimeTimeInSeconds = 10;
+static const size_t kMinValidCallTimeTimeInMilliseconds =
+    kMinValidCallTimeTimeInSeconds * rtc::kNumMillisecsPerSec;
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+static const double k2Pi = 6.28318530717959;
+#endif
+
+AudioDeviceBuffer::AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
+                                     bool create_detached)
+    : task_queue_(task_queue_factory->CreateTaskQueue(
+          kTimerQueueName,
+          TaskQueueFactory::Priority::NORMAL)),
+      audio_transport_cb_(nullptr),
+      rec_sample_rate_(0),
+      play_sample_rate_(0),
+      rec_channels_(0),
+      play_channels_(0),
+      playing_(false),
+      recording_(false),
+      typing_status_(false),
+      play_delay_ms_(0),
+      rec_delay_ms_(0),
+      num_stat_reports_(0),
+      last_timer_task_time_(0),
+      rec_stat_count_(0),
+      play_stat_count_(0),
+      play_start_time_(0),
+      only_silence_recorded_(true),
+      log_stats_(false) {
+  RTC_LOG(LS_INFO) << "AudioDeviceBuffer::ctor";
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  phase_ = 0.0;
+  RTC_LOG(LS_WARNING) << "AUDIO_DEVICE_PLAYS_SINUS_TONE is defined!";
+#endif
+  if (create_detached) {
+    main_thread_checker_.Detach();
+  }
+}
+
+AudioDeviceBuffer::~AudioDeviceBuffer() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DCHECK(!playing_);
+  RTC_DCHECK(!recording_);
+  RTC_LOG(LS_INFO) << "AudioDeviceBuffer::~dtor";
+}
+
+int32_t AudioDeviceBuffer::RegisterAudioCallback(
+    AudioTransport* audio_callback) {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (playing_ || recording_) {
+    RTC_LOG(LS_ERROR) << "Failed to set audio transport since media was active";
+    return -1;
+  }
+  audio_transport_cb_ = audio_callback;
+  return 0;
+}
+
+void AudioDeviceBuffer::StartPlayout() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  // TODO(henrika): allow for usage of DCHECK(!playing_) here instead. Today the
+  // ADM allows calling Start(), Start() by ignoring the second call but it
+  // makes more sense to only allow one call.
+  if (playing_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Clear members tracking playout stats and do it on the task queue.
+  task_queue_.PostTask([this] { ResetPlayStats(); });
+  // Start a periodic timer based on task queue if not already done by the
+  // recording side.
+  if (!recording_) {
+    StartPeriodicLogging();
+  }
+  const int64_t now_time = rtc::TimeMillis();
+  // Clear members that are only touched on the main (creating) thread.
+  play_start_time_ = now_time;
+  playing_ = true;
+}
+
+void AudioDeviceBuffer::StartRecording() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (recording_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Clear members tracking recording stats and do it on the task queue.
+  task_queue_.PostTask([this] { ResetRecStats(); });
+  // Start a periodic timer based on task queue if not already done by the
+  // playout side.
+  if (!playing_) {
+    StartPeriodicLogging();
+  }
+  // Clear members that will be touched on the main (creating) thread.
+  rec_start_time_ = rtc::TimeMillis();
+  recording_ = true;
+  // And finally a member which can be modified on the native audio thread.
+  // It is safe to do so since we know by design that the owning ADM has not
+  // yet started the native audio recording.
+  only_silence_recorded_ = true;
+}
+
+void AudioDeviceBuffer::StopPlayout() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!playing_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  playing_ = false;
+  // Stop periodic logging if no more media is active.
+  if (!recording_) {
+    StopPeriodicLogging();
+  }
+  RTC_LOG(LS_INFO) << "total playout time: "
+                   << rtc::TimeSince(play_start_time_);
+}
+
+void AudioDeviceBuffer::StopRecording() {
+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
+  if (!recording_) {
+    return;
+  }
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  recording_ = false;
+  // Stop periodic logging if no more media is active.
+  if (!playing_) {
+    StopPeriodicLogging();
+  }
+  // Add UMA histogram to keep track of the case when only zeros have been
+  // recorded. Measurements (max of absolute level) are taken twice per second,
+  // which means that if e.g 10 seconds of audio has been recorded, a total of
+  // 20 level estimates must all be identical to zero to trigger the histogram.
+  // `only_silence_recorded_` can only be cleared on the native audio thread
+  // that drives audio capture but we know by design that the audio has stopped
+  // when this method is called, hence there should not be aby conflicts. Also,
+  // the fact that `only_silence_recorded_` can be affected during the complete
+  // call makes chances of conflicts with potentially one last callback very
+  // small.
+  const size_t time_since_start = rtc::TimeSince(rec_start_time_);
+  if (time_since_start > kMinValidCallTimeTimeInMilliseconds) {
+    const int only_zeros = static_cast<int>(only_silence_recorded_);
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros", only_zeros);
+    RTC_LOG(LS_INFO) << "HISTOGRAM(WebRTC.Audio.RecordedOnlyZeros): "
+                     << only_zeros;
+  }
+  RTC_LOG(LS_INFO) << "total recording time: " << time_since_start;
+}
+
+int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
+  RTC_LOG(LS_INFO) << "SetRecordingSampleRate(" << fsHz << ")";
+  rec_sample_rate_ = fsHz;
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
+  RTC_LOG(LS_INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
+  play_sample_rate_ = fsHz;
+  return 0;
+}
+
+uint32_t AudioDeviceBuffer::RecordingSampleRate() const {
+  return rec_sample_rate_;
+}
+
+uint32_t AudioDeviceBuffer::PlayoutSampleRate() const {
+  return play_sample_rate_;
+}
+
+int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
+  RTC_LOG(LS_INFO) << "SetRecordingChannels(" << channels << ")";
+  rec_channels_ = channels;
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
+  RTC_LOG(LS_INFO) << "SetPlayoutChannels(" << channels << ")";
+  play_channels_ = channels;
+  return 0;
+}
+
+size_t AudioDeviceBuffer::RecordingChannels() const {
+  return rec_channels_;
+}
+
+size_t AudioDeviceBuffer::PlayoutChannels() const {
+  return play_channels_;
+}
+
+int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
+  typing_status_ = typing_status;
+  return 0;
+}
+
+void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) {
+  play_delay_ms_ = play_delay_ms;
+  rec_delay_ms_ = rec_delay_ms;
+}
+
+int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
+                                             size_t samples_per_channel) {
+  return SetRecordedBuffer(audio_buffer, samples_per_channel, absl::nullopt);
+}
+
+int32_t AudioDeviceBuffer::SetRecordedBuffer(
+    const void* audio_buffer,
+    size_t samples_per_channel,
+    absl::optional<int64_t> capture_timestamp_ns) {
+  // Copy the complete input buffer to the local buffer.
+  const size_t old_size = rec_buffer_.size();
+  rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
+                      rec_channels_ * samples_per_channel);
+  // Keep track of the size of the recording buffer. Only updated when the
+  // size changes, which is a rare event.
+  if (old_size != rec_buffer_.size()) {
+    RTC_LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
+  }
+
+  if (capture_timestamp_ns) {
+    int64_t align_offsync_estimation_time = rtc::TimeMicros();
+    if (align_offsync_estimation_time -
+            rtc::TimestampAligner::kMinFrameIntervalUs >
+        align_offsync_estimation_time_) {
+      align_offsync_estimation_time_ = align_offsync_estimation_time;
+      capture_timestamp_ns_ =
+          rtc::kNumNanosecsPerMicrosec *
+          timestamp_aligner_.TranslateTimestamp(
+              *capture_timestamp_ns / rtc::kNumNanosecsPerMicrosec,
+              align_offsync_estimation_time);
+    } else {
+      // The Timestamp aligner is designed to prevent timestamps that are too
+      // similar, and produces warnings if it is called to often. We do not care
+      // about that here, so we do this workaround. If we where to call the
+      // aligner within a millisecond, we instead call this, that do not update
+      // the clock offset estimation. This get us timestamps without generating
+      // warnings, but could generate two timestamps within a millisecond.
+      capture_timestamp_ns_ =
+          rtc::kNumNanosecsPerMicrosec *
+          timestamp_aligner_.TranslateTimestamp(*capture_timestamp_ns /
+                                                rtc::kNumNanosecsPerMicrosec);
+    }
+  }
+  // Derive a new level value twice per second and check if it is non-zero.
+  int16_t max_abs = 0;
+  RTC_DCHECK_LT(rec_stat_count_, 50);
+  if (++rec_stat_count_ >= 50) {
+    // Returns the largest absolute value in a signed 16-bit vector.
+    max_abs = WebRtcSpl_MaxAbsValueW16(rec_buffer_.data(), rec_buffer_.size());
+    rec_stat_count_ = 0;
+    // Set `only_silence_recorded_` to false as soon as at least one detection
+    // of a non-zero audio packet is found. It can only be restored to true
+    // again by restarting the call.
+    if (max_abs > 0) {
+      only_silence_recorded_ = false;
+    }
+  }
+  // Update recording stats which is used as base for periodic logging of the
+  // audio input state.
+  UpdateRecStats(max_abs, samples_per_channel);
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::DeliverRecordedData() {
+  if (!audio_transport_cb_) {
+    RTC_LOG(LS_WARNING) << "Invalid audio transport";
+    return 0;
+  }
+  const size_t frames = rec_buffer_.size() / rec_channels_;
+  const size_t bytes_per_frame = rec_channels_ * sizeof(int16_t);
+  uint32_t new_mic_level_dummy = 0;
+  uint32_t total_delay_ms = play_delay_ms_ + rec_delay_ms_;
+  int32_t res = audio_transport_cb_->RecordedDataIsAvailable(
+      rec_buffer_.data(), frames, bytes_per_frame, rec_channels_,
+      rec_sample_rate_, total_delay_ms, 0, 0, typing_status_,
+      new_mic_level_dummy, capture_timestamp_ns_);
+  if (res == -1) {
+    RTC_LOG(LS_ERROR) << "RecordedDataIsAvailable() failed";
+  }
+  return 0;
+}
+
+int32_t AudioDeviceBuffer::RequestPlayoutData(size_t samples_per_channel) {
+  TRACE_EVENT1("webrtc", "AudioDeviceBuffer::RequestPlayoutData",
+               "samples_per_channel", samples_per_channel);
+
+  // The consumer can change the requested size on the fly and we therefore
+  // resize the buffer accordingly. Also takes place at the first call to this
+  // method.
+  const size_t total_samples = play_channels_ * samples_per_channel;
+  if (play_buffer_.size() != total_samples) {
+    play_buffer_.SetSize(total_samples);
+    RTC_LOG(LS_INFO) << "Size of playout buffer: " << play_buffer_.size();
+  }
+
+  size_t num_samples_out(0);
+  // It is currently supported to start playout without a valid audio
+  // transport object. Leads to warning and silence.
+  if (!audio_transport_cb_) {
+    RTC_LOG(LS_WARNING) << "Invalid audio transport";
+    return 0;
+  }
+
+  // Retrieve new 16-bit PCM audio data using the audio transport instance.
+  int64_t elapsed_time_ms = -1;
+  int64_t ntp_time_ms = -1;
+  const size_t bytes_per_frame = play_channels_ * sizeof(int16_t);
+  uint32_t res = audio_transport_cb_->NeedMorePlayData(
+      samples_per_channel, bytes_per_frame, play_channels_, play_sample_rate_,
+      play_buffer_.data(), num_samples_out, &elapsed_time_ms, &ntp_time_ms);
+  if (res != 0) {
+    RTC_LOG(LS_ERROR) << "NeedMorePlayData() failed";
+  }
+
+  // Derive a new level value twice per second.
+  int16_t max_abs = 0;
+  RTC_DCHECK_LT(play_stat_count_, 50);
+  if (++play_stat_count_ >= 50) {
+    // Returns the largest absolute value in a signed 16-bit vector.
+    max_abs =
+        WebRtcSpl_MaxAbsValueW16(play_buffer_.data(), play_buffer_.size());
+    play_stat_count_ = 0;
+  }
+  // Update playout stats which is used as base for periodic logging of the
+  // audio output state.
+  UpdatePlayStats(max_abs, num_samples_out / play_channels_);
+  return static_cast<int32_t>(num_samples_out / play_channels_);
+}
+
+int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
+  RTC_DCHECK_GT(play_buffer_.size(), 0);
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  const double phase_increment =
+      k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
+  int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
+  if (play_channels_ == 1) {
+    for (size_t i = 0; i < play_buffer_.size(); ++i) {
+      destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
+      phase_ += phase_increment;
+    }
+  } else if (play_channels_ == 2) {
+    for (size_t i = 0; i < play_buffer_.size() / 2; ++i) {
+      destination_r[2 * i] = destination_r[2 * i + 1] =
+          static_cast<int16_t>((sin(phase_) * (1 << 14)));
+      phase_ += phase_increment;
+    }
+  }
+#else
+  memcpy(audio_buffer, play_buffer_.data(),
+         play_buffer_.size() * sizeof(int16_t));
+#endif
+  // Return samples per channel or number of frames.
+  return static_cast<int32_t>(play_buffer_.size() / play_channels_);
+}
+
+void AudioDeviceBuffer::StartPeriodicLogging() {
+  task_queue_.PostTask([this] { LogStats(AudioDeviceBuffer::LOG_START); });
+}
+
+void AudioDeviceBuffer::StopPeriodicLogging() {
+  task_queue_.PostTask([this] { LogStats(AudioDeviceBuffer::LOG_STOP); });
+}
+
+void AudioDeviceBuffer::LogStats(LogState state) {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  int64_t now_time = rtc::TimeMillis();
+
+  if (state == AudioDeviceBuffer::LOG_START) {
+    // Reset counters at start. We will not add any logging in this state but
+    // the timer will started by posting a new (delayed) task.
+    num_stat_reports_ = 0;
+    last_timer_task_time_ = now_time;
+    log_stats_ = true;
+  } else if (state == AudioDeviceBuffer::LOG_STOP) {
+    // Stop logging and posting new tasks.
+    log_stats_ = false;
+  } else if (state == AudioDeviceBuffer::LOG_ACTIVE) {
+    // Keep logging unless logging was disabled while task was posted.
+  }
+
+  // Avoid adding more logs since we are in STOP mode.
+  if (!log_stats_) {
+    return;
+  }
+
+  int64_t next_callback_time = now_time + kTimerIntervalInMilliseconds;
+  int64_t time_since_last = rtc::TimeDiff(now_time, last_timer_task_time_);
+  last_timer_task_time_ = now_time;
+
+  Stats stats;
+  {
+    MutexLock lock(&lock_);
+    stats = stats_;
+    stats_.max_rec_level = 0;
+    stats_.max_play_level = 0;
+  }
+
+  // Cache current sample rate from atomic members.
+  const uint32_t rec_sample_rate = rec_sample_rate_;
+  const uint32_t play_sample_rate = play_sample_rate_;
+
+  // Log the latest statistics but skip the first two rounds just after state
+  // was set to LOG_START to ensure that we have at least one full stable
+  // 10-second interval for sample-rate estimation. Hence, first printed log
+  // will be after ~20 seconds.
+  if (++num_stat_reports_ > 2 &&
+      static_cast<size_t>(time_since_last) > kTimerIntervalInMilliseconds / 2) {
+    uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples;
+    float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
+    uint32_t abs_diff_rate_in_percent = 0;
+    if (rec_sample_rate > 0 && rate > 0) {
+      abs_diff_rate_in_percent = static_cast<uint32_t>(
+          0.5f +
+          ((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
+                               abs_diff_rate_in_percent);
+      RTC_LOG(LS_INFO) << "[REC : " << time_since_last << "msec, "
+                       << rec_sample_rate / 1000 << "kHz] callbacks: "
+                       << stats.rec_callbacks - last_stats_.rec_callbacks
+                       << ", "
+                          "samples: "
+                       << diff_samples
+                       << ", "
+                          "rate: "
+                       << static_cast<int>(rate + 0.5)
+                       << ", "
+                          "rate diff: "
+                       << abs_diff_rate_in_percent
+                       << "%, "
+                          "level: "
+                       << stats.max_rec_level;
+    }
+
+    diff_samples = stats.play_samples - last_stats_.play_samples;
+    rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
+    abs_diff_rate_in_percent = 0;
+    if (play_sample_rate > 0 && rate > 0) {
+      abs_diff_rate_in_percent = static_cast<uint32_t>(
+          0.5f +
+          ((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
+      RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
+                               abs_diff_rate_in_percent);
+      RTC_LOG(LS_INFO) << "[PLAY: " << time_since_last << "msec, "
+                       << play_sample_rate / 1000 << "kHz] callbacks: "
+                       << stats.play_callbacks - last_stats_.play_callbacks
+                       << ", "
+                          "samples: "
+                       << diff_samples
+                       << ", "
+                          "rate: "
+                       << static_cast<int>(rate + 0.5)
+                       << ", "
+                          "rate diff: "
+                       << abs_diff_rate_in_percent
+                       << "%, "
+                          "level: "
+                       << stats.max_play_level;
+    }
+  }
+  last_stats_ = stats;
+
+  int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
+  RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
+
+  // Keep posting new (delayed) tasks until state is changed to kLogStop.
+  task_queue_.PostDelayedTask(
+      [this] { AudioDeviceBuffer::LogStats(AudioDeviceBuffer::LOG_ACTIVE); },
+      TimeDelta::Millis(time_to_wait_ms));
+}
+
+void AudioDeviceBuffer::ResetRecStats() {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  last_stats_.ResetRecStats();
+  MutexLock lock(&lock_);
+  stats_.ResetRecStats();
+}
+
+void AudioDeviceBuffer::ResetPlayStats() {
+  RTC_DCHECK_RUN_ON(&task_queue_);
+  last_stats_.ResetPlayStats();
+  MutexLock lock(&lock_);
+  stats_.ResetPlayStats();
+}
+
+void AudioDeviceBuffer::UpdateRecStats(int16_t max_abs,
+                                       size_t samples_per_channel) {
+  MutexLock lock(&lock_);
+  ++stats_.rec_callbacks;
+  stats_.rec_samples += samples_per_channel;
+  if (max_abs > stats_.max_rec_level) {
+    stats_.max_rec_level = max_abs;
+  }
+}
+
+void AudioDeviceBuffer::UpdatePlayStats(int16_t max_abs,
+                                        size_t samples_per_channel) {
+  MutexLock lock(&lock_);
+  ++stats_.play_callbacks;
+  stats_.play_samples += samples_per_channel;
+  if (max_abs > stats_.max_play_level) {
+    stats_.max_play_level = max_abs;
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_buffer.h

@@ -0,0 +1,253 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <atomic>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/timestamp_aligner.h"
+
+namespace webrtc {
+
+// Delta times between two successive playout callbacks are limited to this
+// value before added to an internal array.
+const size_t kMaxDeltaTimeInMs = 500;
+// TODO(henrika): remove when no longer used by external client.
+const size_t kMaxBufferSizeBytes = 3840;  // 10ms in stereo @ 96kHz
+
+class AudioDeviceBuffer {
+ public:
+  enum LogState {
+    LOG_START = 0,
+    LOG_STOP,
+    LOG_ACTIVE,
+  };
+
+  struct Stats {
+    void ResetRecStats() {
+      rec_callbacks = 0;
+      rec_samples = 0;
+      max_rec_level = 0;
+    }
+
+    void ResetPlayStats() {
+      play_callbacks = 0;
+      play_samples = 0;
+      max_play_level = 0;
+    }
+
+    // Total number of recording callbacks where the source provides 10ms audio
+    // data each time.
+    uint64_t rec_callbacks = 0;
+
+    // Total number of playback callbacks where the sink asks for 10ms audio
+    // data each time.
+    uint64_t play_callbacks = 0;
+
+    // Total number of recorded audio samples.
+    uint64_t rec_samples = 0;
+
+    // Total number of played audio samples.
+    uint64_t play_samples = 0;
+
+    // Contains max level (max(abs(x))) of recorded audio packets over the last
+    // 10 seconds where a new measurement is done twice per second. The level
+    // is reset to zero at each call to LogStats().
+    int16_t max_rec_level = 0;
+
+    // Contains max level of recorded audio packets over the last 10 seconds
+    // where a new measurement is done twice per second.
+    int16_t max_play_level = 0;
+  };
+
+  // If `create_detached` is true, the created buffer can be used on another
+  // thread compared to the one on which it was created. It's useful for
+  // testing.
+  explicit AudioDeviceBuffer(TaskQueueFactory* task_queue_factory,
+                             bool create_detached = false);
+  virtual ~AudioDeviceBuffer();
+
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback);
+
+  void StartPlayout();
+  void StartRecording();
+  void StopPlayout();
+  void StopRecording();
+
+  int32_t SetRecordingSampleRate(uint32_t fsHz);
+  int32_t SetPlayoutSampleRate(uint32_t fsHz);
+  uint32_t RecordingSampleRate() const;
+  uint32_t PlayoutSampleRate() const;
+
+  int32_t SetRecordingChannels(size_t channels);
+  int32_t SetPlayoutChannels(size_t channels);
+  size_t RecordingChannels() const;
+  size_t PlayoutChannels() const;
+
+  // TODO(bugs.webrtc.org/13621) Deprecate this function
+  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
+                                    size_t samples_per_channel);
+
+  virtual int32_t SetRecordedBuffer(
+      const void* audio_buffer,
+      size_t samples_per_channel,
+      absl::optional<int64_t> capture_timestamp_ns);
+  virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
+  virtual int32_t DeliverRecordedData();
+  uint32_t NewMicLevel() const;
+
+  virtual int32_t RequestPlayoutData(size_t samples_per_channel);
+  virtual int32_t GetPlayoutData(void* audio_buffer);
+
+  int32_t SetTypingStatus(bool typing_status);
+
+ private:
+  // Starts/stops periodic logging of audio stats.
+  void StartPeriodicLogging();
+  void StopPeriodicLogging();
+
+  // Called periodically on the internal thread created by the TaskQueue.
+  // Updates some stats but dooes it on the task queue to ensure that access of
+  // members is serialized hence avoiding usage of locks.
+  // state = LOG_START => members are initialized and the timer starts.
+  // state = LOG_STOP => no logs are printed and the timer stops.
+  // state = LOG_ACTIVE => logs are printed and the timer is kept alive.
+  void LogStats(LogState state);
+
+  // Updates counters in each play/record callback. These counters are later
+  // (periodically) read by LogStats() using a lock.
+  void UpdateRecStats(int16_t max_abs, size_t samples_per_channel);
+  void UpdatePlayStats(int16_t max_abs, size_t samples_per_channel);
+
+  // Clears all members tracking stats for recording and playout.
+  // These methods both run on the task queue.
+  void ResetRecStats();
+  void ResetPlayStats();
+
+  // This object lives on the main (creating) thread and most methods are
+  // called on that same thread. When audio has started some methods will be
+  // called on either a native audio thread for playout or a native thread for
+  // recording. Some members are not annotated since they are "protected by
+  // design" and adding e.g. a race checker can cause failures for very few
+  // edge cases and it is IMHO not worth the risk to use them in this class.
+  // TODO(henrika): see if it is possible to refactor and annotate all members.
+
+  // Main thread on which this object is created.
+  SequenceChecker main_thread_checker_;
+
+  Mutex lock_;
+
+  // Task queue used to invoke LogStats() periodically. Tasks are executed on a
+  // worker thread but it does not necessarily have to be the same thread for
+  // each task.
+  rtc::TaskQueue task_queue_;
+
+  // Raw pointer to AudioTransport instance. Supplied to RegisterAudioCallback()
+  // and it must outlive this object. It is not possible to change this member
+  // while any media is active. It is possible to start media without calling
+  // RegisterAudioCallback() but that will lead to ignored audio callbacks in
+  // both directions where native audio will be active but no audio samples will
+  // be transported.
+  AudioTransport* audio_transport_cb_;
+
+  // Sample rate in Hertz. Accessed atomically.
+  std::atomic<uint32_t> rec_sample_rate_;
+  std::atomic<uint32_t> play_sample_rate_;
+
+  // Number of audio channels. Accessed atomically.
+  std::atomic<size_t> rec_channels_;
+  std::atomic<size_t> play_channels_;
+
+  // Keeps track of if playout/recording are active or not. A combination
+  // of these states are used to determine when to start and stop the timer.
+  // Only used on the creating thread and not used to control any media flow.
+  bool playing_ RTC_GUARDED_BY(main_thread_checker_);
+  bool recording_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Buffer used for audio samples to be played out. Size can be changed
+  // dynamically. The 16-bit samples are interleaved, hence the size is
+  // proportional to the number of channels.
+  rtc::BufferT<int16_t> play_buffer_;
+
+  // Byte buffer used for recorded audio samples. Size can be changed
+  // dynamically.
+  rtc::BufferT<int16_t> rec_buffer_;
+
+  // Contains true of a key-press has been detected.
+  bool typing_status_;
+
+  // Delay values used by the AEC.
+  int play_delay_ms_;
+  int rec_delay_ms_;
+
+  // Capture timestamp.
+  absl::optional<int64_t> capture_timestamp_ns_;
+  // The last time the Timestamp Aligner was used to estimate clock offset
+  // between system clock and capture time from audio.
+  // This is used to prevent estimating the clock offset too often.
+  absl::optional<int64_t> align_offsync_estimation_time_;
+
+  // Counts number of times LogStats() has been called.
+  size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
+
+  // Time stamp of last timer task (drives logging).
+  int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
+
+  // Counts number of audio callbacks modulo 50 to create a signal when
+  // a new storage of audio stats shall be done.
+  int16_t rec_stat_count_;
+  int16_t play_stat_count_;
+
+  // Time stamps of when playout and recording starts.
+  int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+  int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
+
+  // Contains counters for playout and recording statistics.
+  Stats stats_ RTC_GUARDED_BY(lock_);
+
+  // Stores current stats at each timer task. Used to calculate differences
+  // between two successive timer events.
+  Stats last_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Set to true at construction and modified to false as soon as one audio-
+  // level estimate larger than zero is detected.
+  bool only_silence_recorded_;
+
+  // Set to true when logging of audio stats is enabled for the first time in
+  // StartPeriodicLogging() and set to false by StopPeriodicLogging().
+  // Setting this member to false prevents (possiby invalid) log messages from
+  // being printed in the LogStats() task.
+  bool log_stats_ RTC_GUARDED_BY(task_queue_);
+
+  // Used for converting capture timestaps (received from AudioRecordThread
+  // via AudioRecordJni::DataIsRecorded) to RTC clock.
+  rtc::TimestampAligner timestamp_aligner_;
+
+// Should *never* be defined in production builds. Only used for testing.
+// When defined, the output signal will be replaced by a sinus tone at 440Hz.
+#ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
+  double phase_;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_config.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
+
+// Enumerators
+//
+enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
+
+// Platform specifics
+//
+#if defined(_WIN32)
+#if (_MSC_VER >= 1400)
+#if !defined(WEBRTC_DUMMY_FILE_DEVICES)
+// Windows Core Audio is the default audio layer in Windows.
+// Only supported for VS 2005 and higher.
+#define WEBRTC_WINDOWS_CORE_AUDIO_BUILD
+#endif
+#endif
+#endif
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_data_observer.cc

@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device_data_observer.h"
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// A wrapper over AudioDeviceModule that registers itself as AudioTransport
+// callback and redirects the PCM data to AudioDeviceDataObserver callback.
+class ADMWrapper : public AudioDeviceModule, public AudioTransport {
+ public:
+  ADMWrapper(rtc::scoped_refptr<AudioDeviceModule> impl,
+             AudioDeviceDataObserver* legacy_observer,
+             std::unique_ptr<AudioDeviceDataObserver> observer)
+      : impl_(impl),
+        legacy_observer_(legacy_observer),
+        observer_(std::move(observer)) {
+    is_valid_ = impl_.get() != nullptr;
+  }
+  ADMWrapper(AudioLayer audio_layer,
+             TaskQueueFactory* task_queue_factory,
+             AudioDeviceDataObserver* legacy_observer,
+             std::unique_ptr<AudioDeviceDataObserver> observer)
+      : ADMWrapper(AudioDeviceModule::Create(audio_layer, task_queue_factory),
+                   legacy_observer,
+                   std::move(observer)) {}
+  ~ADMWrapper() override {
+    audio_transport_ = nullptr;
+    observer_ = nullptr;
+  }
+
+  // Make sure we have a valid ADM before returning it to user.
+  bool IsValid() { return is_valid_; }
+
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  size_t nSamples,
+                                  size_t nBytesPerSample,
+                                  size_t nChannels,
+                                  uint32_t samples_per_sec,
+                                  uint32_t total_delay_ms,
+                                  int32_t clockDrift,
+                                  uint32_t currentMicLevel,
+                                  bool keyPressed,
+                                  uint32_t& newMicLevel) override {
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+        total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+        /*capture_timestamp_ns=*/absl::nullopt);
+  }
+
+  // AudioTransport methods overrides.
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samples_per_sec,
+      uint32_t total_delay_ms,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> capture_timestamp_ns) override {
+    int32_t res = 0;
+    // Capture PCM data of locally captured audio.
+    if (observer_) {
+      observer_->OnCaptureData(audioSamples, nSamples, nBytesPerSample,
+                               nChannels, samples_per_sec);
+    }
+
+    // Send to the actual audio transport.
+    if (audio_transport_) {
+      res = audio_transport_->RecordedDataIsAvailable(
+          audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+          total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+          capture_timestamp_ns);
+    }
+
+    return res;
+  }
+
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samples_per_sec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
+    int32_t res = 0;
+    // Set out parameters to safe values to be sure not to return corrupted
+    // data.
+    nSamplesOut = 0;
+    *elapsed_time_ms = -1;
+    *ntp_time_ms = -1;
+    // Request data from audio transport.
+    if (audio_transport_) {
+      res = audio_transport_->NeedMorePlayData(
+          nSamples, nBytesPerSample, nChannels, samples_per_sec, audioSamples,
+          nSamplesOut, elapsed_time_ms, ntp_time_ms);
+    }
+
+    // Capture rendered data.
+    if (observer_) {
+      observer_->OnRenderData(audioSamples, nSamples, nBytesPerSample,
+                              nChannels, samples_per_sec);
+    }
+
+    return res;
+  }
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {
+    RTC_DCHECK_NOTREACHED();
+  }
+
+  // Override AudioDeviceModule's RegisterAudioCallback method to remember the
+  // actual audio transport (e.g.: voice engine).
+  int32_t RegisterAudioCallback(AudioTransport* audio_callback) override {
+    // Remember the audio callback to forward PCM data
+    audio_transport_ = audio_callback;
+    return 0;
+  }
+
+  // AudioDeviceModule pass through method overrides.
+  int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override {
+    return impl_->ActiveAudioLayer(audio_layer);
+  }
+  int32_t Init() override {
+    int res = impl_->Init();
+    if (res != 0) {
+      return res;
+    }
+    // Register self as the audio transport callback for underlying ADM impl.
+    impl_->RegisterAudioCallback(this);
+    return res;
+  }
+  int32_t Terminate() override { return impl_->Terminate(); }
+  bool Initialized() const override { return impl_->Initialized(); }
+  int16_t PlayoutDevices() override { return impl_->PlayoutDevices(); }
+  int16_t RecordingDevices() override { return impl_->RecordingDevices(); }
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return impl_->PlayoutDeviceName(index, name, guid);
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return impl_->RecordingDeviceName(index, name, guid);
+  }
+  int32_t SetPlayoutDevice(uint16_t index) override {
+    return impl_->SetPlayoutDevice(index);
+  }
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override {
+    return impl_->SetPlayoutDevice(device);
+  }
+  int32_t SetRecordingDevice(uint16_t index) override {
+    return impl_->SetRecordingDevice(index);
+  }
+  int32_t SetRecordingDevice(WindowsDeviceType device) override {
+    return impl_->SetRecordingDevice(device);
+  }
+  int32_t PlayoutIsAvailable(bool* available) override {
+    return impl_->PlayoutIsAvailable(available);
+  }
+  int32_t InitPlayout() override { return impl_->InitPlayout(); }
+  bool PlayoutIsInitialized() const override {
+    return impl_->PlayoutIsInitialized();
+  }
+  int32_t RecordingIsAvailable(bool* available) override {
+    return impl_->RecordingIsAvailable(available);
+  }
+  int32_t InitRecording() override { return impl_->InitRecording(); }
+  bool RecordingIsInitialized() const override {
+    return impl_->RecordingIsInitialized();
+  }
+  int32_t StartPlayout() override { return impl_->StartPlayout(); }
+  int32_t StopPlayout() override { return impl_->StopPlayout(); }
+  bool Playing() const override { return impl_->Playing(); }
+  int32_t StartRecording() override { return impl_->StartRecording(); }
+  int32_t StopRecording() override { return impl_->StopRecording(); }
+  bool Recording() const override { return impl_->Recording(); }
+  int32_t InitSpeaker() override { return impl_->InitSpeaker(); }
+  bool SpeakerIsInitialized() const override {
+    return impl_->SpeakerIsInitialized();
+  }
+  int32_t InitMicrophone() override { return impl_->InitMicrophone(); }
+  bool MicrophoneIsInitialized() const override {
+    return impl_->MicrophoneIsInitialized();
+  }
+  int32_t SpeakerVolumeIsAvailable(bool* available) override {
+    return impl_->SpeakerVolumeIsAvailable(available);
+  }
+  int32_t SetSpeakerVolume(uint32_t volume) override {
+    return impl_->SetSpeakerVolume(volume);
+  }
+  int32_t SpeakerVolume(uint32_t* volume) const override {
+    return impl_->SpeakerVolume(volume);
+  }
+  int32_t MaxSpeakerVolume(uint32_t* max_volume) const override {
+    return impl_->MaxSpeakerVolume(max_volume);
+  }
+  int32_t MinSpeakerVolume(uint32_t* min_volume) const override {
+    return impl_->MinSpeakerVolume(min_volume);
+  }
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override {
+    return impl_->MicrophoneVolumeIsAvailable(available);
+  }
+  int32_t SetMicrophoneVolume(uint32_t volume) override {
+    return impl_->SetMicrophoneVolume(volume);
+  }
+  int32_t MicrophoneVolume(uint32_t* volume) const override {
+    return impl_->MicrophoneVolume(volume);
+  }
+  int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override {
+    return impl_->MaxMicrophoneVolume(max_volume);
+  }
+  int32_t MinMicrophoneVolume(uint32_t* min_volume) const override {
+    return impl_->MinMicrophoneVolume(min_volume);
+  }
+  int32_t SpeakerMuteIsAvailable(bool* available) override {
+    return impl_->SpeakerMuteIsAvailable(available);
+  }
+  int32_t SetSpeakerMute(bool enable) override {
+    return impl_->SetSpeakerMute(enable);
+  }
+  int32_t SpeakerMute(bool* enabled) const override {
+    return impl_->SpeakerMute(enabled);
+  }
+  int32_t MicrophoneMuteIsAvailable(bool* available) override {
+    return impl_->MicrophoneMuteIsAvailable(available);
+  }
+  int32_t SetMicrophoneMute(bool enable) override {
+    return impl_->SetMicrophoneMute(enable);
+  }
+  int32_t MicrophoneMute(bool* enabled) const override {
+    return impl_->MicrophoneMute(enabled);
+  }
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    return impl_->StereoPlayoutIsAvailable(available);
+  }
+  int32_t SetStereoPlayout(bool enable) override {
+    return impl_->SetStereoPlayout(enable);
+  }
+  int32_t StereoPlayout(bool* enabled) const override {
+    return impl_->StereoPlayout(enabled);
+  }
+  int32_t StereoRecordingIsAvailable(bool* available) const override {
+    return impl_->StereoRecordingIsAvailable(available);
+  }
+  int32_t SetStereoRecording(bool enable) override {
+    return impl_->SetStereoRecording(enable);
+  }
+  int32_t StereoRecording(bool* enabled) const override {
+    return impl_->StereoRecording(enabled);
+  }
+  int32_t PlayoutDelay(uint16_t* delay_ms) const override {
+    return impl_->PlayoutDelay(delay_ms);
+  }
+  bool BuiltInAECIsAvailable() const override {
+    return impl_->BuiltInAECIsAvailable();
+  }
+  bool BuiltInAGCIsAvailable() const override {
+    return impl_->BuiltInAGCIsAvailable();
+  }
+  bool BuiltInNSIsAvailable() const override {
+    return impl_->BuiltInNSIsAvailable();
+  }
+  int32_t EnableBuiltInAEC(bool enable) override {
+    return impl_->EnableBuiltInAEC(enable);
+  }
+  int32_t EnableBuiltInAGC(bool enable) override {
+    return impl_->EnableBuiltInAGC(enable);
+  }
+  int32_t EnableBuiltInNS(bool enable) override {
+    return impl_->EnableBuiltInNS(enable);
+  }
+  int32_t GetPlayoutUnderrunCount() const override {
+    return impl_->GetPlayoutUnderrunCount();
+  }
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
+    return impl_->GetPlayoutAudioParameters(params);
+  }
+  int GetRecordAudioParameters(AudioParameters* params) const override {
+    return impl_->GetRecordAudioParameters(params);
+  }
+#endif  // WEBRTC_IOS
+
+ protected:
+  rtc::scoped_refptr<AudioDeviceModule> impl_;
+  AudioDeviceDataObserver* legacy_observer_ = nullptr;
+  std::unique_ptr<AudioDeviceDataObserver> observer_;
+  AudioTransport* audio_transport_ = nullptr;
+  bool is_valid_ = false;
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    std::unique_ptr<AudioDeviceDataObserver> observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(impl, observer.get(),
+                                                        std::move(observer));
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    AudioDeviceDataObserver* legacy_observer) {
+  auto audio_device =
+      rtc::make_ref_counted<ADMWrapper>(impl, legacy_observer, nullptr);
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<AudioDeviceDataObserver> observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(
+      audio_layer, task_queue_factory, observer.get(), std::move(observer));
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    AudioDeviceDataObserver* legacy_observer) {
+  auto audio_device = rtc::make_ref_counted<ADMWrapper>(
+      audio_layer, task_queue_factory, legacy_observer, nullptr);
+
+  if (!audio_device->IsValid()) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_generic.cc

@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_generic.h"
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+bool AudioDeviceGeneric::BuiltInAECIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInAGCIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInAGC(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+bool AudioDeviceGeneric::BuiltInNSIsAvailable() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return false;
+}
+
+int32_t AudioDeviceGeneric::EnableBuiltInNS(bool enable) {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+int32_t AudioDeviceGeneric::GetPlayoutUnderrunCount() const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+#if defined(WEBRTC_IOS)
+int AudioDeviceGeneric::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+
+int AudioDeviceGeneric::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG_F(LS_ERROR) << "Not supported on this platform";
+  return -1;
+}
+#endif  // WEBRTC_IOS
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_generic.h

@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_
+
+#include <stdint.h>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric {
+ public:
+  // For use with UMA logging. Must be kept in sync with histograms.xml in
+  // Chrome, located at
+  // https://cs.chromium.org/chromium/src/tools/metrics/histograms/histograms.xml
+  enum class InitStatus {
+    OK = 0,
+    PLAYOUT_ERROR = 1,
+    RECORDING_ERROR = 2,
+    OTHER_ERROR = 3,
+    NUM_STATUSES = 4
+  };
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const = 0;
+
+  // Main initializaton and termination
+  virtual InitStatus Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) = 0;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool& available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const = 0;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const = 0;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool& enabled) const = 0;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool& enabled) const = 0;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool& enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool& available) = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool& enabled) const = 0;
+
+  // Delay information and control
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const = 0;
+
+  // Android only
+  virtual bool BuiltInAECIsAvailable() const;
+  virtual bool BuiltInAGCIsAvailable() const;
+  virtual bool BuiltInNSIsAvailable() const;
+
+  // Windows Core Audio and Android only.
+  virtual int32_t EnableBuiltInAEC(bool enable);
+  virtual int32_t EnableBuiltInAGC(bool enable);
+  virtual int32_t EnableBuiltInNS(bool enable);
+
+  // Play underrun count.
+  virtual int32_t GetPlayoutUnderrunCount() const;
+
+// iOS only.
+// TODO(henrika): add Android support.
+#if defined(WEBRTC_IOS)
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const;
+#endif  // WEBRTC_IOS
+
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
+
+  virtual ~AudioDeviceGeneric() {}
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_impl.cc

@@ -0,0 +1,986 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_impl.h"
+
+#include <stddef.h>
+
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_device/audio_device_config.h"  // IWYU pragma: keep
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+#if defined(_WIN32)
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+#include "modules/audio_device/win/audio_device_core_win.h"
+#endif
+#elif defined(WEBRTC_ANDROID)
+#include <stdlib.h>
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+#include "modules/audio_device/android/aaudio_player.h"
+#include "modules/audio_device/android/aaudio_recorder.h"
+#endif
+#include "modules/audio_device/android/audio_device_template.h"
+#include "modules/audio_device/android/audio_manager.h"
+#include "modules/audio_device/android/audio_record_jni.h"
+#include "modules/audio_device/android/audio_screen_record_jni.h"
+#include "modules/audio_device/android/audio_merged_screen_record_jni.h"
+#include "modules/audio_device/android/audio_track_jni.h"
+#include "modules/audio_device/android/opensles_player.h"
+#include "modules/audio_device/android/opensles_recorder.h"
+#elif defined(WEBRTC_LINUX)
+#if defined(WEBRTC_ENABLE_LINUX_ALSA)
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+#endif
+#if defined(WEBRTC_ENABLE_LINUX_PULSE)
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+#endif
+#elif defined(WEBRTC_IOS)
+#include "sdk/objc/native/src/audio/audio_device_ios.h"
+#elif defined(WEBRTC_MAC)
+#include "modules/audio_device/mac/audio_device_mac.h"
+#endif
+#if defined(WEBRTC_DUMMY_FILE_DEVICES)
+#include "modules/audio_device/dummy/file_audio_device.h"
+#include "modules/audio_device/dummy/file_audio_device_factory.h"
+#endif
+#include "modules/audio_device/dummy/audio_device_dummy.h"
+
+#define CHECKinitialized_() \
+  {                         \
+    if (!initialized_) {    \
+      return -1;            \
+    }                       \
+  }
+
+#define CHECKinitialized__BOOL() \
+  {                              \
+    if (!initialized_) {         \
+      return false;              \
+    }                            \
+  }
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
+}
+
+// static
+rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
+  // dedicated factory method which should be used instead.
+  if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
+    RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() "
+                         "factory method instead for this option.";
+    return nullptr;
+  } else if (audio_layer == AudioDeviceModule::kAndroidJavaAudio ||
+             audio_layer == AudioDeviceModule::kAndroidOpenSLESAudio ||
+             audio_layer == AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio ||
+             audio_layer == kAndroidAAudioAudio ||
+             audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
+    RTC_LOG(LS_ERROR) << "Use the CreateAndroidAudioDeviceModule() "
+                         "factory method instead for this option.";
+    return nullptr;
+  }
+
+  // Create the generic reference counted (platform independent) implementation.
+  auto audio_device = rtc::make_ref_counted<AudioDeviceModuleImpl>(
+      audio_layer, task_queue_factory);
+
+  // Ensure that the current platform is supported.
+  if (audio_device->CheckPlatform() == -1) {
+    return nullptr;
+  }
+
+  // Create the platform-dependent implementation.
+  if (audio_device->CreatePlatformSpecificObjects() == -1) {
+    return nullptr;
+  }
+
+  // Ensure that the generic audio buffer can communicate with the platform
+  // specific parts.
+  if (audio_device->AttachAudioBuffer() == -1) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+AudioDeviceModuleImpl::AudioDeviceModuleImpl(
+    AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory)
+    : audio_layer_(audio_layer), audio_device_buffer_(task_queue_factory) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+AudioDeviceModuleImpl::AudioDeviceModuleImpl(
+    AudioLayer audio_layer,
+    std::unique_ptr<AudioDeviceGeneric> audio_device,
+    TaskQueueFactory* task_queue_factory,
+    bool create_detached)
+    : audio_layer_(audio_layer),
+      audio_device_buffer_(task_queue_factory, create_detached),
+      audio_device_(std::move(audio_device)) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+int32_t AudioDeviceModuleImpl::CheckPlatform() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Ensure that the current platform is supported
+  PlatformType platform(kPlatformNotSupported);
+#if defined(_WIN32)
+  platform = kPlatformWin32;
+  RTC_LOG(LS_INFO) << "current platform is Win32";
+#elif defined(WEBRTC_ANDROID)
+  platform = kPlatformAndroid;
+  RTC_LOG(LS_INFO) << "current platform is Android";
+#elif defined(WEBRTC_LINUX)
+  platform = kPlatformLinux;
+  RTC_LOG(LS_INFO) << "current platform is Linux";
+#elif defined(WEBRTC_IOS)
+  platform = kPlatformIOS;
+  RTC_LOG(LS_INFO) << "current platform is IOS";
+#elif defined(WEBRTC_MAC)
+  platform = kPlatformMac;
+  RTC_LOG(LS_INFO) << "current platform is Mac";
+#elif defined(WEBRTC_FUCHSIA)
+  platform = kPlatformFuchsia;
+  RTC_LOG(LS_INFO) << "current platform is Fuchsia";
+#endif
+  if (platform == kPlatformNotSupported) {
+    RTC_LOG(LS_ERROR)
+        << "current platform is not supported => this module will self "
+           "destruct!";
+    return -1;
+  }
+  platform_type_ = platform;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (audio_device_ != nullptr) {
+    RTC_LOG(LS_INFO) << "Reusing provided audio device";
+    return 0;
+  }
+// Dummy ADM implementations if build flags are set.
+#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+  audio_device_.reset(new AudioDeviceDummy());
+  RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
+#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
+  audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
+  if (audio_device_) {
+    RTC_LOG(LS_INFO) << "Will use file-playing dummy device.";
+  } else {
+    // Create a dummy device instead.
+    audio_device_.reset(new AudioDeviceDummy());
+    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
+  }
+
+// Real (non-dummy) ADM implementations.
+#else
+  AudioLayer audio_layer(PlatformAudioLayer());
+// Windows ADM implementation.
+#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+  if ((audio_layer == kWindowsCoreAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    RTC_LOG(LS_INFO) << "Attempting to use the Windows Core Audio APIs...";
+    if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
+      audio_device_.reset(new AudioDeviceWindowsCore());
+      RTC_LOG(LS_INFO) << "Windows Core Audio APIs will be utilized";
+    }
+  }
+#endif  // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
+
+#if defined(WEBRTC_ANDROID)
+  // Create an Android audio manager.
+  audio_manager_android_.reset(new AudioManager());
+  // Select best possible combination of audio layers.
+  if (audio_layer == kPlatformDefaultAudio) {
+    if (audio_manager_android_->IsAAudioSupported()) {
+      // Use of AAudio for both playout and recording has highest priority.
+      audio_layer = kAndroidAAudioAudio;
+    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
+               audio_manager_android_->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for both playout and recording.
+      audio_layer = kAndroidOpenSLESAudio;
+    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
+               !audio_manager_android_->IsLowLatencyRecordSupported()) {
+      // Use OpenSL ES for output on devices that only supports the
+      // low-latency output audio path.
+      audio_layer = kAndroidJavaInputAndOpenSLESOutputAudio;
+    } else {
+      // Use Java-based audio in both directions when low-latency output is
+      // not supported.
+      audio_layer = kAndroidJavaAudio;
+    }
+  }
+  AudioManager* audio_manager = audio_manager_android_.get();
+  if (audio_layer == kAndroidJavaAudio) {
+    // Java audio for both input and output audio.
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
+        audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidScreenAudio) {
+    // Java audio for both input and output audio.
+    audio_device_.reset(new AudioDeviceTemplate<AudioScreenRecordJni, AudioTrackJni>(
+            audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidMergedScreenAudio) {
+    // Java audio for both input and output audio.
+    audio_device_.reset(new AudioDeviceTemplate<AudioMergedScreenRecordJni, AudioTrackJni>(
+            audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidOpenSLESAudio) {
+    // OpenSL ES based audio for both input and output audio.
+    audio_device_.reset(
+        new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
+            audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidJavaInputAndOpenSLESOutputAudio) {
+    // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
+    // This combination provides low-latency output audio and at the same
+    // time support for HW AEC using the AudioRecord Java API.
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
+        audio_layer, audio_manager));
+  } else if (audio_layer == kAndroidAAudioAudio) {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+    // AAudio based audio for both input and output.
+    audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
+        audio_layer, audio_manager));
+#endif
+  } else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
+#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
+    // Java audio for input and AAudio for output audio (i.e. mixed APIs).
+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
+        audio_layer, audio_manager));
+#endif
+  } else {
+    RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
+    audio_device_.reset(nullptr);
+  }
+#endif
+// END #if defined(WEBRTC_ANDROID)
+
+// Linux ADM implementation.
+// Note that, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when
+// WEBRTC_LINUX is defined. WEBRTC_ENABLE_LINUX_PULSE depends on the
+// 'rtc_include_pulse_audio' build flag.
+// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
+// PulseAudio is the default selection.
+#if !defined(WEBRTC_ANDROID) && defined(WEBRTC_LINUX)
+#if !defined(WEBRTC_ENABLE_LINUX_PULSE)
+  // Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
+  // - kPlatformDefaultAudio => ALSA, and
+  // - kLinuxAlsaAudio => ALSA, and
+  // - kLinuxPulseAudio => Invalid selection.
+  RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
+  if ((audio_layer == kLinuxAlsaAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    audio_device_.reset(new AudioDeviceLinuxALSA());
+    RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
+  }
+#else
+  // Build flag 'rtc_include_pulse_audio' is set to true (default). In this
+  // mode:
+  // - kPlatformDefaultAudio => PulseAudio, and
+  // - kLinuxPulseAudio => PulseAudio, and
+  // - kLinuxAlsaAudio => ALSA (supported but not default).
+  RTC_LOG(LS_INFO) << "PulseAudio support is enabled.";
+  if ((audio_layer == kLinuxPulseAudio) ||
+      (audio_layer == kPlatformDefaultAudio)) {
+    // Linux PulseAudio implementation is default.
+    audio_device_.reset(new AudioDeviceLinuxPulse());
+    RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized";
+  } else if (audio_layer == kLinuxAlsaAudio) {
+    audio_device_.reset(new AudioDeviceLinuxALSA());
+    RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized.";
+  }
+#endif  // #if !defined(WEBRTC_ENABLE_LINUX_PULSE)
+#endif  // #if defined(WEBRTC_LINUX)
+
+// iOS ADM implementation.
+#if defined(WEBRTC_IOS)
+  if (audio_layer == kPlatformDefaultAudio) {
+    audio_device_.reset(
+        new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
+    RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
+  }
+// END #if defined(WEBRTC_IOS)
+
+// Mac OS X ADM implementation.
+#elif defined(WEBRTC_MAC)
+  if (audio_layer == kPlatformDefaultAudio) {
+    audio_device_.reset(new AudioDeviceMac());
+    RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized.";
+  }
+#endif  // WEBRTC_MAC
+
+  // Dummy ADM implementation.
+  if (audio_layer == kDummyAudio) {
+    audio_device_.reset(new AudioDeviceDummy());
+    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized.";
+  }
+#endif  // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
+
+  if (!audio_device_) {
+    RTC_LOG(LS_ERROR)
+        << "Failed to create the platform specific ADM implementation.";
+    return -1;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  audio_device_->AttachAudioBuffer(&audio_device_buffer_);
+  return 0;
+}
+
+AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+}
+
+int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  AudioLayer activeAudio;
+  if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
+    return -1;
+  }
+  *audioLayer = activeAudio;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::Init() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (initialized_)
+    return 0;
+  RTC_CHECK(audio_device_);
+  AudioDeviceGeneric::InitStatus status = audio_device_->Init();
+  RTC_HISTOGRAM_ENUMERATION(
+      "WebRTC.Audio.InitializationResult", static_cast<int>(status),
+      static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
+  if (status != AudioDeviceGeneric::InitStatus::OK) {
+    RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
+    return -1;
+  }
+  initialized_ = true;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::Terminate() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  if (!initialized_)
+    return 0;
+  if (audio_device_->Terminate() == -1) {
+    return -1;
+  }
+  initialized_ = false;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::Initialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
+  return initialized_;
+}
+
+int32_t AudioDeviceModuleImpl::InitSpeaker() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->InitSpeaker();
+}
+
+int32_t AudioDeviceModuleImpl::InitMicrophone() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->InitMicrophone();
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+  CHECKinitialized_();
+  return audio_device_->SetSpeakerVolume(volume);
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint32_t level = 0;
+  if (audio_device_->SpeakerVolume(level) == -1) {
+    return -1;
+  }
+  *volume = level;
+  RTC_LOG(LS_INFO) << "output: " << *volume;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::SpeakerIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isInitialized = audio_device_->SpeakerIsInitialized();
+  RTC_LOG(LS_INFO) << "output: " << isInitialized;
+  return isInitialized;
+}
+
+bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isInitialized = audio_device_->MicrophoneIsInitialized();
+  RTC_LOG(LS_INFO) << "output: " << isInitialized;
+  return isInitialized;
+}
+
+int32_t AudioDeviceModuleImpl::MaxSpeakerVolume(uint32_t* maxVolume) const {
+  CHECKinitialized_();
+  uint32_t maxVol = 0;
+  if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+  *maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MinSpeakerVolume(uint32_t* minVolume) const {
+  CHECKinitialized_();
+  uint32_t minVol = 0;
+  if (audio_device_->MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+  *minVolume = minVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  return audio_device_->SetSpeakerMute(enable);
+}
+
+int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool muted = false;
+  if (audio_device_->SpeakerMute(muted) == -1) {
+    return -1;
+  }
+  *enabled = muted;
+  RTC_LOG(LS_INFO) << "output: " << muted;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  return (audio_device_->SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool muted = false;
+  if (audio_device_->MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+  *enabled = muted;
+  RTC_LOG(LS_INFO) << "output: " << muted;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+  CHECKinitialized_();
+  return (audio_device_->SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint32_t level = 0;
+  if (audio_device_->MicrophoneVolume(level) == -1) {
+    return -1;
+  }
+  *volume = level;
+  RTC_LOG(LS_INFO) << "output: " << *volume;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable(
+    bool* available) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  if (audio_device_->RecordingIsInitialized()) {
+    RTC_LOG(LS_ERROR)
+        << "unable to set stereo mode after recording is initialized";
+    return -1;
+  }
+  if (audio_device_->SetStereoRecording(enable) == -1) {
+    if (enable) {
+      RTC_LOG(LS_WARNING) << "failed to enable stereo recording";
+    }
+    return -1;
+  }
+  int8_t nChannels(1);
+  if (enable) {
+    nChannels = 2;
+  }
+  audio_device_buffer_.SetRecordingChannels(nChannels);
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool stereo = false;
+  if (audio_device_->StereoRecording(stereo) == -1) {
+    return -1;
+  }
+  *enabled = stereo;
+  RTC_LOG(LS_INFO) << "output: " << stereo;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  if (audio_device_->PlayoutIsInitialized()) {
+    RTC_LOG(LS_ERROR)
+        << "unable to set stereo mode while playing side is initialized";
+    return -1;
+  }
+  if (audio_device_->SetStereoPlayout(enable)) {
+    RTC_LOG(LS_WARNING) << "stereo playout is not supported";
+    return -1;
+  }
+  int8_t nChannels(1);
+  if (enable) {
+    nChannels = 2;
+  }
+  audio_device_buffer_.SetPlayoutChannels(nChannels);
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool stereo = false;
+  if (audio_device_->StereoPlayout(stereo) == -1) {
+    return -1;
+  }
+  *enabled = stereo;
+  RTC_LOG(LS_INFO) << "output: " << stereo;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  bool isAvailable = false;
+  if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
+    return -1;
+  }
+  *available = isAvailable;
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MaxMicrophoneVolume(uint32_t* maxVolume) const {
+  CHECKinitialized_();
+  uint32_t maxVol(0);
+  if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+  *maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::MinMicrophoneVolume(uint32_t* minVolume) const {
+  CHECKinitialized_();
+  uint32_t minVol(0);
+  if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+  *minVolume = minVol;
+  return 0;
+}
+
+int16_t AudioDeviceModuleImpl::PlayoutDevices() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
+  RTC_LOG(LS_INFO) << "output: " << nPlayoutDevices;
+  return (int16_t)(nPlayoutDevices);
+}
+
+int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+  CHECKinitialized_();
+  return audio_device_->SetPlayoutDevice(index);
+}
+
+int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->SetPlayoutDevice(device);
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+  CHECKinitialized_();
+  if (name == NULL) {
+    return -1;
+  }
+  if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
+    return -1;
+  }
+  if (name != NULL) {
+    RTC_LOG(LS_INFO) << "output: name = " << name;
+  }
+  if (guid != NULL) {
+    RTC_LOG(LS_INFO) << "output: guid = " << guid;
+  }
+  return 0;
+}
+
+int32_t AudioDeviceModuleImpl::RecordingDeviceName(
+    uint16_t index,
+    char name[kAdmMaxDeviceNameSize],
+    char guid[kAdmMaxGuidSize]) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+  CHECKinitialized_();
+  if (name == NULL) {
+    return -1;
+  }
+  if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
+    return -1;
+  }
+  if (name != NULL) {
+    RTC_LOG(LS_INFO) << "output: name = " << name;
+  }
+  if (guid != NULL) {
+    RTC_LOG(LS_INFO) << "output: guid = " << guid;
+  }
+  return 0;
+}
+
+int16_t AudioDeviceModuleImpl::RecordingDevices() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  uint16_t nRecordingDevices = audio_device_->RecordingDevices();
+  RTC_LOG(LS_INFO) << "output: " << nRecordingDevices;
+  return (int16_t)nRecordingDevices;
+}
+
+int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+  CHECKinitialized_();
+  return audio_device_->SetRecordingDevice(index);
+}
+
+int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  return audio_device_->SetRecordingDevice(device);
+}
+
+int32_t AudioDeviceModuleImpl::InitPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (PlayoutIsInitialized()) {
+    return 0;
+  }
+  int32_t result = audio_device_->InitPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::InitRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (RecordingIsInitialized()) {
+    return 0;
+  }
+  int32_t result = audio_device_->InitRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::PlayoutIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->PlayoutIsInitialized();
+}
+
+bool AudioDeviceModuleImpl::RecordingIsInitialized() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->RecordingIsInitialized();
+}
+
+int32_t AudioDeviceModuleImpl::StartPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (Playing()) {
+    return 0;
+  }
+  audio_device_buffer_.StartPlayout();
+  int32_t result = audio_device_->StartPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::StopPlayout() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  int32_t result = audio_device_->StopPlayout();
+  audio_device_buffer_.StopPlayout();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::Playing() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->Playing();
+}
+
+int32_t AudioDeviceModuleImpl::StartRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  if (Recording()) {
+    return 0;
+  }
+  audio_device_buffer_.StartRecording();
+  int32_t result = audio_device_->StartRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+int32_t AudioDeviceModuleImpl::StopRecording() {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized_();
+  int32_t result = audio_device_->StopRecording();
+  audio_device_buffer_.StopRecording();
+  RTC_LOG(LS_INFO) << "output: " << result;
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
+                        static_cast<int>(result == 0));
+  return result;
+}
+
+bool AudioDeviceModuleImpl::Recording() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  return audio_device_->Recording();
+}
+
+int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
+    AudioTransport* audioCallback) {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return audio_device_buffer_.RegisterAudioCallback(audioCallback);
+}
+
+int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const {
+  CHECKinitialized_();
+  uint16_t delay = 0;
+  if (audio_device_->PlayoutDelay(delay) == -1) {
+    RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
+    return -1;
+  }
+  *delayMS = delay;
+  return 0;
+}
+
+bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInAECIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInAEC(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInAGC(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  CHECKinitialized__BOOL();
+  bool isAvailable = audio_device_->BuiltInNSIsAvailable();
+  RTC_LOG(LS_INFO) << "output: " << isAvailable;
+  return isAvailable;
+}
+
+int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
+  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+  CHECKinitialized_();
+  int32_t ok = audio_device_->EnableBuiltInNS(enable);
+  RTC_LOG(LS_INFO) << "output: " << ok;
+  return ok;
+}
+
+int32_t AudioDeviceModuleImpl::GetPlayoutUnderrunCount() const {
+  CHECKinitialized_();
+  int32_t underrunCount = audio_device_->GetPlayoutUnderrunCount();
+  return underrunCount;
+}
+
+#if defined(WEBRTC_IOS)
+int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  int r = audio_device_->GetPlayoutAudioParameters(params);
+  RTC_LOG(LS_INFO) << "output: " << r;
+  return r;
+}
+
+int AudioDeviceModuleImpl::GetRecordAudioParameters(
+    AudioParameters* params) const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  int r = audio_device_->GetRecordAudioParameters(params);
+  RTC_LOG(LS_INFO) << "output: " << r;
+  return r;
+}
+#endif  // WEBRTC_IOS
+
+AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return platform_type_;
+}
+
+AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
+    const {
+  RTC_LOG(LS_INFO) << __FUNCTION__;
+  return audio_layer_;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_impl.h

@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
+
+#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric;
+class AudioManager;
+
+class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
+ public:
+  enum PlatformType {
+    kPlatformNotSupported = 0,
+    kPlatformWin32 = 1,
+    kPlatformWinCe = 2,
+    kPlatformLinux = 3,
+    kPlatformMac = 4,
+    kPlatformAndroid = 5,
+    kPlatformIOS = 6,
+    // Fuchsia isn't fully supported, as there is no implementation for
+    // AudioDeviceGeneric which will be created for Fuchsia, so
+    // `CreatePlatformSpecificObjects()` call will fail unless usable
+    // implementation will be provided by the user.
+    kPlatformFuchsia = 7,
+  };
+
+  int32_t CheckPlatform();
+  int32_t CreatePlatformSpecificObjects();
+  int32_t AttachAudioBuffer();
+
+  AudioDeviceModuleImpl(AudioLayer audio_layer,
+                        TaskQueueFactory* task_queue_factory);
+  // If `create_detached` is true, created ADM can be used on another thread
+  // compared to the one on which it was created. It's useful for testing.
+  AudioDeviceModuleImpl(AudioLayer audio_layer,
+                        std::unique_ptr<AudioDeviceGeneric> audio_device,
+                        TaskQueueFactory* task_queue_factory,
+                        bool create_detached);
+  ~AudioDeviceModuleImpl() override;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+
+  // Main initializaton and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+
+  bool BuiltInAECIsAvailable() const override;
+  int32_t EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int32_t EnableBuiltInNS(bool enable) override;
+
+  // Play underrun count.
+  int32_t GetPlayoutUnderrunCount() const override;
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+#endif  // WEBRTC_IOS
+
+#if defined(WEBRTC_ANDROID)
+  // Only use this acccessor for test purposes on Android.
+  AudioManager* GetAndroidAudioManagerForTest() {
+    return audio_manager_android_.get();
+  }
+#endif
+  AudioDeviceBuffer* GetAudioDeviceBuffer() { return &audio_device_buffer_; }
+
+  int RestartPlayoutInternally() override { return -1; }
+  int RestartRecordingInternally() override { return -1; }
+  int SetPlayoutSampleRate(uint32_t sample_rate) override { return -1; }
+  int SetRecordingSampleRate(uint32_t sample_rate) override { return -1; }
+
+ private:
+  PlatformType Platform() const;
+  AudioLayer PlatformAudioLayer() const;
+
+  AudioLayer audio_layer_;
+  PlatformType platform_type_ = kPlatformNotSupported;
+  bool initialized_ = false;
+#if defined(WEBRTC_ANDROID)
+  // Should be declared first to ensure that it outlives other resources.
+  std::unique_ptr<AudioManager> audio_manager_android_;
+#endif
+  AudioDeviceBuffer audio_device_buffer_;
+  std::unique_ptr<AudioDeviceGeneric> audio_device_;
+};
+
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_name.cc

@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/audio_device_name.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+const char AudioDeviceName::kDefaultDeviceId[] = "default";
+
+AudioDeviceName::AudioDeviceName(absl::string_view device_name,
+                                 absl::string_view unique_id)
+    : device_name(device_name), unique_id(unique_id) {}
+
+bool AudioDeviceName::IsValid() {
+  return !device_name.empty() && !unique_id.empty();
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/audio_device_name.h

@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
+
+#include <deque>
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+struct AudioDeviceName {
+  // Represents a default device. Note that, on Windows there are two different
+  // types of default devices (Default and Default Communication). They can
+  // either be two different physical devices or be two different roles for one
+  // single device. Hence, this id must be combined with a "role parameter" on
+  // Windows to uniquely identify a default device.
+  static const char kDefaultDeviceId[];
+
+  AudioDeviceName() = default;
+  AudioDeviceName(absl::string_view device_name, absl::string_view unique_id);
+
+  ~AudioDeviceName() = default;
+
+  // Support copy and move.
+  AudioDeviceName(const AudioDeviceName& other) = default;
+  AudioDeviceName(AudioDeviceName&&) = default;
+  AudioDeviceName& operator=(const AudioDeviceName&) = default;
+  AudioDeviceName& operator=(AudioDeviceName&&) = default;
+
+  bool IsValid();
+
+  std::string device_name;  // Friendly name of the device.
+  std::string unique_id;    // Unique identifier for the device.
+};
+
+typedef std::deque<AudioDeviceName> AudioDeviceNames;
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/audio_device_dummy.cc

@@ -0,0 +1,226 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/audio_device_dummy.h"
+
+namespace webrtc {
+
+int32_t AudioDeviceDummy::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  return -1;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceDummy::Init() {
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceDummy::Terminate() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Initialized() const {
+  return true;
+}
+
+int16_t AudioDeviceDummy::PlayoutDevices() {
+  return -1;
+}
+
+int16_t AudioDeviceDummy::RecordingDevices() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::RecordingDeviceName(uint16_t index,
+                                              char name[kAdmMaxDeviceNameSize],
+                                              char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetPlayoutDevice(uint16_t index) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetRecordingDevice(uint16_t index) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::InitPlayout() {
+  return -1;
+}
+
+bool AudioDeviceDummy::PlayoutIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::RecordingIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::InitRecording() {
+  return -1;
+}
+
+bool AudioDeviceDummy::RecordingIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::StartPlayout() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StopPlayout() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Playing() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::StartRecording() {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StopRecording() {
+  return 0;
+}
+
+bool AudioDeviceDummy::Recording() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::InitSpeaker() {
+  return -1;
+}
+
+bool AudioDeviceDummy::SpeakerIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::InitMicrophone() {
+  return -1;
+}
+
+bool AudioDeviceDummy::MicrophoneIsInitialized() const {
+  return false;
+}
+
+int32_t AudioDeviceDummy::SpeakerVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MinMicrophoneVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetSpeakerMute(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SpeakerMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetMicrophoneMute(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::MicrophoneMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available) {
+  return -1;
+}
+int32_t AudioDeviceDummy::SetStereoPlayout(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoPlayout(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoRecordingIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::SetStereoRecording(bool enable) {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const {
+  return -1;
+}
+
+int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const {
+  return -1;
+}
+
+void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/audio_device_dummy.h

@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_
+
+#include <stdint.h>
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceDummy : public AudioDeviceGeneric {
+ public:
+  AudioDeviceDummy() {}
+  virtual ~AudioDeviceDummy() {}
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/file_audio_device.cc

@@ -0,0 +1,508 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/file_audio_device.h"
+
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+const int kRecordingFixedSampleRate = 48000;
+const size_t kRecordingNumChannels = 2;
+const int kPlayoutFixedSampleRate = 48000;
+const size_t kPlayoutNumChannels = 2;
+const size_t kPlayoutBufferSize =
+    kPlayoutFixedSampleRate / 100 * kPlayoutNumChannels * 2;
+const size_t kRecordingBufferSize =
+    kRecordingFixedSampleRate / 100 * kRecordingNumChannels * 2;
+
+FileAudioDevice::FileAudioDevice(absl::string_view inputFilename,
+                                 absl::string_view outputFilename)
+    : _ptrAudioBuffer(NULL),
+      _recordingBuffer(NULL),
+      _playoutBuffer(NULL),
+      _recordingFramesLeft(0),
+      _playoutFramesLeft(0),
+      _recordingBufferSizeIn10MS(0),
+      _recordingFramesIn10MS(0),
+      _playoutFramesIn10MS(0),
+      _playing(false),
+      _recording(false),
+      _lastCallPlayoutMillis(0),
+      _lastCallRecordMillis(0),
+      _outputFilename(outputFilename),
+      _inputFilename(inputFilename) {}
+
+FileAudioDevice::~FileAudioDevice() {}
+
+int32_t FileAudioDevice::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  return -1;
+}
+
+AudioDeviceGeneric::InitStatus FileAudioDevice::Init() {
+  return InitStatus::OK;
+}
+
+int32_t FileAudioDevice::Terminate() {
+  return 0;
+}
+
+bool FileAudioDevice::Initialized() const {
+  return true;
+}
+
+int16_t FileAudioDevice::PlayoutDevices() {
+  return 1;
+}
+
+int16_t FileAudioDevice::RecordingDevices() {
+  return 1;
+}
+
+int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index,
+                                           char name[kAdmMaxDeviceNameSize],
+                                           char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::RecordingDeviceName(uint16_t index,
+                                             char name[kAdmMaxDeviceNameSize],
+                                             char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(uint16_t index) {
+  if (index == 0) {
+    _playout_index = index;
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(uint16_t index) {
+  if (index == 0) {
+    _record_index = index;
+    return _record_index;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::PlayoutIsAvailable(bool& available) {
+  if (_playout_index == 0) {
+    available = true;
+    return _playout_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitPlayout() {
+  MutexLock lock(&mutex_);
+
+  if (_playing) {
+    return -1;
+  }
+
+  _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
+
+  if (_ptrAudioBuffer) {
+    // Update webrtc audio buffer with the selected parameters
+    _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
+    _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::PlayoutIsInitialized() const {
+  return _playoutFramesIn10MS != 0;
+}
+
+int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
+  if (_record_index == 0) {
+    available = true;
+    return _record_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitRecording() {
+  MutexLock lock(&mutex_);
+
+  if (_recording) {
+    return -1;
+  }
+
+  _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);
+
+  if (_ptrAudioBuffer) {
+    _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
+    _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::RecordingIsInitialized() const {
+  return _recordingFramesIn10MS != 0;
+}
+
+int32_t FileAudioDevice::StartPlayout() {
+  if (_playing) {
+    return 0;
+  }
+
+  _playing = true;
+  _playoutFramesLeft = 0;
+
+  if (!_playoutBuffer) {
+    _playoutBuffer = new int8_t[kPlayoutBufferSize];
+  }
+  if (!_playoutBuffer) {
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  if (!_outputFilename.empty()) {
+    _outputFile = FileWrapper::OpenWriteOnly(_outputFilename);
+    if (!_outputFile.is_open()) {
+      RTC_LOG(LS_ERROR) << "Failed to open playout file: " << _outputFilename;
+      _playing = false;
+      delete[] _playoutBuffer;
+      _playoutBuffer = NULL;
+      return -1;
+    }
+  }
+
+  _ptrThreadPlay = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (PlayThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_play_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  RTC_LOG(LS_INFO) << "Started playout capture to output file: "
+                   << _outputFilename;
+  return 0;
+}
+
+int32_t FileAudioDevice::StopPlayout() {
+  {
+    MutexLock lock(&mutex_);
+    _playing = false;
+  }
+
+  // stop playout thread first
+  if (!_ptrThreadPlay.empty())
+    _ptrThreadPlay.Finalize();
+
+  MutexLock lock(&mutex_);
+
+  _playoutFramesLeft = 0;
+  delete[] _playoutBuffer;
+  _playoutBuffer = NULL;
+  _outputFile.Close();
+
+  RTC_LOG(LS_INFO) << "Stopped playout capture to output file: "
+                   << _outputFilename;
+  return 0;
+}
+
+bool FileAudioDevice::Playing() const {
+  return _playing;
+}
+
+int32_t FileAudioDevice::StartRecording() {
+  _recording = true;
+
+  // Make sure we only create the buffer once.
+  _recordingBufferSizeIn10MS =
+      _recordingFramesIn10MS * kRecordingNumChannels * 2;
+  if (!_recordingBuffer) {
+    _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+  }
+
+  if (!_inputFilename.empty()) {
+    _inputFile = FileWrapper::OpenReadOnly(_inputFilename);
+    if (!_inputFile.is_open()) {
+      RTC_LOG(LS_ERROR) << "Failed to open audio input file: "
+                        << _inputFilename;
+      _recording = false;
+      delete[] _recordingBuffer;
+      _recordingBuffer = NULL;
+      return -1;
+    }
+  }
+
+  _ptrThreadRec = rtc::PlatformThread::SpawnJoinable(
+      [this] {
+        while (RecThreadProcess()) {
+        }
+      },
+      "webrtc_audio_module_capture_thread",
+      rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime));
+
+  RTC_LOG(LS_INFO) << "Started recording from input file: " << _inputFilename;
+
+  return 0;
+}
+
+int32_t FileAudioDevice::StopRecording() {
+  {
+    MutexLock lock(&mutex_);
+    _recording = false;
+  }
+
+  if (!_ptrThreadRec.empty())
+    _ptrThreadRec.Finalize();
+
+  MutexLock lock(&mutex_);
+  _recordingFramesLeft = 0;
+  if (_recordingBuffer) {
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+  }
+  _inputFile.Close();
+
+  RTC_LOG(LS_INFO) << "Stopped recording from input file: " << _inputFilename;
+  return 0;
+}
+
+bool FileAudioDevice::Recording() const {
+  return _recording;
+}
+
+int32_t FileAudioDevice::InitSpeaker() {
+  return -1;
+}
+
+bool FileAudioDevice::SpeakerIsInitialized() const {
+  return false;
+}
+
+int32_t FileAudioDevice::InitMicrophone() {
+  return 0;
+}
+
+bool FileAudioDevice::MicrophoneIsInitialized() const {
+  return true;
+}
+
+int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinMicrophoneVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetSpeakerMute(bool enable) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneMute(bool enable) {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+int32_t FileAudioDevice::SetStereoPlayout(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoPlayout(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::SetStereoRecording(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecording(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
+  return 0;
+}
+
+void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  MutexLock lock(&mutex_);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+bool FileAudioDevice::PlayThreadProcess() {
+  if (!_playing) {
+    return false;
+  }
+  int64_t currentTime = rtc::TimeMillis();
+  mutex_.Lock();
+
+  if (_lastCallPlayoutMillis == 0 ||
+      currentTime - _lastCallPlayoutMillis >= 10) {
+    mutex_.Unlock();
+    _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+    mutex_.Lock();
+
+    _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+    RTC_DCHECK_EQ(_playoutFramesIn10MS, _playoutFramesLeft);
+    if (_outputFile.is_open()) {
+      _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
+    }
+    _lastCallPlayoutMillis = currentTime;
+  }
+  _playoutFramesLeft = 0;
+  mutex_.Unlock();
+
+  int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+  if (deltaTimeMillis < 10) {
+    SleepMs(10 - deltaTimeMillis);
+  }
+
+  return true;
+}
+
+bool FileAudioDevice::RecThreadProcess() {
+  if (!_recording) {
+    return false;
+  }
+
+  int64_t currentTime = rtc::TimeMillis();
+  mutex_.Lock();
+
+  if (_lastCallRecordMillis == 0 || currentTime - _lastCallRecordMillis >= 10) {
+    if (_inputFile.is_open()) {
+      if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
+        _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                           _recordingFramesIn10MS);
+      } else {
+        _inputFile.Rewind();
+      }
+      _lastCallRecordMillis = currentTime;
+      mutex_.Unlock();
+      _ptrAudioBuffer->DeliverRecordedData();
+      mutex_.Lock();
+    }
+  }
+
+  mutex_.Unlock();
+
+  int64_t deltaTimeMillis = rtc::TimeMillis() - currentTime;
+  if (deltaTimeMillis < 10) {
+    SleepMs(10 - deltaTimeMillis);
+  }
+
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/file_audio_device.h

@@ -0,0 +1,163 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
+#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// This is a fake audio device which plays audio from a file as its microphone
+// and plays out into a file.
+class FileAudioDevice : public AudioDeviceGeneric {
+ public:
+  // Constructs a file audio device with `id`. It will read audio from
+  // `inputFilename` and record output audio to `outputFilename`.
+  //
+  // The input file should be a readable 48k stereo raw file, and the output
+  // file should point to a writable location. The output format will also be
+  // 48k stereo raw audio.
+  FileAudioDevice(absl::string_view inputFilename,
+                  absl::string_view outputFilename);
+  virtual ~FileAudioDevice();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+ private:
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  int32_t _playout_index;
+  int32_t _record_index;
+  AudioDeviceBuffer* _ptrAudioBuffer;
+  int8_t* _recordingBuffer;  // In bytes.
+  int8_t* _playoutBuffer;    // In bytes.
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+  Mutex mutex_;
+
+  size_t _recordingBufferSizeIn10MS;
+  size_t _recordingFramesIn10MS;
+  size_t _playoutFramesIn10MS;
+
+  rtc::PlatformThread _ptrThreadRec;
+  rtc::PlatformThread _ptrThreadPlay;
+
+  bool _playing;
+  bool _recording;
+  int64_t _lastCallPlayoutMillis;
+  int64_t _lastCallRecordMillis;
+
+  FileWrapper _outputFile;
+  FileWrapper _inputFile;
+  std::string _outputFilename;
+  std::string _inputFilename;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_FILE_AUDIO_DEVICE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/file_audio_device_factory.cc

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/dummy/file_audio_device_factory.h"
+
+#include <stdio.h>
+
+#include <cstdlib>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_device/dummy/file_audio_device.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+
+bool FileAudioDeviceFactory::_isConfigured = false;
+char FileAudioDeviceFactory::_inputAudioFilename[MAX_FILENAME_LEN] = "";
+char FileAudioDeviceFactory::_outputAudioFilename[MAX_FILENAME_LEN] = "";
+
+FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice() {
+  // Bail out here if the files haven't been set explicitly.
+  // audio_device_impl.cc should then fall back to dummy audio.
+  if (!_isConfigured) {
+    RTC_LOG(LS_WARNING)
+        << "WebRTC configured with WEBRTC_DUMMY_FILE_DEVICES but "
+           "no device files supplied. Will fall back to dummy "
+           "audio.";
+
+    return nullptr;
+  }
+  return new FileAudioDevice(_inputAudioFilename, _outputAudioFilename);
+}
+
+void FileAudioDeviceFactory::SetFilenamesToUse(
+    absl::string_view inputAudioFilename,
+    absl::string_view outputAudioFilename) {
+#ifdef WEBRTC_DUMMY_FILE_DEVICES
+  RTC_DCHECK_LT(inputAudioFilename.size(), MAX_FILENAME_LEN);
+  RTC_DCHECK_LT(outputAudioFilename.size(), MAX_FILENAME_LEN);
+
+  // Copy the strings since we don't know the lifetime of the input pointers.
+  rtc::strcpyn(_inputAudioFilename, MAX_FILENAME_LEN, inputAudioFilename);
+  rtc::strcpyn(_outputAudioFilename, MAX_FILENAME_LEN, outputAudioFilename);
+  _isConfigured = true;
+#else
+  // Sanity: must be compiled with the right define to run this.
+  printf(
+      "Trying to use dummy file devices, but is not compiled "
+      "with WEBRTC_DUMMY_FILE_DEVICES. Bailing out.\n");
+  std::exit(1);
+#endif
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/dummy/file_audio_device_factory.h

@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
+#define AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_
+
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+class FileAudioDevice;
+
+// This class is used by audio_device_impl.cc when WebRTC is compiled with
+// WEBRTC_DUMMY_FILE_DEVICES. The application must include this file and set the
+// filenames to use before the audio device module is initialized. This is
+// intended for test tools which use the audio device module.
+class FileAudioDeviceFactory {
+ public:
+  static FileAudioDevice* CreateFileAudioDevice();
+
+  // The input file must be a readable 48k stereo raw file. The output
+  // file must be writable. The strings will be copied.
+  static void SetFilenamesToUse(absl::string_view inputAudioFilename,
+                                absl::string_view outputAudioFilename);
+
+ private:
+  enum : uint32_t { MAX_FILENAME_LEN = 512 };
+  static bool _isConfigured;
+  static char _inputAudioFilename[MAX_FILENAME_LEN];
+  static char _outputAudioFilename[MAX_FILENAME_LEN];
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/fine_audio_buffer.cc

@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/fine_audio_buffer.h"
+
+#include <cstdint>
+#include <cstring>
+
+#include "api/array_view.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer)
+    : audio_device_buffer_(audio_device_buffer),
+      playout_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->PlayoutSampleRate() * 10 / 1000)),
+      record_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
+          audio_device_buffer->RecordingSampleRate() * 10 / 1000)),
+      playout_channels_(audio_device_buffer->PlayoutChannels()),
+      record_channels_(audio_device_buffer->RecordingChannels()) {
+  RTC_DCHECK(audio_device_buffer_);
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  if (IsReadyForPlayout()) {
+    RTC_DLOG(LS_INFO) << "playout_samples_per_channel_10ms: "
+                      << playout_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "playout_channels: " << playout_channels_;
+  }
+  if (IsReadyForRecord()) {
+    RTC_DLOG(LS_INFO) << "record_samples_per_channel_10ms: "
+                      << record_samples_per_channel_10ms_;
+    RTC_DLOG(LS_INFO) << "record_channels: " << record_channels_;
+  }
+}
+
+FineAudioBuffer::~FineAudioBuffer() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+}
+
+void FineAudioBuffer::ResetPlayout() {
+  playout_buffer_.Clear();
+}
+
+void FineAudioBuffer::ResetRecord() {
+  record_buffer_.Clear();
+}
+
+bool FineAudioBuffer::IsReadyForPlayout() const {
+  return playout_samples_per_channel_10ms_ > 0 && playout_channels_ > 0;
+}
+
+bool FineAudioBuffer::IsReadyForRecord() const {
+  return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
+}
+
+void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
+                                     int playout_delay_ms) {
+  RTC_DCHECK(IsReadyForPlayout());
+  // Ask WebRTC for new data in chunks of 10ms until we have enough to
+  // fulfill the request. It is possible that the buffer already contains
+  // enough samples from the last round.
+  while (playout_buffer_.size() < audio_buffer.size()) {
+    // Get 10ms decoded audio from WebRTC. The ADB knows about number of
+    // channels; hence we can ask for number of samples per channel here.
+    if (audio_device_buffer_->RequestPlayoutData(
+            playout_samples_per_channel_10ms_) ==
+        static_cast<int32_t>(playout_samples_per_channel_10ms_)) {
+      // Append 10ms to the end of the local buffer taking number of channels
+      // into account.
+      const size_t num_elements_10ms =
+          playout_channels_ * playout_samples_per_channel_10ms_;
+      const size_t written_elements = playout_buffer_.AppendData(
+          num_elements_10ms, [&](rtc::ArrayView<int16_t> buf) {
+            const size_t samples_per_channel_10ms =
+                audio_device_buffer_->GetPlayoutData(buf.data());
+            return playout_channels_ * samples_per_channel_10ms;
+          });
+      RTC_DCHECK_EQ(num_elements_10ms, written_elements);
+    } else {
+      // Provide silence if AudioDeviceBuffer::RequestPlayoutData() fails.
+      // Can e.g. happen when an AudioTransport has not been registered.
+      const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+      std::memset(audio_buffer.data(), 0, num_bytes);
+      return;
+    }
+  }
+
+  // Provide the requested number of bytes to the consumer.
+  const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
+  memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
+  // Move remaining samples to start of buffer to prepare for next round.
+  memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(),
+          (playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t));
+  playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size());
+  // Cache playout latency for usage in DeliverRecordedData();
+  playout_delay_ms_ = playout_delay_ms;
+}
+
+void FineAudioBuffer::DeliverRecordedData(
+    rtc::ArrayView<const int16_t> audio_buffer,
+    int record_delay_ms,
+    absl::optional<int64_t> capture_time_ns) {
+  RTC_DCHECK(IsReadyForRecord());
+  // Always append new data and grow the buffer when needed.
+  record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
+  // Consume samples from buffer in chunks of 10ms until there is not
+  // enough data left. The number of remaining samples in the cache is given by
+  // the new size of the internal `record_buffer_`.
+  const size_t num_elements_10ms =
+      record_channels_ * record_samples_per_channel_10ms_;
+  while (record_buffer_.size() >= num_elements_10ms) {
+    audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
+                                            record_samples_per_channel_10ms_,
+                                            capture_time_ns);
+    audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
+    audio_device_buffer_->DeliverRecordedData();
+    memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
+            (record_buffer_.size() - num_elements_10ms) * sizeof(int16_t));
+    record_buffer_.SetSize(record_buffer_.size() - num_elements_10ms);
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/fine_audio_buffer.h

@@ -0,0 +1,103 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
+
+#include <cstddef>
+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDeviceBuffer;
+
+// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM
+// audio samples corresponding to 10ms of data. It then allows for this data
+// to be pulled in a finer or coarser granularity. I.e. interacting with this
+// class instead of directly with the AudioDeviceBuffer one can ask for any
+// number of audio data samples. This class also ensures that audio data can be
+// delivered to the ADB in 10ms chunks when the size of the provided audio
+// buffers differs from 10ms.
+// As an example: calling DeliverRecordedData() with 5ms buffers will deliver
+// accumulated 10ms worth of data to the ADB every second call.
+class FineAudioBuffer {
+ public:
+  // `device_buffer` is a buffer that provides 10ms of audio data.
+  FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer);
+  ~FineAudioBuffer();
+
+  // Clears buffers and counters dealing with playout and/or recording.
+  void ResetPlayout();
+  void ResetRecord();
+
+  // Utility methods which returns true if valid parameters are acquired at
+  // constructions.
+  bool IsReadyForPlayout() const;
+  bool IsReadyForRecord() const;
+
+  // Copies audio samples into `audio_buffer` where number of requested
+  // elements is specified by `audio_buffer.size()`. The producer will always
+  // fill up the audio buffer and if no audio exists, the buffer will contain
+  // silence instead. The provided delay estimate in `playout_delay_ms` should
+  // contain an estimate of the latency between when an audio frame is read from
+  // WebRTC and when it is played out on the speaker.
+  void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
+                      int playout_delay_ms);
+
+  // Consumes the audio data in `audio_buffer` and sends it to the WebRTC layer
+  // in chunks of 10ms. The sum of the provided delay estimate in
+  // `record_delay_ms` and the latest `playout_delay_ms` in GetPlayoutData()
+  // are given to the AEC in the audio processing module.
+  // They can be fixed values on most platforms and they are ignored if an
+  // external (hardware/built-in) AEC is used.
+  // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
+  // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
+  // cache. Call #3 restarts the scheme above.
+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
+                           int record_delay_ms) {
+    DeliverRecordedData(audio_buffer, record_delay_ms, absl::nullopt);
+  }
+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
+                           int record_delay_ms,
+                           absl::optional<int64_t> capture_time_ns);
+
+ private:
+  // Device buffer that works with 10ms chunks of data both for playout and
+  // for recording. I.e., the WebRTC side will always be asked for audio to be
+  // played out in 10ms chunks and recorded audio will be sent to WebRTC in
+  // 10ms chunks as well. This raw pointer is owned by the constructor of this
+  // class and the owner must ensure that the pointer is valid during the life-
+  // time of this object.
+  AudioDeviceBuffer* const audio_device_buffer_;
+  // Number of audio samples per channel per 10ms. Set once at construction
+  // based on parameters in `audio_device_buffer`.
+  const size_t playout_samples_per_channel_10ms_;
+  const size_t record_samples_per_channel_10ms_;
+  // Number of audio channels. Set once at construction based on parameters in
+  // `audio_device_buffer`.
+  const size_t playout_channels_;
+  const size_t record_channels_;
+  // Storage for output samples from which a consumer can read audio buffers
+  // in any size using GetPlayoutData().
+  rtc::BufferT<int16_t> playout_buffer_;
+  // Storage for input samples that are about to be delivered to the WebRTC
+  // ADB or remains from the last successful delivery of a 10ms audio buffer.
+  rtc::BufferT<int16_t> record_buffer_;
+  // Contains latest delay estimate given to GetPlayoutData().
+  int playout_delay_ms_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device.h

@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
+
+#include "absl/types/optional.h"
+#include "api/ref_count.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+
+namespace webrtc {
+
+class AudioDeviceModuleForTest;
+
+class AudioDeviceModule : public webrtc::RefCountInterface {
+ public:
+  enum AudioLayer {
+    kPlatformDefaultAudio = 0,
+    kWindowsCoreAudio,
+    kWindowsCoreAudio2,
+    kLinuxAlsaAudio,
+    kLinuxPulseAudio,
+    kAndroidJavaAudio,
+    kAndroidOpenSLESAudio,
+    kAndroidJavaInputAndOpenSLESOutputAudio,
+    kAndroidAAudioAudio,
+    kAndroidJavaInputAndAAudioOutputAudio,
+    kDummyAudio,
+    kAndroidScreenAudio,
+    kAndroidMergedScreenAudio
+  };
+
+  enum WindowsDeviceType {
+    kDefaultCommunicationDevice = -1,
+    kDefaultDevice = -2
+  };
+
+  struct Stats {
+    // The fields below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#playoutstats-dict*
+    double synthesized_samples_duration_s = 0;
+    uint64_t synthesized_samples_events = 0;
+    double total_samples_duration_s = 0;
+    double total_playout_delay_s = 0;
+    uint64_t total_samples_count = 0;
+  };
+
+ public:
+  // Creates a default ADM for usage in production code.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+  // Creates an ADM with support for extra test methods. Don't use this factory
+  // in production code.
+  static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
+      AudioLayer audio_layer,
+      TaskQueueFactory* task_queue_factory);
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
+
+  // Full-duplex transportation of PCM audio
+  virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
+
+  // Main initialization and termination
+  virtual int32_t Init() = 0;
+  virtual int32_t Terminate() = 0;
+  virtual bool Initialized() const = 0;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() = 0;
+  virtual int16_t RecordingDevices() = 0;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) = 0;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) = 0;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
+  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
+  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
+  virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool* available) = 0;
+  virtual int32_t InitPlayout() = 0;
+  virtual bool PlayoutIsInitialized() const = 0;
+  virtual int32_t RecordingIsAvailable(bool* available) = 0;
+  virtual int32_t InitRecording() = 0;
+  virtual bool RecordingIsInitialized() const = 0;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() = 0;
+  virtual int32_t StopPlayout() = 0;
+  virtual bool Playing() const = 0;
+  virtual int32_t StartRecording() = 0;
+  virtual int32_t StopRecording() = 0;
+  virtual bool Recording() const = 0;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() = 0;
+  virtual bool SpeakerIsInitialized() const = 0;
+  virtual int32_t InitMicrophone() = 0;
+  virtual bool MicrophoneIsInitialized() const = 0;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
+  virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
+  virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
+  virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
+  virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetSpeakerMute(bool enable) = 0;
+  virtual int32_t SpeakerMute(bool* enabled) const = 0;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
+  virtual int32_t SetMicrophoneMute(bool enable) = 0;
+  virtual int32_t MicrophoneMute(bool* enabled) const = 0;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoPlayout(bool enable) = 0;
+  virtual int32_t StereoPlayout(bool* enabled) const = 0;
+  virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
+  virtual int32_t SetStereoRecording(bool enable) = 0;
+  virtual int32_t StereoRecording(bool* enabled) const = 0;
+
+  // Playout delay
+  virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
+
+  // Only supported on Android.
+  virtual bool BuiltInAECIsAvailable() const = 0;
+  virtual bool BuiltInAGCIsAvailable() const = 0;
+  virtual bool BuiltInNSIsAvailable() const = 0;
+
+  // Enables the built-in audio effects. Only supported on Android.
+  virtual int32_t EnableBuiltInAEC(bool enable) = 0;
+  virtual int32_t EnableBuiltInAGC(bool enable) = 0;
+  virtual int32_t EnableBuiltInNS(bool enable) = 0;
+
+  // Play underrun count. Only supported on Android.
+  // TODO(alexnarest): Make it abstract after upstream projects support it.
+  virtual int32_t GetPlayoutUnderrunCount() const { return -1; }
+
+  // Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will
+  // not be present in the stats.
+  virtual absl::optional<Stats> GetStats() const { return absl::nullopt; }
+
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
+  virtual int GetPlayoutAudioParameters(AudioParameters* params) const = 0;
+  virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
+#endif  // WEBRTC_IOS
+
+ protected:
+  ~AudioDeviceModule() override {}
+};
+
+// Extends the default ADM interface with some extra test methods.
+// Intended for usage in tests only and requires a unique factory method.
+class AudioDeviceModuleForTest : public AudioDeviceModule {
+ public:
+  // Triggers internal restart sequences of audio streaming. Can be used by
+  // tests to emulate events corresponding to e.g. removal of an active audio
+  // device or other actions which causes the stream to be disconnected.
+  virtual int RestartPlayoutInternally() = 0;
+  virtual int RestartRecordingInternally() = 0;
+
+  virtual int SetPlayoutSampleRate(uint32_t sample_rate) = 0;
+  virtual int SetRecordingSampleRate(uint32_t sample_rate) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device_data_observer.h

@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+// This interface will capture the raw PCM data of both the local captured as
+// well as the mixed/rendered remote audio.
+class AudioDeviceDataObserver {
+ public:
+  virtual void OnCaptureData(const void* audio_samples,
+                             size_t num_samples,
+                             size_t bytes_per_sample,
+                             size_t num_channels,
+                             uint32_t samples_per_sec) = 0;
+
+  virtual void OnRenderData(const void* audio_samples,
+                            size_t num_samples,
+                            size_t bytes_per_sample,
+                            size_t num_channels,
+                            uint32_t samples_per_sec) = 0;
+
+  AudioDeviceDataObserver() = default;
+  virtual ~AudioDeviceDataObserver() = default;
+};
+
+// Creates an ADMWrapper around an ADM instance that registers
+// the provided AudioDeviceDataObserver.
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    std::unique_ptr<AudioDeviceDataObserver> observer);
+
+// Creates an ADMWrapper around an ADM instance that registers
+// the provided AudioDeviceDataObserver.
+ABSL_DEPRECATED("")
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    rtc::scoped_refptr<AudioDeviceModule> impl,
+    AudioDeviceDataObserver* observer);
+
+// Creates an ADM instance with AudioDeviceDataObserver registered.
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<AudioDeviceDataObserver> observer);
+
+// Creates an ADM instance with AudioDeviceDataObserver registered.
+ABSL_DEPRECATED("")
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
+    AudioDeviceModule::AudioLayer audio_layer,
+    TaskQueueFactory* task_queue_factory,
+    AudioDeviceDataObserver* observer);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DATA_OBSERVER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device_default.h

@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
+
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+namespace webrtc_impl {
+
+// AudioDeviceModuleDefault template adds default implementation for all
+// AudioDeviceModule methods to the class, which inherits from
+// AudioDeviceModuleDefault<T>.
+template <typename T>
+class AudioDeviceModuleDefault : public T {
+ public:
+  AudioDeviceModuleDefault() {}
+  virtual ~AudioDeviceModuleDefault() {}
+
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
+    return 0;
+  }
+  int32_t Init() override { return 0; }
+  int32_t InitSpeaker() override { return 0; }
+  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+  int32_t SetStereoPlayout(bool enable) override { return 0; }
+  int32_t StopPlayout() override { return 0; }
+  int32_t InitMicrophone() override { return 0; }
+  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+  int32_t SetStereoRecording(bool enable) override { return 0; }
+  int32_t StopRecording() override { return 0; }
+
+  int32_t Terminate() override { return 0; }
+
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer* audioLayer) const override {
+    return 0;
+  }
+  bool Initialized() const override { return true; }
+  int16_t PlayoutDevices() override { return 0; }
+  int16_t RecordingDevices() override { return 0; }
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
+  int32_t InitPlayout() override { return 0; }
+  bool PlayoutIsInitialized() const override { return true; }
+  int32_t RecordingIsAvailable(bool* available) override { return 0; }
+  int32_t InitRecording() override { return 0; }
+  bool RecordingIsInitialized() const override { return true; }
+  int32_t StartPlayout() override { return 0; }
+  bool Playing() const override { return false; }
+  int32_t StartRecording() override { return 0; }
+  bool Recording() const override { return false; }
+  bool SpeakerIsInitialized() const override { return true; }
+  bool MicrophoneIsInitialized() const override { return true; }
+  int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
+  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override { return 0; }
+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
+  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetSpeakerMute(bool enable) override { return 0; }
+  int32_t SpeakerMute(bool* enabled) const override { return 0; }
+  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
+  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
+    *available = false;
+    return 0;
+  }
+  int32_t StereoPlayout(bool* enabled) const override { return 0; }
+  int32_t StereoRecordingIsAvailable(bool* available) const override {
+    *available = false;
+    return 0;
+  }
+  int32_t StereoRecording(bool* enabled) const override { return 0; }
+  int32_t PlayoutDelay(uint16_t* delayMS) const override {
+    *delayMS = 0;
+    return 0;
+  }
+  bool BuiltInAECIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
+  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
+  bool BuiltInNSIsAvailable() const override { return false; }
+  int32_t EnableBuiltInNS(bool enable) override { return -1; }
+
+  int32_t GetPlayoutUnderrunCount() const override { return -1; }
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+  int GetRecordAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+#endif  // WEBRTC_IOS
+};
+
+}  // namespace webrtc_impl
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device_defines.h

@@ -0,0 +1,177 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
+
+#include <stddef.h>
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+static const int kAdmMaxDeviceNameSize = 128;
+static const int kAdmMaxFileNameSize = 512;
+static const int kAdmMaxGuidSize = 128;
+
+static const int kAdmMinPlayoutBufferSizeMs = 10;
+static const int kAdmMaxPlayoutBufferSizeMs = 250;
+
+// ----------------------------------------------------------------------------
+//  AudioTransport
+// ----------------------------------------------------------------------------
+
+class AudioTransport {
+ public:
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                          size_t nSamples,
+                                          size_t nBytesPerSample,
+                                          size_t nChannels,
+                                          uint32_t samplesPerSec,
+                                          uint32_t totalDelayMS,
+                                          int32_t clockDrift,
+                                          uint32_t currentMicLevel,
+                                          bool keyPressed,
+                                          uint32_t& newMicLevel) = 0;  // NOLINT
+
+  virtual int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimatedCaptureTimeNS) {  // NOLINT
+    // TODO(webrtc:13620) Make the default behaver of the new API to behave as
+    // the old API. This can be pure virtual if all uses of the old API is
+    // removed.
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samplesPerSec,
+        totalDelayMS, clockDrift, currentMicLevel, keyPressed, newMicLevel);
+  }
+
+  // Implementation has to setup safe values for all specified out parameters.
+  virtual int32_t NeedMorePlayData(size_t nSamples,
+                                   size_t nBytesPerSample,
+                                   size_t nChannels,
+                                   uint32_t samplesPerSec,
+                                   void* audioSamples,
+                                   size_t& nSamplesOut,  // NOLINT
+                                   int64_t* elapsed_time_ms,
+                                   int64_t* ntp_time_ms) = 0;  // NOLINT
+
+  // Method to pull mixed render audio data from all active VoE channels.
+  // The data will not be passed as reference for audio processing internally.
+  virtual void PullRenderData(int bits_per_sample,
+                              int sample_rate,
+                              size_t number_of_channels,
+                              size_t number_of_frames,
+                              void* audio_data,
+                              int64_t* elapsed_time_ms,
+                              int64_t* ntp_time_ms) = 0;
+
+ protected:
+  virtual ~AudioTransport() {}
+};
+
+// Helper class for storage of fundamental audio parameters such as sample rate,
+// number of channels, native buffer size etc.
+// Note that one audio frame can contain more than one channel sample and each
+// sample is assumed to be a 16-bit PCM sample. Hence, one audio frame in
+// stereo contains 2 * (16/8) = 4 bytes of data.
+class AudioParameters {
+ public:
+  // This implementation does only support 16-bit PCM samples.
+  static const size_t kBitsPerSample = 16;
+  AudioParameters()
+      : sample_rate_(0),
+        channels_(0),
+        frames_per_buffer_(0),
+        frames_per_10ms_buffer_(0) {}
+  AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer)
+      : sample_rate_(sample_rate),
+        channels_(channels),
+        frames_per_buffer_(frames_per_buffer),
+        frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {}
+  void reset(int sample_rate, size_t channels, size_t frames_per_buffer) {
+    sample_rate_ = sample_rate;
+    channels_ = channels;
+    frames_per_buffer_ = frames_per_buffer;
+    frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100);
+  }
+  size_t bits_per_sample() const { return kBitsPerSample; }
+  void reset(int sample_rate, size_t channels, double buffer_duration) {
+    reset(sample_rate, channels,
+          static_cast<size_t>(sample_rate * buffer_duration + 0.5));
+  }
+  void reset(int sample_rate, size_t channels) {
+    reset(sample_rate, channels, static_cast<size_t>(0));
+  }
+  int sample_rate() const { return sample_rate_; }
+  size_t channels() const { return channels_; }
+  size_t frames_per_buffer() const { return frames_per_buffer_; }
+  size_t frames_per_10ms_buffer() const { return frames_per_10ms_buffer_; }
+  size_t GetBytesPerFrame() const { return channels_ * kBitsPerSample / 8; }
+  size_t GetBytesPerBuffer() const {
+    return frames_per_buffer_ * GetBytesPerFrame();
+  }
+  // The WebRTC audio device buffer (ADB) only requires that the sample rate
+  // and number of channels are configured. Hence, to be "valid", only these
+  // two attributes must be set.
+  bool is_valid() const { return ((sample_rate_ > 0) && (channels_ > 0)); }
+  // Most platforms also require that a native buffer size is defined.
+  // An audio parameter instance is considered to be "complete" if it is both
+  // "valid" (can be used by the ADB) and also has a native frame size.
+  bool is_complete() const { return (is_valid() && (frames_per_buffer_ > 0)); }
+  size_t GetBytesPer10msBuffer() const {
+    return frames_per_10ms_buffer_ * GetBytesPerFrame();
+  }
+  double GetBufferSizeInMilliseconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return frames_per_buffer_ / (sample_rate_ / 1000.0);
+  }
+  double GetBufferSizeInSeconds() const {
+    if (sample_rate_ == 0)
+      return 0.0;
+    return static_cast<double>(frames_per_buffer_) / (sample_rate_);
+  }
+  std::string ToString() const {
+    char ss_buf[1024];
+    rtc::SimpleStringBuilder ss(ss_buf);
+    ss << "AudioParameters: ";
+    ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
+    ss << ", frames_per_buffer=" << frames_per_buffer();
+    ss << ", frames_per_10ms_buffer=" << frames_per_10ms_buffer();
+    ss << ", bytes_per_frame=" << GetBytesPerFrame();
+    ss << ", bytes_per_buffer=" << GetBytesPerBuffer();
+    ss << ", bytes_per_10ms_buffer=" << GetBytesPer10msBuffer();
+    ss << ", size_in_ms=" << GetBufferSizeInMilliseconds();
+    return ss.str();
+  }
+
+ private:
+  int sample_rate_;
+  size_t channels_;
+  size_t frames_per_buffer_;
+  size_t frames_per_10ms_buffer_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device_factory.cc

@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/audio_device_factory.h"
+
+#include <memory>
+
+#if defined(WEBRTC_WIN)
+#include "modules/audio_device/win/audio_device_module_win.h"
+#include "modules/audio_device/win/core_audio_input_win.h"
+#include "modules/audio_device/win/core_audio_output_win.h"
+#include "modules/audio_device/win/core_audio_utility_win.h"
+#endif
+
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return CreateWindowsCoreAudioAudioDeviceModuleForTest(task_queue_factory,
+                                                        automatic_restart);
+}
+
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleForTest(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  // Returns NULL if Core Audio is not supported or if COM has not been
+  // initialized correctly using ScopedCOMInitializer.
+  if (!webrtc_win::core_audio_utility::IsSupported()) {
+    RTC_LOG(LS_ERROR)
+        << "Unable to create ADM since Core Audio is not supported";
+    return nullptr;
+  }
+  return CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
+      std::make_unique<webrtc_win::CoreAudioInput>(automatic_restart),
+      std::make_unique<webrtc_win::CoreAudioOutput>(automatic_restart),
+      task_queue_factory);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/audio_device_factory.h

@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+// Creates an AudioDeviceModule (ADM) for Windows based on the Core Audio API.
+// The creating thread must be a COM thread; otherwise nullptr will be returned.
+// By default `automatic_restart` is set to true and it results in support for
+// automatic restart of audio if e.g. the existing device is removed. If set to
+// false, no attempt to restart audio is performed under these conditions.
+//
+// Example (assuming webrtc namespace):
+//
+//  public:
+//   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice() {
+//     task_queue_factory_ = CreateDefaultTaskQueueFactory();
+//     // Tell COM that this thread shall live in the MTA.
+//     com_initializer_ = std::make_unique<ScopedCOMInitializer>(
+//         ScopedCOMInitializer::kMTA);
+//     if (!com_initializer_->Succeeded()) {
+//       return nullptr;
+//     }
+//     // Create the ADM with support for automatic restart if devices are
+//     // unplugged.
+//     return CreateWindowsCoreAudioAudioDeviceModule(
+//         task_queue_factory_.get());
+//   }
+//
+//   private:
+//    std::unique_ptr<ScopedCOMInitializer> com_initializer_;
+//    std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+//
+rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart = true);
+
+rtc::scoped_refptr<AudioDeviceModuleForTest>
+CreateWindowsCoreAudioAudioDeviceModuleForTest(
+    TaskQueueFactory* task_queue_factory,
+    bool automatic_restart = true);
+
+}  // namespace webrtc
+
+#endif  //  MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/fake_audio_device.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
+
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_default.h"
+
+namespace webrtc {
+
+class FakeAudioDeviceModule
+    : public webrtc_impl::AudioDeviceModuleDefault<AudioDeviceModule> {
+ public:
+  // TODO(bugs.webrtc.org/12701): Fix all users of this class to managed
+  // references using scoped_refptr. Current code doesn't always use refcounting
+  // for this class.
+  void AddRef() const override {}
+  webrtc::RefCountReleaseStatus Release() const override {
+    return webrtc::RefCountReleaseStatus::kDroppedLastRef;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/mock_audio_device.h

@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_
+
+#include <string>
+
+#include "api/make_ref_counted.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioDeviceModule : public AudioDeviceModule {
+ public:
+  static rtc::scoped_refptr<MockAudioDeviceModule> CreateNice() {
+    return rtc::make_ref_counted<::testing::NiceMock<MockAudioDeviceModule>>();
+  }
+  static rtc::scoped_refptr<MockAudioDeviceModule> CreateStrict() {
+    return rtc::make_ref_counted<
+        ::testing::StrictMock<MockAudioDeviceModule>>();
+  }
+
+  // AudioDeviceModule.
+  MOCK_METHOD(int32_t,
+              ActiveAudioLayer,
+              (AudioLayer * audioLayer),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              RegisterAudioCallback,
+              (AudioTransport * audioCallback),
+              (override));
+  MOCK_METHOD(int32_t, Init, (), (override));
+  MOCK_METHOD(int32_t, Terminate, (), (override));
+  MOCK_METHOD(bool, Initialized, (), (const, override));
+  MOCK_METHOD(int16_t, PlayoutDevices, (), (override));
+  MOCK_METHOD(int16_t, RecordingDevices, (), (override));
+  MOCK_METHOD(int32_t,
+              PlayoutDeviceName,
+              (uint16_t index,
+               char name[kAdmMaxDeviceNameSize],
+               char guid[kAdmMaxGuidSize]),
+              (override));
+  MOCK_METHOD(int32_t,
+              RecordingDeviceName,
+              (uint16_t index,
+               char name[kAdmMaxDeviceNameSize],
+               char guid[kAdmMaxGuidSize]),
+              (override));
+  MOCK_METHOD(int32_t, SetPlayoutDevice, (uint16_t index), (override));
+  MOCK_METHOD(int32_t,
+              SetPlayoutDevice,
+              (WindowsDeviceType device),
+              (override));
+  MOCK_METHOD(int32_t, SetRecordingDevice, (uint16_t index), (override));
+  MOCK_METHOD(int32_t,
+              SetRecordingDevice,
+              (WindowsDeviceType device),
+              (override));
+  MOCK_METHOD(int32_t, PlayoutIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, InitPlayout, (), (override));
+  MOCK_METHOD(bool, PlayoutIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, RecordingIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, InitRecording, (), (override));
+  MOCK_METHOD(bool, RecordingIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, StartPlayout, (), (override));
+  MOCK_METHOD(int32_t, StopPlayout, (), (override));
+  MOCK_METHOD(bool, Playing, (), (const, override));
+  MOCK_METHOD(int32_t, StartRecording, (), (override));
+  MOCK_METHOD(int32_t, StopRecording, (), (override));
+  MOCK_METHOD(bool, Recording, (), (const, override));
+  MOCK_METHOD(int32_t, InitSpeaker, (), (override));
+  MOCK_METHOD(bool, SpeakerIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, InitMicrophone, (), (override));
+  MOCK_METHOD(bool, MicrophoneIsInitialized, (), (const, override));
+  MOCK_METHOD(int32_t, SpeakerVolumeIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, SetSpeakerVolume, (uint32_t volume), (override));
+  MOCK_METHOD(int32_t, SpeakerVolume, (uint32_t * volume), (const, override));
+  MOCK_METHOD(int32_t,
+              MaxSpeakerVolume,
+              (uint32_t * maxVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MinSpeakerVolume,
+              (uint32_t * minVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MicrophoneVolumeIsAvailable,
+              (bool* available),
+              (override));
+  MOCK_METHOD(int32_t, SetMicrophoneVolume, (uint32_t volume), (override));
+  MOCK_METHOD(int32_t,
+              MicrophoneVolume,
+              (uint32_t * volume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MaxMicrophoneVolume,
+              (uint32_t * maxVolume),
+              (const, override));
+  MOCK_METHOD(int32_t,
+              MinMicrophoneVolume,
+              (uint32_t * minVolume),
+              (const, override));
+  MOCK_METHOD(int32_t, SpeakerMuteIsAvailable, (bool* available), (override));
+  MOCK_METHOD(int32_t, SetSpeakerMute, (bool enable), (override));
+  MOCK_METHOD(int32_t, SpeakerMute, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              MicrophoneMuteIsAvailable,
+              (bool* available),
+              (override));
+  MOCK_METHOD(int32_t, SetMicrophoneMute, (bool enable), (override));
+  MOCK_METHOD(int32_t, MicrophoneMute, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              StereoPlayoutIsAvailable,
+              (bool* available),
+              (const, override));
+  MOCK_METHOD(int32_t, SetStereoPlayout, (bool enable), (override));
+  MOCK_METHOD(int32_t, StereoPlayout, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t,
+              StereoRecordingIsAvailable,
+              (bool* available),
+              (const, override));
+  MOCK_METHOD(int32_t, SetStereoRecording, (bool enable), (override));
+  MOCK_METHOD(int32_t, StereoRecording, (bool* enabled), (const, override));
+  MOCK_METHOD(int32_t, PlayoutDelay, (uint16_t * delayMS), (const, override));
+  MOCK_METHOD(bool, BuiltInAECIsAvailable, (), (const, override));
+  MOCK_METHOD(bool, BuiltInAGCIsAvailable, (), (const, override));
+  MOCK_METHOD(bool, BuiltInNSIsAvailable, (), (const, override));
+  MOCK_METHOD(int32_t, EnableBuiltInAEC, (bool enable), (override));
+  MOCK_METHOD(int32_t, EnableBuiltInAGC, (bool enable), (override));
+  MOCK_METHOD(int32_t, EnableBuiltInNS, (bool enable), (override));
+  MOCK_METHOD(int32_t, GetPlayoutUnderrunCount, (), (const, override));
+#if defined(WEBRTC_IOS)
+  MOCK_METHOD(int,
+              GetPlayoutAudioParameters,
+              (AudioParameters * params),
+              (const, override));
+  MOCK_METHOD(int,
+              GetRecordAudioParameters,
+              (AudioParameters * params),
+              (const, override));
+#endif  // WEBRTC_IOS
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_DEVICE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/mock_audio_transport.h

@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioTransport : public AudioTransport {
+ public:
+  MockAudioTransport() {}
+  ~MockAudioTransport() {}
+
+  MOCK_METHOD(int32_t,
+              RecordedDataIsAvailable,
+              (const void* audioSamples,
+               size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               uint32_t totalDelayMS,
+               int32_t clockDrift,
+               uint32_t currentMicLevel,
+               bool keyPressed,
+               uint32_t& newMicLevel),
+              (override));
+
+  MOCK_METHOD(int32_t,
+              RecordedDataIsAvailable,
+              (const void* audioSamples,
+               size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               uint32_t totalDelayMS,
+               int32_t clockDrift,
+               uint32_t currentMicLevel,
+               bool keyPressed,
+               uint32_t& newMicLevel,
+               absl::optional<int64_t> estimated_capture_time_ns),
+              (override));
+
+  MOCK_METHOD(int32_t,
+              NeedMorePlayData,
+              (size_t nSamples,
+               size_t nBytesPerSample,
+               size_t nChannels,
+               uint32_t samplesPerSec,
+               void* audioSamples,
+               size_t& nSamplesOut,
+               int64_t* elapsed_time_ms,
+               int64_t* ntp_time_ms),
+              (override));
+
+  MOCK_METHOD(void,
+              PullRenderData,
+              (int bits_per_sample,
+               int sample_rate,
+               size_t number_of_channels,
+               size_t number_of_frames,
+               void* audio_data,
+               int64_t* elapsed_time_ms,
+               int64_t* ntp_time_ms),
+              (override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/test_audio_device.cc

@@ -0,0 +1,540 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_device/include/test_audio_device.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstdlib>
+#include <memory>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/make_ref_counted.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_device/audio_device_impl.h"
+#include "modules/audio_device/include/audio_device_default.h"
+#include "modules/audio_device/test_audio_device_impl.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFrameLengthUs = 10000;
+constexpr int kFramesPerSecond = rtc::kNumMicrosecsPerSec / kFrameLengthUs;
+
+class TestAudioDeviceModuleImpl : public AudioDeviceModuleImpl {
+ public:
+  TestAudioDeviceModuleImpl(
+      TaskQueueFactory* task_queue_factory,
+      std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+      std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+      float speed = 1)
+      : AudioDeviceModuleImpl(
+            AudioLayer::kDummyAudio,
+            std::make_unique<TestAudioDevice>(task_queue_factory,
+                                              std::move(capturer),
+                                              std::move(renderer),
+                                              speed),
+            task_queue_factory,
+            /*create_detached=*/true) {}
+
+  ~TestAudioDeviceModuleImpl() override = default;
+};
+
+// A fake capturer that generates pulses with random samples between
+// -max_amplitude and +max_amplitude.
+class PulsedNoiseCapturerImpl final
+    : public TestAudioDeviceModule::PulsedNoiseCapturer {
+ public:
+  // Assuming 10ms audio packets.
+  PulsedNoiseCapturerImpl(int16_t max_amplitude,
+                          int sampling_frequency_in_hz,
+                          int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        fill_with_zero_(false),
+        random_generator_(1),
+        max_amplitude_(max_amplitude),
+        num_channels_(num_channels) {
+    RTC_DCHECK_GT(max_amplitude, 0);
+  }
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
+    fill_with_zero_ = !fill_with_zero_;
+    int16_t max_amplitude;
+    {
+      MutexLock lock(&lock_);
+      max_amplitude = max_amplitude_;
+    }
+    buffer->SetData(
+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
+            num_channels_,
+        [&](rtc::ArrayView<int16_t> data) {
+          if (fill_with_zero_) {
+            std::fill(data.begin(), data.end(), 0);
+          } else {
+            std::generate(data.begin(), data.end(), [&]() {
+              return random_generator_.Rand(-max_amplitude, max_amplitude);
+            });
+          }
+          return data.size();
+        });
+    return true;
+  }
+
+  void SetMaxAmplitude(int16_t amplitude) override {
+    MutexLock lock(&lock_);
+    max_amplitude_ = amplitude;
+  }
+
+ private:
+  int sampling_frequency_in_hz_;
+  bool fill_with_zero_;
+  Random random_generator_;
+  Mutex lock_;
+  int16_t max_amplitude_ RTC_GUARDED_BY(lock_);
+  const int num_channels_;
+};
+
+class WavFileReader final : public TestAudioDeviceModule::Capturer {
+ public:
+  WavFileReader(absl::string_view filename,
+                int sampling_frequency_in_hz,
+                int num_channels,
+                bool repeat)
+      : WavFileReader(std::make_unique<WavReader>(filename),
+                      sampling_frequency_in_hz,
+                      num_channels,
+                      repeat) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
+    buffer->SetData(
+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
+            num_channels_,
+        [&](rtc::ArrayView<int16_t> data) {
+          size_t read = wav_reader_->ReadSamples(data.size(), data.data());
+          if (read < data.size() && repeat_) {
+            do {
+              wav_reader_->Reset();
+              size_t delta = wav_reader_->ReadSamples(
+                  data.size() - read, data.subview(read).data());
+              RTC_CHECK_GT(delta, 0) << "No new data read from file";
+              read += delta;
+            } while (read < data.size());
+          }
+          return read;
+        });
+    return buffer->size() > 0;
+  }
+
+ private:
+  WavFileReader(std::unique_ptr<WavReader> wav_reader,
+                int sampling_frequency_in_hz,
+                int num_channels,
+                bool repeat)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels),
+        wav_reader_(std::move(wav_reader)),
+        repeat_(repeat) {
+    RTC_CHECK_EQ(wav_reader_->sample_rate(), sampling_frequency_in_hz);
+    RTC_CHECK_EQ(wav_reader_->num_channels(), num_channels);
+  }
+
+  const int sampling_frequency_in_hz_;
+  const int num_channels_;
+  std::unique_ptr<WavReader> wav_reader_;
+  const bool repeat_;
+};
+
+class WavFileWriter final : public TestAudioDeviceModule::Renderer {
+ public:
+  WavFileWriter(absl::string_view filename,
+                int sampling_frequency_in_hz,
+                int num_channels)
+      : WavFileWriter(std::make_unique<WavWriter>(filename,
+                                                  sampling_frequency_in_hz,
+                                                  num_channels),
+                      sampling_frequency_in_hz,
+                      num_channels) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    wav_writer_->WriteSamples(data.data(), data.size());
+    return true;
+  }
+
+ private:
+  WavFileWriter(std::unique_ptr<WavWriter> wav_writer,
+                int sampling_frequency_in_hz,
+                int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        wav_writer_(std::move(wav_writer)),
+        num_channels_(num_channels) {}
+
+  int sampling_frequency_in_hz_;
+  std::unique_ptr<WavWriter> wav_writer_;
+  const int num_channels_;
+};
+
+class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
+ public:
+  BoundedWavFileWriter(absl::string_view filename,
+                       int sampling_frequency_in_hz,
+                       int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        wav_writer_(filename, sampling_frequency_in_hz, num_channels),
+        num_channels_(num_channels),
+        silent_audio_(
+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
+                num_channels,
+            0),
+        started_writing_(false),
+        trailing_zeros_(0) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    const int16_t kAmplitudeThreshold = 5;
+
+    const int16_t* begin = data.begin();
+    const int16_t* end = data.end();
+    if (!started_writing_) {
+      // Cut off silence at the beginning.
+      while (begin < end) {
+        if (std::abs(*begin) > kAmplitudeThreshold) {
+          started_writing_ = true;
+          break;
+        }
+        ++begin;
+      }
+    }
+    if (started_writing_) {
+      // Cut off silence at the end.
+      while (begin < end) {
+        if (*(end - 1) != 0) {
+          break;
+        }
+        --end;
+      }
+      if (begin < end) {
+        // If it turns out that the silence was not final, need to write all the
+        // skipped zeros and continue writing audio.
+        while (trailing_zeros_ > 0) {
+          const size_t zeros_to_write =
+              std::min(trailing_zeros_, silent_audio_.size());
+          wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write);
+          trailing_zeros_ -= zeros_to_write;
+        }
+        wav_writer_.WriteSamples(begin, end - begin);
+      }
+      // Save the number of zeros we skipped in case this needs to be restored.
+      trailing_zeros_ += data.end() - end;
+    }
+    return true;
+  }
+
+ private:
+  int sampling_frequency_in_hz_;
+  WavWriter wav_writer_;
+  const int num_channels_;
+  std::vector<int16_t> silent_audio_;
+  bool started_writing_;
+  size_t trailing_zeros_;
+};
+
+class DiscardRenderer final : public TestAudioDeviceModule::Renderer {
+ public:
+  explicit DiscardRenderer(int sampling_frequency_in_hz, int num_channels)
+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels) {}
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override { return true; }
+
+ private:
+  int sampling_frequency_in_hz_;
+  const int num_channels_;
+};
+
+class RawFileReader final : public TestAudioDeviceModule::Capturer {
+ public:
+  RawFileReader(absl::string_view input_file_name,
+                int sampling_frequency_in_hz,
+                int num_channels,
+                bool repeat)
+      : input_file_name_(input_file_name),
+        sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels),
+        repeat_(repeat),
+        read_buffer_(
+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
+                num_channels * 2,
+            0) {
+    input_file_ = FileWrapper::OpenReadOnly(input_file_name_);
+    RTC_CHECK(input_file_.is_open())
+        << "Failed to open audio input file: " << input_file_name_;
+  }
+
+  ~RawFileReader() override { input_file_.Close(); }
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
+    buffer->SetData(
+        TestAudioDeviceModule::SamplesPerFrame(SamplingFrequency()) *
+            NumChannels(),
+        [&](rtc::ArrayView<int16_t> data) {
+          rtc::ArrayView<int8_t> read_buffer_view = ReadBufferView();
+          size_t size = data.size() * 2;
+          size_t read = input_file_.Read(read_buffer_view.data(), size);
+          if (read < size && repeat_) {
+            do {
+              input_file_.Rewind();
+              size_t delta = input_file_.Read(
+                  read_buffer_view.subview(read).data(), size - read);
+              RTC_CHECK_GT(delta, 0) << "No new data to read from file";
+              read += delta;
+            } while (read < size);
+          }
+          memcpy(data.data(), read_buffer_view.data(), size);
+          return read / 2;
+        });
+    return buffer->size() > 0;
+  }
+
+ private:
+  rtc::ArrayView<int8_t> ReadBufferView() { return read_buffer_; }
+
+  const std::string input_file_name_;
+  const int sampling_frequency_in_hz_;
+  const int num_channels_;
+  const bool repeat_;
+  FileWrapper input_file_;
+  std::vector<int8_t> read_buffer_;
+};
+
+class RawFileWriter : public TestAudioDeviceModule::Renderer {
+ public:
+  RawFileWriter(absl::string_view output_file_name,
+                int sampling_frequency_in_hz,
+                int num_channels)
+      : output_file_name_(output_file_name),
+        sampling_frequency_in_hz_(sampling_frequency_in_hz),
+        num_channels_(num_channels),
+        silent_audio_(
+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
+                num_channels * 2,
+            0),
+        write_buffer_(
+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
+                num_channels * 2,
+            0),
+        started_writing_(false),
+        trailing_zeros_(0) {
+    output_file_ = FileWrapper::OpenWriteOnly(output_file_name_);
+    RTC_CHECK(output_file_.is_open())
+        << "Failed to open playout file" << output_file_name_;
+  }
+  ~RawFileWriter() override { output_file_.Close(); }
+
+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
+
+  int NumChannels() const override { return num_channels_; }
+
+  bool Render(rtc::ArrayView<const int16_t> data) override {
+    const int16_t kAmplitudeThreshold = 5;
+
+    const int16_t* begin = data.begin();
+    const int16_t* end = data.end();
+    if (!started_writing_) {
+      // Cut off silence at the beginning.
+      while (begin < end) {
+        if (std::abs(*begin) > kAmplitudeThreshold) {
+          started_writing_ = true;
+          break;
+        }
+        ++begin;
+      }
+    }
+    if (started_writing_) {
+      // Cut off silence at the end.
+      while (begin < end) {
+        if (*(end - 1) != 0) {
+          break;
+        }
+        --end;
+      }
+      if (begin < end) {
+        // If it turns out that the silence was not final, need to write all the
+        // skipped zeros and continue writing audio.
+        while (trailing_zeros_ > 0) {
+          const size_t zeros_to_write =
+              std::min(trailing_zeros_, silent_audio_.size());
+          output_file_.Write(silent_audio_.data(), zeros_to_write * 2);
+          trailing_zeros_ -= zeros_to_write;
+        }
+        WriteInt16(begin, end);
+      }
+      // Save the number of zeros we skipped in case this needs to be restored.
+      trailing_zeros_ += data.end() - end;
+    }
+    return true;
+  }
+
+ private:
+  void WriteInt16(const int16_t* begin, const int16_t* end) {
+    int size = (end - begin) * sizeof(int16_t);
+    memcpy(write_buffer_.data(), begin, size);
+    output_file_.Write(write_buffer_.data(), size);
+  }
+
+  const std::string output_file_name_;
+  const int sampling_frequency_in_hz_;
+  const int num_channels_;
+  FileWrapper output_file_;
+  std::vector<int8_t> silent_audio_;
+  std::vector<int8_t> write_buffer_;
+  bool started_writing_;
+  size_t trailing_zeros_;
+};
+
+}  // namespace
+
+size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) {
+  return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
+}
+
+rtc::scoped_refptr<AudioDeviceModule> TestAudioDeviceModule::Create(
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+    float speed) {
+  auto audio_device = rtc::make_ref_counted<TestAudioDeviceModuleImpl>(
+      task_queue_factory, std::move(capturer), std::move(renderer), speed);
+
+  // Ensure that the current platform is supported.
+  if (audio_device->CheckPlatform() == -1) {
+    return nullptr;
+  }
+
+  // Create the platform-dependent implementation.
+  if (audio_device->CreatePlatformSpecificObjects() == -1) {
+    return nullptr;
+  }
+
+  // Ensure that the generic audio buffer can communicate with the platform
+  // specific parts.
+  if (audio_device->AttachAudioBuffer() == -1) {
+    return nullptr;
+  }
+
+  return audio_device;
+}
+
+std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer>
+TestAudioDeviceModule::CreatePulsedNoiseCapturer(int16_t max_amplitude,
+                                                 int sampling_frequency_in_hz,
+                                                 int num_channels) {
+  return std::make_unique<PulsedNoiseCapturerImpl>(
+      max_amplitude, sampling_frequency_in_hz, num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateDiscardRenderer(int sampling_frequency_in_hz,
+                                             int num_channels) {
+  return std::make_unique<DiscardRenderer>(sampling_frequency_in_hz,
+                                           num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels) {
+  return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz,
+                                         num_channels, false);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename,
+                                           bool repeat) {
+  WavReader reader(filename);
+  int sampling_frequency_in_hz = reader.sample_rate();
+  int num_channels = rtc::checked_cast<int>(reader.num_channels());
+  return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz,
+                                         num_channels, repeat);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateWavFileWriter(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels) {
+  return std::make_unique<WavFileWriter>(filename, sampling_frequency_in_hz,
+                                         num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateBoundedWavFileWriter(absl::string_view filename,
+                                                  int sampling_frequency_in_hz,
+                                                  int num_channels) {
+  return std::make_unique<BoundedWavFileWriter>(
+      filename, sampling_frequency_in_hz, num_channels);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+TestAudioDeviceModule::CreateRawFileReader(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels,
+                                           bool repeat) {
+  return std::make_unique<RawFileReader>(filename, sampling_frequency_in_hz,
+                                         num_channels, repeat);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+TestAudioDeviceModule::CreateRawFileWriter(absl::string_view filename,
+                                           int sampling_frequency_in_hz,
+                                           int num_channels) {
+  return std::make_unique<RawFileWriter>(filename, sampling_frequency_in_hz,
+                                         num_channels);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/test_audio_device.h

@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+#define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// This is test API and is in development, so it can be changed/removed without
+// notice.
+
+// This class exists for historical reasons. For now it only contains static
+// methods to create test AudioDeviceModule. Implementation details of that
+// module are considered private. This class isn't intended to be instantiated.
+class TestAudioDeviceModule {
+ public:
+  // Returns the number of samples that Capturers and Renderers with this
+  // sampling frequency will work with every time Capture or Render is called.
+  static size_t SamplesPerFrame(int sampling_frequency_in_hz);
+
+  class Capturer {
+   public:
+    virtual ~Capturer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // capturer produces.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of captured audio data.
+    virtual int NumChannels() const = 0;
+    // Replaces the contents of `buffer` with 10ms of captured audio data
+    // (see TestAudioDeviceModule::SamplesPerFrame). Returns true if the
+    // capturer can keep producing data, or false when the capture finishes.
+    virtual bool Capture(rtc::BufferT<int16_t>* buffer) = 0;
+  };
+
+  class Renderer {
+   public:
+    virtual ~Renderer() {}
+    // Returns the sampling frequency in Hz of the audio data that this
+    // renderer receives.
+    virtual int SamplingFrequency() const = 0;
+    // Returns the number of channels of audio data to be required.
+    virtual int NumChannels() const = 0;
+    // Renders the passed audio data and returns true if the renderer wants
+    // to keep receiving data, or false otherwise.
+    virtual bool Render(rtc::ArrayView<const int16_t> data) = 0;
+  };
+
+  // A fake capturer that generates pulses with random samples between
+  // -max_amplitude and +max_amplitude.
+  class PulsedNoiseCapturer : public Capturer {
+   public:
+    ~PulsedNoiseCapturer() override {}
+
+    virtual void SetMaxAmplitude(int16_t amplitude) = 0;
+  };
+
+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
+  // frames will be processed every 10ms / `speed`.
+  // `capturer` is an object that produces audio data. Can be nullptr if this
+  // device is never used for recording.
+  // `renderer` is an object that receives audio data that would have been
+  // played out. Can be nullptr if this device is never used for playing.
+  // Use one of the Create... functions to get these instances.
+  static rtc::scoped_refptr<AudioDeviceModule> Create(
+      TaskQueueFactory* task_queue_factory,
+      std::unique_ptr<Capturer> capturer,
+      std::unique_ptr<Renderer> renderer,
+      float speed = 1);
+
+  // Returns a Capturer instance that generates a signal of `num_channels`
+  // channels where every second frame is zero and every second frame is evenly
+  // distributed random noise with max amplitude `max_amplitude`.
+  static std::unique_ptr<PulsedNoiseCapturer> CreatePulsedNoiseCapturer(
+      int16_t max_amplitude,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that does nothing with the audio data.
+  static std::unique_ptr<Renderer> CreateDiscardRenderer(
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // WavReader and WavWriter creation based on file name.
+
+  // Returns a Capturer instance that gets its data from a WAV file. The sample
+  // rate and channels will be checked against the Wav file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Capturer instance that gets its data from a file.
+  // Automatically detects sample rate and num of channels.
+  // `repeat` - if true, the file will be replayed from the start when we reach
+  // the end of file.
+  static std::unique_ptr<Capturer> CreateWavFileReader(
+      absl::string_view filename,
+      bool repeat = false);
+
+  // Returns a Renderer instance that writes its data to a file.
+  static std::unique_ptr<Renderer> CreateWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Renderer instance that writes its data to a WAV file, cutting
+  // off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz,
+      int num_channels = 1);
+
+  // Returns a Capturer instance that gets its data from a raw file (*.raw).
+  static std::unique_ptr<Capturer> CreateRawFileReader(
+      absl::string_view filename,
+      int sampling_frequency_in_hz = 48000,
+      int num_channels = 2,
+      bool repeat = true);
+
+  // Returns a Renderer instance that writes its data to a raw file (*.raw),
+  // cutting off silence at the beginning (not necessarily perfect silence, see
+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
+  static std::unique_ptr<Renderer> CreateRawFileWriter(
+      absl::string_view filename,
+      int sampling_frequency_in_hz = 48000,
+      int num_channels = 2);
+
+ private:
+  TestAudioDeviceModule() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/include/test_audio_device_unittest.cc

@@ -0,0 +1,528 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/include/test_audio_device.h"
+
+#include <algorithm>
+#include <array>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "common_audio/wav_file.h"
+#include "common_audio/wav_header.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+void RunWavTest(const std::vector<int16_t>& input_samples,
+                const std::vector<int16_t>& expected_samples) {
+  const ::testing::TestInfo* const test_info =
+      ::testing::UnitTest::GetInstance()->current_test_info();
+
+  const std::string output_filename = test::OutputPathWithRandomDirectory() +
+                                      "BoundedWavFileWriterTest_" +
+                                      test_info->name() + ".wav";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Test through file name API.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
+
+    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &input_samples[i],
+          std::min(kSamplesPerFrame, input_samples.size() - i))));
+    }
+  }
+
+  {
+    WavReader reader(output_filename);
+    std::vector<int16_t> read_samples(expected_samples.size());
+    EXPECT_EQ(expected_samples.size(),
+              reader.ReadSamples(read_samples.size(), read_samples.data()));
+    EXPECT_EQ(expected_samples, read_samples);
+
+    EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
+  }
+
+  remove(output_filename.c_str());
+}
+
+TEST(BoundedWavFileWriterTest, NoSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75,   1234,  243,    -1231, -22222, 0,    3,      88,
+      1222, -1213, -13222, -7,    -3525,  5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples = kInputSamples;
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, SomeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
+                                                     kInputSamples.end());
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
+                                                     kInputSamples.end());
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, SomeEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 9);
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234,  243,    -1231, -22222, 0,    0, 0,
+      0,  -1213, -13222, -7,    -3525,  5787, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 2);
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, DoubleSilence) {
+  static const std::vector<int16_t> kInputSamples = {0,     -1213, -13222, -7,
+                                                     -3525, 5787,  0,      0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
+                                                     kInputSamples.end() - 2);
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 4);
+  RunWavTest(kInputSamples, kExpectedSamples);
+}
+
+TEST(WavFileReaderTest, RepeatedTrueWithSingleFrameFileReadTwice) {
+  static const std::vector<int16_t> kInputSamples = {75,     1234, 243, -1231,
+                                                     -22222, 0,    3,   88};
+  static const rtc::BufferT<int16_t> kExpectedSamples(kInputSamples.data(),
+                                                      kInputSamples.size());
+
+  const std::string output_filename = test::OutputPathWithRandomDirectory() +
+                                      "WavFileReaderTest_RepeatedTrue_" +
+                                      ".wav";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Create raw file to read.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateWavFileWriter(output_filename, 800);
+
+    for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &kInputSamples[i],
+          std::min(kSamplesPerFrame, kInputSamples.size() - i))));
+    }
+  }
+
+  {
+    std::unique_ptr<TestAudioDeviceModule::Capturer> reader =
+        TestAudioDeviceModule::CreateWavFileReader(output_filename, true);
+    rtc::BufferT<int16_t> buffer(kExpectedSamples.size());
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+  }
+
+  remove(output_filename.c_str());
+}
+
+void RunRawTestNoRepeat(const std::vector<int16_t>& input_samples,
+                        const std::vector<int16_t>& expected_samples) {
+  const ::testing::TestInfo* const test_info =
+      ::testing::UnitTest::GetInstance()->current_test_info();
+
+  const std::string output_filename = test::OutputPathWithRandomDirectory() +
+                                      "RawFileTest_" + test_info->name() +
+                                      ".raw";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Test through file name API.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateRawFileWriter(
+            output_filename, /*sampling_frequency_in_hz=*/800);
+
+    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &input_samples[i],
+          std::min(kSamplesPerFrame, input_samples.size() - i))));
+    }
+  }
+
+  {
+    std::unique_ptr<TestAudioDeviceModule::Capturer> reader =
+        TestAudioDeviceModule::CreateRawFileReader(
+            output_filename, /*sampling_frequency_in_hz=*/800,
+            /*num_channels=*/2, /*repeat=*/false);
+    rtc::BufferT<int16_t> buffer(expected_samples.size());
+    rtc::BufferT<int16_t> expected_buffer(expected_samples.size());
+    expected_buffer.SetData(expected_samples);
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(expected_buffer, buffer);
+    EXPECT_FALSE(reader->Capture(&buffer));
+    EXPECT_TRUE(buffer.empty());
+  }
+
+  remove(output_filename.c_str());
+}
+
+TEST(RawFileWriterTest, NoSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75,   1234,  243,    -1231, -22222, 0,    3,      88,
+      1222, -1213, -13222, -7,    -3525,  5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples = kInputSamples;
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, SomeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
+                                                     kInputSamples.end());
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, NegativeStartSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
+                                                     kInputSamples.end());
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, SomeEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 9);
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, DoubleEndSilence) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234,  243,    -1231, -22222, 0,    0, 0,
+      0,  -1213, -13222, -7,    -3525,  5787, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 2);
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, DoubleSilence) {
+  static const std::vector<int16_t> kInputSamples = {0,     -1213, -13222, -7,
+                                                     -3525, 5787,  0,      0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
+                                                     kInputSamples.end() - 2);
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, EndSilenceCutoff) {
+  static const std::vector<int16_t> kInputSamples = {
+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
+                                                     kInputSamples.end() - 4);
+  RunRawTestNoRepeat(kInputSamples, kExpectedSamples);
+}
+
+TEST(RawFileWriterTest, Repeat) {
+  static const std::vector<int16_t> kInputSamples = {
+      75,   1234,  243,    -1231, -22222, 0,    3,      88,
+      1222, -1213, -13222, -7,    -3525,  5787, -25247, 8};
+  static const rtc::BufferT<int16_t> kExpectedSamples(kInputSamples.data(),
+                                                      kInputSamples.size());
+
+  const ::testing::TestInfo* const test_info =
+      ::testing::UnitTest::GetInstance()->current_test_info();
+
+  const std::string output_filename = test::OutputPathWithRandomDirectory() +
+                                      "RawFileTest_" + test_info->name() + "_" +
+                                      std::to_string(std::rand()) + ".raw";
+
+  static const size_t kSamplesPerFrame = 8;
+  static const int kSampleRate = kSamplesPerFrame * 100;
+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
+            kSamplesPerFrame);
+
+  // Test through file name API.
+  {
+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
+        TestAudioDeviceModule::CreateRawFileWriter(
+            output_filename, /*sampling_frequency_in_hz=*/800);
+
+    for (size_t i = 0; i < kInputSamples.size(); i += kSamplesPerFrame) {
+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
+          &kInputSamples[i],
+          std::min(kSamplesPerFrame, kInputSamples.size() - i))));
+    }
+  }
+
+  {
+    std::unique_ptr<TestAudioDeviceModule::Capturer> reader =
+        TestAudioDeviceModule::CreateRawFileReader(
+            output_filename, /*sampling_frequency_in_hz=*/800,
+            /*num_channels=*/2, /*repeat=*/true);
+    rtc::BufferT<int16_t> buffer(kExpectedSamples.size());
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+    EXPECT_TRUE(reader->Capture(&buffer));
+    EXPECT_EQ(kExpectedSamples, buffer);
+  }
+
+  remove(output_filename.c_str());
+}
+
+TEST(PulsedNoiseCapturerTest, SetMaxAmplitude) {
+  const int16_t kAmplitude = 50;
+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          kAmplitude, /*sampling_frequency_in_hz=*/8000);
+  rtc::BufferT<int16_t> recording_buffer;
+
+  // Verify that the capturer doesn't create entries louder than than
+  // kAmplitude. Since the pulse generator alternates between writing
+  // zeroes and actual entries, we need to do the capturing twice.
+  capturer->Capture(&recording_buffer);
+  capturer->Capture(&recording_buffer);
+  int16_t max_sample =
+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
+  EXPECT_LE(max_sample, kAmplitude);
+
+  // Increase the amplitude and verify that the samples can now be louder
+  // than the previous max.
+  capturer->SetMaxAmplitude(kAmplitude * 2);
+  capturer->Capture(&recording_buffer);
+  capturer->Capture(&recording_buffer);
+  max_sample =
+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
+  EXPECT_GT(max_sample, kAmplitude);
+}
+
+using ::testing::ElementsAre;
+
+constexpr Timestamp kStartTime = Timestamp::Millis(10000);
+
+class TestAudioTransport : public AudioTransport {
+ public:
+  enum class Mode { kPlaying, kRecording };
+
+  explicit TestAudioTransport(Mode mode) : mode_(mode) {}
+  ~TestAudioTransport() override = default;
+
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t samples_per_channel,
+      size_t bytes_per_sample,
+      size_t number_of_channels,
+      uint32_t samples_per_second,
+      uint32_t total_delay_ms,
+      int32_t clock_drift,
+      uint32_t current_mic_level,
+      bool key_pressed,
+      uint32_t& new_mic_level,
+      absl::optional<int64_t> estimated_capture_time_ns) override {
+    new_mic_level = 1;
+
+    if (mode_ != Mode::kRecording) {
+      EXPECT_TRUE(false)
+          << "NeedMorePlayData mustn't be called when mode isn't kRecording";
+      return -1;
+    }
+
+    MutexLock lock(&mutex_);
+    samples_per_channel_.push_back(samples_per_channel);
+    number_of_channels_.push_back(number_of_channels);
+    bytes_per_sample_.push_back(bytes_per_sample);
+    samples_per_second_.push_back(samples_per_second);
+    return 0;
+  }
+
+  int32_t NeedMorePlayData(size_t samples_per_channel,
+                           size_t bytes_per_sample,
+                           size_t number_of_channels,
+                           uint32_t samples_per_second,
+                           void* audio_samples,
+                           size_t& samples_out,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
+    const size_t num_bytes = samples_per_channel * number_of_channels;
+    std::memset(audio_samples, 1, num_bytes);
+    samples_out = samples_per_channel * number_of_channels;
+    *elapsed_time_ms = 0;
+    *ntp_time_ms = 0;
+
+    if (mode_ != Mode::kPlaying) {
+      EXPECT_TRUE(false)
+          << "NeedMorePlayData mustn't be called when mode isn't kPlaying";
+      return -1;
+    }
+
+    MutexLock lock(&mutex_);
+    samples_per_channel_.push_back(samples_per_channel);
+    number_of_channels_.push_back(number_of_channels);
+    bytes_per_sample_.push_back(bytes_per_sample);
+    samples_per_second_.push_back(samples_per_second);
+    return 0;
+  }
+
+  int32_t RecordedDataIsAvailable(const void* audio_samples,
+                                  size_t samples_per_channel,
+                                  size_t bytes_per_sample,
+                                  size_t number_of_channels,
+                                  uint32_t samples_per_second,
+                                  uint32_t total_delay_ms,
+                                  int32_t clockDrift,
+                                  uint32_t current_mic_level,
+                                  bool key_pressed,
+                                  uint32_t& new_mic_level) override {
+    RTC_CHECK(false) << "This methods should be never executed";
+  }
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {
+    RTC_CHECK(false) << "This methods should be never executed";
+  }
+
+  std::vector<size_t> samples_per_channel() const {
+    MutexLock lock(&mutex_);
+    return samples_per_channel_;
+  }
+  std::vector<size_t> number_of_channels() const {
+    MutexLock lock(&mutex_);
+    return number_of_channels_;
+  }
+  std::vector<size_t> bytes_per_sample() const {
+    MutexLock lock(&mutex_);
+    return bytes_per_sample_;
+  }
+  std::vector<size_t> samples_per_second() const {
+    MutexLock lock(&mutex_);
+    return samples_per_second_;
+  }
+
+ private:
+  const Mode mode_;
+
+  mutable Mutex mutex_;
+  std::vector<size_t> samples_per_channel_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> number_of_channels_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> bytes_per_sample_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> samples_per_second_ RTC_GUARDED_BY(mutex_);
+};
+
+TEST(TestAudioDeviceModuleTest, CreatedADMCanRecord) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording);
+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          /*max_amplitude=*/1000,
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  rtc::scoped_refptr<AudioDeviceModule> adm = TestAudioDeviceModule::Create(
+      time_controller.GetTaskQueueFactory(), std::move(capturer),
+      /*renderer=*/nullptr);
+
+  ASSERT_EQ(adm->RegisterAudioCallback(&audio_transport), 0);
+  ASSERT_EQ(adm->Init(), 0);
+
+  EXPECT_FALSE(adm->RecordingIsInitialized());
+  ASSERT_EQ(adm->InitRecording(), 0);
+  EXPECT_TRUE(adm->RecordingIsInitialized());
+  ASSERT_EQ(adm->StartRecording(), 0);
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_TRUE(adm->Recording());
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_EQ(adm->StopRecording(), 0);
+
+  EXPECT_THAT(audio_transport.samples_per_channel(),
+              ElementsAre(480, 480, 480));
+  EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
+  EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
+  EXPECT_THAT(audio_transport.samples_per_second(),
+              ElementsAre(48000, 48000, 48000));
+}
+
+TEST(TestAudioDeviceModuleTest, CreatedADMCanPlay) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying);
+  std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
+      TestAudioDeviceModule::CreateDiscardRenderer(
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  rtc::scoped_refptr<AudioDeviceModule> adm =
+      TestAudioDeviceModule::Create(time_controller.GetTaskQueueFactory(),
+                                    /*capturer=*/nullptr, std::move(renderer));
+
+  ASSERT_EQ(adm->RegisterAudioCallback(&audio_transport), 0);
+  ASSERT_EQ(adm->Init(), 0);
+
+  EXPECT_FALSE(adm->PlayoutIsInitialized());
+  ASSERT_EQ(adm->InitPlayout(), 0);
+  EXPECT_TRUE(adm->PlayoutIsInitialized());
+  ASSERT_EQ(adm->StartPlayout(), 0);
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_TRUE(adm->Playing());
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_EQ(adm->StopPlayout(), 0);
+
+  EXPECT_THAT(audio_transport.samples_per_channel(),
+              ElementsAre(480, 480, 480));
+  EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
+  EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
+  EXPECT_THAT(audio_transport.samples_per_second(),
+              ElementsAre(48000, 48000, 48000));
+}
+
+}  // namespace
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/alsasymboltable_linux.cc

@@ -0,0 +1,40 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/alsasymboltable_linux.h

@@ -0,0 +1,148 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_alsa {
+
+// The ALSA symbols we need, as an X-Macro list.
+// This list must contain precisely every libasound function that is used in
+// alsasoundsystem.cc.
+#define ALSA_SYMBOLS_LIST                      \
+  X(snd_device_name_free_hint)                 \
+  X(snd_device_name_get_hint)                  \
+  X(snd_device_name_hint)                      \
+  X(snd_pcm_avail_update)                      \
+  X(snd_pcm_close)                             \
+  X(snd_pcm_delay)                             \
+  X(snd_pcm_drop)                              \
+  X(snd_pcm_open)                              \
+  X(snd_pcm_prepare)                           \
+  X(snd_pcm_readi)                             \
+  X(snd_pcm_recover)                           \
+  X(snd_pcm_resume)                            \
+  X(snd_pcm_reset)                             \
+  X(snd_pcm_state)                             \
+  X(snd_pcm_set_params)                        \
+  X(snd_pcm_get_params)                        \
+  X(snd_pcm_start)                             \
+  X(snd_pcm_stream)                            \
+  X(snd_pcm_frames_to_bytes)                   \
+  X(snd_pcm_bytes_to_frames)                   \
+  X(snd_pcm_wait)                              \
+  X(snd_pcm_writei)                            \
+  X(snd_pcm_info_get_class)                    \
+  X(snd_pcm_info_get_subdevices_avail)         \
+  X(snd_pcm_info_get_subdevice_name)           \
+  X(snd_pcm_info_set_subdevice)                \
+  X(snd_pcm_info_get_id)                       \
+  X(snd_pcm_info_set_device)                   \
+  X(snd_pcm_info_set_stream)                   \
+  X(snd_pcm_info_get_name)                     \
+  X(snd_pcm_info_get_subdevices_count)         \
+  X(snd_pcm_info_sizeof)                       \
+  X(snd_pcm_hw_params)                         \
+  X(snd_pcm_hw_params_malloc)                  \
+  X(snd_pcm_hw_params_free)                    \
+  X(snd_pcm_hw_params_any)                     \
+  X(snd_pcm_hw_params_set_access)              \
+  X(snd_pcm_hw_params_set_format)              \
+  X(snd_pcm_hw_params_set_channels)            \
+  X(snd_pcm_hw_params_set_rate_near)           \
+  X(snd_pcm_hw_params_set_buffer_size_near)    \
+  X(snd_card_next)                             \
+  X(snd_card_get_name)                         \
+  X(snd_config_update)                         \
+  X(snd_config_copy)                           \
+  X(snd_config_get_id)                         \
+  X(snd_ctl_open)                              \
+  X(snd_ctl_close)                             \
+  X(snd_ctl_card_info)                         \
+  X(snd_ctl_card_info_sizeof)                  \
+  X(snd_ctl_card_info_get_id)                  \
+  X(snd_ctl_card_info_get_name)                \
+  X(snd_ctl_pcm_next_device)                   \
+  X(snd_ctl_pcm_info)                          \
+  X(snd_mixer_load)                            \
+  X(snd_mixer_free)                            \
+  X(snd_mixer_detach)                          \
+  X(snd_mixer_close)                           \
+  X(snd_mixer_open)                            \
+  X(snd_mixer_attach)                          \
+  X(snd_mixer_first_elem)                      \
+  X(snd_mixer_elem_next)                       \
+  X(snd_mixer_selem_get_name)                  \
+  X(snd_mixer_selem_is_active)                 \
+  X(snd_mixer_selem_register)                  \
+  X(snd_mixer_selem_set_playback_volume_all)   \
+  X(snd_mixer_selem_get_playback_volume)       \
+  X(snd_mixer_selem_has_playback_volume)       \
+  X(snd_mixer_selem_get_playback_volume_range) \
+  X(snd_mixer_selem_has_playback_switch)       \
+  X(snd_mixer_selem_get_playback_switch)       \
+  X(snd_mixer_selem_set_playback_switch_all)   \
+  X(snd_mixer_selem_has_capture_switch)        \
+  X(snd_mixer_selem_get_capture_switch)        \
+  X(snd_mixer_selem_set_capture_switch_all)    \
+  X(snd_mixer_selem_has_capture_volume)        \
+  X(snd_mixer_selem_set_capture_volume_all)    \
+  X(snd_mixer_selem_get_capture_volume)        \
+  X(snd_mixer_selem_get_capture_volume_range)  \
+  X(snd_dlopen)                                \
+  X(snd_dlclose)                               \
+  X(snd_config)                                \
+  X(snd_config_search)                         \
+  X(snd_config_get_string)                     \
+  X(snd_config_search_definition)              \
+  X(snd_config_get_type)                       \
+  X(snd_config_delete)                         \
+  X(snd_config_iterator_entry)                 \
+  X(snd_config_iterator_first)                 \
+  X(snd_config_iterator_next)                  \
+  X(snd_config_iterator_end)                   \
+  X(snd_config_delete_compound_members)        \
+  X(snd_config_get_integer)                    \
+  X(snd_config_get_bool)                       \
+  X(snd_dlsym)                                 \
+  X(snd_strerror)                              \
+  X(snd_lib_error)                             \
+  X(snd_lib_error_set_handler)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
+ALSA_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
+
+}  // namespace adm_linux_alsa
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h

@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H_
+
+#include <memory>
+
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+#include <alsa/asoundlib.h>
+#include <sys/ioctl.h>
+#include <sys/soundcard.h>
+
+typedef webrtc::adm_linux_alsa::AlsaSymbolTable WebRTCAlsaSymbolTable;
+WebRTCAlsaSymbolTable* GetAlsaSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxALSA();
+  virtual ~AudioDeviceLinuxALSA();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+ private:
+  int32_t InitRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopRecordingLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t StopPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitPlayoutLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t GetDevicesInfo(int32_t function,
+                         bool playback,
+                         int32_t enumDeviceNo = 0,
+                         char* enumDeviceName = NULL,
+                         int32_t ednLen = 0) const;
+  int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
+
+  bool KeyPressed() const;
+
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+
+  inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
+  inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  Mutex mutex_;
+
+  rtc::PlatformThread _ptrThreadRec;
+  rtc::PlatformThread _ptrThreadPlay;
+
+  AudioMixerManagerLinuxALSA _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  snd_pcm_t* _handleRecord;
+  snd_pcm_t* _handlePlayout;
+
+  snd_pcm_uframes_t _recordingBuffersizeInFrame;
+  snd_pcm_uframes_t _recordingPeriodSizeInFrame;
+  snd_pcm_uframes_t _playoutBufferSizeInFrame;
+  snd_pcm_uframes_t _playoutPeriodSizeInFrame;
+
+  ssize_t _recordingBufferSizeIn10MS;
+  ssize_t _playoutBufferSizeIn10MS;
+  uint32_t _recordingFramesIn10MS;
+  uint32_t _playoutFramesIn10MS;
+
+  uint32_t _recordingFreq;
+  uint32_t _playoutFreq;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  int8_t* _recordingBuffer;  // in byte
+  int8_t* _playoutBuffer;    // in byte
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+
+  snd_pcm_sframes_t _recordingDelay;
+  snd_pcm_sframes_t _playoutDelay;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h

@@ -0,0 +1,349 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H_
+
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+#if defined(WEBRTC_USE_X11)
+#include <X11/Xlib.h>
+#endif
+
+#include <pulse/pulseaudio.h>
+#include <stddef.h>
+#include <stdint.h>
+
+// We define this flag if it's missing from our headers, because we want to be
+// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
+// if run against a recent version of the library.
+#ifndef PA_STREAM_ADJUST_LATENCY
+#define PA_STREAM_ADJUST_LATENCY 0x2000U
+#endif
+#ifndef PA_STREAM_START_MUTED
+#define PA_STREAM_START_MUTED 0x1000U
+#endif
+
+// Set this constant to 0 to disable latency reading
+const uint32_t WEBRTC_PA_REPORT_LATENCY = 1;
+
+// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
+
+// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
+const uint32_t WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
+
+// Some timing constants for optimal operation. See
+// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
+// for a good explanation of some of the factors that go into this.
+
+// Playback.
+
+// For playback, there is a round-trip delay to fill the server-side playback
+// buffer, so setting too low of a latency is a buffer underflow risk. We will
+// automatically increase the latency if a buffer underflow does occur, but we
+// also enforce a sane minimum at start-up time. Anything lower would be
+// virtually guaranteed to underflow at least once, so there's no point in
+// allowing lower latencies.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
+
+// Every time a playback stream underflows, we will reconfigure it with target
+// latency that is greater by this amount.
+const uint32_t WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
+
+// We also need to configure a suitable request size. Too small and we'd burn
+// CPU from the overhead of transfering small amounts of data at once. Too large
+// and the amount of data remaining in the buffer right before refilling it
+// would be a buffer underflow risk. We set it to half of the buffer size.
+const uint32_t WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
+
+// Capture.
+
+// For capture, low latency is not a buffer overflow risk, but it makes us burn
+// CPU from the overhead of transfering small amounts of data at once, so we set
+// a recommended value that we use for the kLowLatency constant (but if the user
+// explicitly requests something lower then we will honour it).
+// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
+const uint32_t WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
+
+// There is a round-trip delay to ack the data to the server, so the
+// server-side buffer needs extra space to prevent buffer overflow. 20ms is
+// sufficient, but there is no penalty to making it bigger, so we make it huge.
+// (750ms is libpulse's default value for the _total_ buffer size in the
+// kNoLatencyRequirements case.)
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
+
+const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
+
+// Init _configuredLatencyRec/Play to this value to disable latency requirements
+const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
+
+// Set this const to 1 to account for peeked and used data in latency
+// calculation
+const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
+
+typedef webrtc::adm_linux_pulse::PulseAudioSymbolTable WebRTCPulseSymbolTable;
+WebRTCPulseSymbolTable* GetPulseSymbolTable();
+
+namespace webrtc {
+
+class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
+ public:
+  AudioDeviceLinuxPulse();
+  virtual ~AudioDeviceLinuxPulse();
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Playing() const override;
+  int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_) override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+ private:
+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(mutex_) { mutex_.Lock(); }
+  void UnLock() RTC_UNLOCK_FUNCTION(mutex_) { mutex_.Unlock(); }
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+  void WaitForSuccess(pa_operation* paOperation) const;
+
+  bool KeyPressed() const;
+
+  static void PaContextStateCallback(pa_context* c, void* pThis);
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaServerInfoCallback(pa_context* c,
+                                   const pa_server_info* i,
+                                   void* pThis);
+  static void PaStreamStateCallback(pa_stream* p, void* pThis);
+  void PaContextStateCallbackHandler(pa_context* c);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+  void PaServerInfoCallbackHandler(const pa_server_info* i);
+  void PaStreamStateCallbackHandler(pa_stream* p);
+
+  void EnableWriteCallback();
+  void DisableWriteCallback();
+  static void PaStreamWriteCallback(pa_stream* unused,
+                                    size_t buffer_space,
+                                    void* pThis);
+  void PaStreamWriteCallbackHandler(size_t buffer_space);
+  static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamUnderflowCallbackHandler();
+  void EnableReadCallback();
+  void DisableReadCallback();
+  static void PaStreamReadCallback(pa_stream* unused1,
+                                   size_t unused2,
+                                   void* pThis);
+  void PaStreamReadCallbackHandler();
+  static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
+  void PaStreamOverflowCallbackHandler();
+  int32_t LatencyUsecs(pa_stream* stream);
+  int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
+  int32_t ProcessRecordedData(int8_t* bufferData,
+                              uint32_t bufferSizeInSamples,
+                              uint32_t recDelay);
+
+  int32_t CheckPulseAudioVersion();
+  int32_t InitSamplingFrequency();
+  int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
+  int32_t InitPulseAudio();
+  int32_t TerminatePulseAudio();
+
+  void PaLock();
+  void PaUnLock();
+
+  static void RecThreadFunc(void*);
+  static void PlayThreadFunc(void*);
+  bool RecThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+  bool PlayThreadProcess() RTC_LOCKS_EXCLUDED(mutex_);
+
+  AudioDeviceBuffer* _ptrAudioBuffer;
+
+  mutable Mutex mutex_;
+  rtc::Event _timeEventRec;
+  rtc::Event _timeEventPlay;
+  rtc::Event _recStartEvent;
+  rtc::Event _playStartEvent;
+
+  rtc::PlatformThread _ptrThreadPlay;
+  rtc::PlatformThread _ptrThreadRec;
+
+  AudioMixerManagerLinuxPulse _mixerManager;
+
+  uint16_t _inputDeviceIndex;
+  uint16_t _outputDeviceIndex;
+  bool _inputDeviceIsSpecified;
+  bool _outputDeviceIsSpecified;
+
+  int sample_rate_hz_;
+  uint8_t _recChannels;
+  uint8_t _playChannels;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+
+  bool _initialized;
+  bool _recording;
+  bool _playing;
+  bool _recIsInitialized;
+  bool _playIsInitialized;
+  bool _startRec;
+  bool _startPlay;
+  bool update_speaker_volume_at_startup_;
+  bool quit_ RTC_GUARDED_BY(&mutex_);
+
+  uint32_t _sndCardPlayDelay RTC_GUARDED_BY(&mutex_);
+
+  int32_t _writeErrors;
+
+  uint16_t _deviceIndex;
+  int16_t _numPlayDevices;
+  int16_t _numRecDevices;
+  char* _playDeviceName;
+  char* _recDeviceName;
+  char* _playDisplayDeviceName;
+  char* _recDisplayDeviceName;
+  char _paServerVersion[32];
+
+  int8_t* _playBuffer;
+  size_t _playbackBufferSize;
+  size_t _playbackBufferUnused;
+  size_t _tempBufferSpace;
+  int8_t* _recBuffer;
+  size_t _recordBufferSize;
+  size_t _recordBufferUsed;
+  const void* _tempSampleData;
+  size_t _tempSampleDataSize;
+  int32_t _configuredLatencyPlay;
+  int32_t _configuredLatencyRec;
+
+  // PulseAudio
+  uint16_t _paDeviceIndex;
+  bool _paStateChanged;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_mainloop_api* _paMainloopApi;
+  pa_context* _paContext;
+
+  pa_stream* _recStream;
+  pa_stream* _playStream;
+  uint32_t _recStreamFlags;
+  uint32_t _playStreamFlags;
+  pa_buffer_attr _playBufferAttr;
+  pa_buffer_attr _recBufferAttr;
+
+  char _oldKeyState[32];
+#if defined(WEBRTC_USE_X11)
+  Display* _XDisplay;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.cc

@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
+
+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
+#include "rtc_base/logging.h"
+
+// Accesses ALSA functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libasound, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                                            \
+  LATESYM_GET(webrtc::adm_linux_alsa::AlsaSymbolTable, GetAlsaSymbolTable(), \
+              sym)
+
+namespace webrtc {
+
+AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA()
+    : _outputMixerHandle(NULL),
+      _inputMixerHandle(NULL),
+      _outputMixerElement(NULL),
+      _inputMixerElement(NULL) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+}
+
+AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+  Close();
+}
+
+// ============================================================================
+//                                    PUBLIC METHODS
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::Close() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  MutexLock lock(&mutex_);
+
+  CloseSpeakerLocked();
+  CloseMicrophoneLocked();
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeaker() {
+  MutexLock lock(&mutex_);
+  return CloseSpeakerLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseSpeakerLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    _outputMixerHandle = NULL;
+    _outputMixerElement = NULL;
+  }
+  memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophone() {
+  MutexLock lock(&mutex_);
+  return CloseMicrophoneLocked();
+}
+
+int32_t AudioMixerManagerLinuxALSA::CloseMicrophoneLocked() {
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  int errVal = 0;
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 2";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 3";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer 4";
+    _inputMixerHandle = NULL;
+    _inputMixerElement = NULL;
+  }
+  memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenSpeaker(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing output mixer handle
+  //
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing playout mixer";
+
+    LATE(snd_mixer_free)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching playout mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    errVal = LATE(snd_mixer_close)(_outputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+  }
+  _outputMixerHandle = NULL;
+  _outputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_outputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_outputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_outputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_outputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  // Load and find the proper mixer element
+  if (LoadSpeakerMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_outputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the output mixer device is now open ("
+                        << _outputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::OpenMicrophone(char* deviceName) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::OpenMicrophone(name="
+                      << deviceName << ")";
+
+  MutexLock lock(&mutex_);
+
+  int errVal = 0;
+
+  // Close any existing input mixer handle
+  //
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    LATE(snd_mixer_free)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error freeing record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error detaching record mixer: "
+                        << LATE(snd_strerror)(errVal);
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+
+    errVal = LATE(snd_mixer_close)(_inputMixerHandle);
+    if (errVal < 0) {
+      RTC_LOG(LS_ERROR) << "Error snd_mixer_close(handleMixer) errVal="
+                        << errVal;
+    }
+    RTC_LOG(LS_VERBOSE) << "Closing record mixer";
+  }
+  _inputMixerHandle = NULL;
+  _inputMixerElement = NULL;
+
+  errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_open(&_inputMixerHandle, 0) - error";
+    return -1;
+  }
+
+  char controlName[kAdmMaxDeviceNameSize] = {0};
+  GetControlName(controlName, deviceName);
+
+  RTC_LOG(LS_VERBOSE) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ")";
+
+  errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_attach(_inputMixerHandle, " << controlName
+                      << ") error: " << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  strcpy(_inputMixerStr, controlName);
+
+  errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR)
+        << "snd_mixer_selem_register(_inputMixerHandle, NULL, NULL), "
+           "error: "
+        << LATE(snd_strerror)(errVal);
+
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+  // Load and find the proper mixer element
+  if (LoadMicMixerElement() < 0) {
+    return -1;
+  }
+
+  if (_inputMixerHandle != NULL) {
+    RTC_LOG(LS_VERBOSE) << "the input mixer device is now open ("
+                        << _inputMixerHandle << ")";
+  }
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_outputMixerHandle != NULL);
+}
+
+bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_inputMixerHandle != NULL);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_set_playback_volume_all)(
+      _outputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing master volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolume(uint32_t& volume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting outputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SpeakerVolume() => vol="
+                      << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avilable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_playback_volume_range)(
+      _outputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting get_playback_volume_range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxSpeakerVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = maxVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinSpeakerVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinSpeakerVolume(
+     uint32_t minVolume)
+ {
+
+ if (_outputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_playback_volume_range(
+ _outputMixerElement, &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting playback volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ minVol = minVolume;
+ errVal = snd_mixer_selem_set_playback_volume_range(
+ _outputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Playout hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting playback volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(bool& available) {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available(false);
+  SpeakerMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal = LATE(snd_mixer_selem_set_playback_switch_all)(
+      _outputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const {
+  if (_outputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable output mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected speaker destination has a valid mute control.
+  bool available =
+      LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the speaker";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_playback_switch)(
+      _outputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting playback switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable) {
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable="
+                      << enable << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available(false);
+  MicrophoneMuteIsAvailable(available);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement, !enable);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error setting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer exists";
+    return -1;
+  }
+
+  // Ensure that the selected microphone destination has a valid mute control.
+  bool available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
+  if (!available) {
+    RTC_LOG(LS_WARNING) << "it is not possible to mute the microphone";
+    return -1;
+  }
+
+  int value(false);
+
+  // Retrieve one boolean control value for a specified mute-control
+  //
+  int errVal = LATE(snd_mixer_selem_get_capture_switch)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &value);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting capture switch: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  // Note value = 0 (off) means muted
+  enabled = (bool)!value;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  MutexLock lock(&mutex_);
+
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  int errVal =
+      LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement, volume);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error changing microphone volume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+
+  return (0);
+}
+
+// TL: Have done testnig with these but they don't seem reliable and
+// they were therefore not added
+/*
+ // ----------------------------------------------------------------------------
+ //    SetMaxMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMaxMicrophoneVolume(
+     uint32_t maxVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(_inputMixerElement,
+  &minVol, &maxVol);
+ if ((maxVol <= minVol) || (errVal != 0))
+ {
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ maxVol = (long int)maxVolume;
+ printf("min %d max %d", minVol, maxVol);
+ errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol,
+ maxVol); RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " <<
+ minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+
+ // ----------------------------------------------------------------------------
+ //    SetMinMicrophoneVolume
+ // ----------------------------------------------------------------------------
+
+ int32_t AudioMixerManagerLinuxALSA::SetMinMicrophoneVolume(
+ uint32_t minVolume)
+ {
+
+ if (_inputMixerElement == NULL)
+ {
+ RTC_LOG(LS_WARNING) << "no avaliable output mixer element exists";
+ return -1;
+ }
+
+ long int minVol(0);
+ long int maxVol(0);
+
+ int errVal = snd_mixer_selem_get_capture_volume_range(
+ _inputMixerElement, &minVol, &maxVol);
+ if (maxVol <= minVol)
+ {
+ //maxVol = 255;
+ RTC_LOG(LS_WARNING) << "Error getting capture volume range: "
+                 << snd_strerror(errVal);
+ }
+
+ printf("min %d max %d", minVol, maxVol);
+ minVol = (long int)minVolume;
+ errVal = snd_mixer_selem_set_capture_volume_range(
+ _inputMixerElement, minVol, maxVol);
+ RTC_LOG(LS_VERBOSE) << "Capture hardware volume range, min: " << minVol
+                 << ", max: " << maxVol;
+ if (errVal != 0)
+ {
+ RTC_LOG(LS_ERROR) << "Error setting capture volume range: "
+               << snd_strerror(errVal);
+ return -1;
+ }
+
+ return 0;
+ }
+ */
+
+int32_t AudioMixerManagerLinuxALSA::MicrophoneVolume(uint32_t& volume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int vol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume)(
+      _inputMixerElement, (snd_mixer_selem_channel_id_t)0, &vol);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "Error getting inputvolume: "
+                      << LATE(snd_strerror)(errVal);
+    return -1;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol=" << vol;
+
+  volume = static_cast<uint32_t>(vol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  // check if we have mic volume at all
+  if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement)) {
+    RTC_LOG(LS_ERROR) << "No microphone volume available";
+    return -1;
+  }
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  maxVolume = static_cast<uint32_t>(maxVol);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_inputMixerElement == NULL) {
+    RTC_LOG(LS_WARNING) << "no avaliable input mixer element exists";
+    return -1;
+  }
+
+  long int minVol(0);
+  long int maxVol(0);
+
+  int errVal = LATE(snd_mixer_selem_get_capture_volume_range)(
+      _inputMixerElement, &minVol, &maxVol);
+
+  RTC_LOG(LS_VERBOSE) << "Microphone hardware volume range, min: " << minVol
+                      << ", max: " << maxVol;
+  if (maxVol <= minVol) {
+    RTC_LOG(LS_ERROR) << "Error getting microphone volume range: "
+                      << LATE(snd_strerror)(errVal);
+  }
+
+  minVolume = static_cast<uint32_t>(minVol);
+
+  return 0;
+}
+
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
+
+int32_t AudioMixerManagerLinuxALSA::LoadMicMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_inputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _inputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* micElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      if (strcmp(selemName, "Capture") == 0)  // "Capture", "Mic"
+      {
+        _inputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "Capture element set";
+      } else if (strcmp(selemName, "Mic") == 0) {
+        micElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Mic element found";
+      }
+    }
+
+    if (_inputMixerElement) {
+      // Use the first Capture element that is found
+      // The second one may not work
+      break;
+    }
+  }
+
+  if (_inputMixerElement == NULL) {
+    // We didn't find a Capture handle, use Mic.
+    if (micElem != NULL) {
+      _inputMixerElement = micElem;
+      RTC_LOG(LS_VERBOSE) << "Using Mic as capture volume.";
+    } else {
+      _inputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find capture volume on the mixer.";
+
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const {
+  int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
+  if (errVal < 0) {
+    RTC_LOG(LS_ERROR) << "snd_mixer_load(_outputMixerHandle), error: "
+                      << LATE(snd_strerror)(errVal);
+    _outputMixerHandle = NULL;
+    return -1;
+  }
+
+  snd_mixer_elem_t* elem = NULL;
+  snd_mixer_elem_t* masterElem = NULL;
+  snd_mixer_elem_t* speakerElem = NULL;
+  unsigned mixerIdx = 0;
+  const char* selemName = NULL;
+
+  // Find and store handles to the right mixer elements
+  for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem;
+       elem = LATE(snd_mixer_elem_next)(elem), mixerIdx++) {
+    if (LATE(snd_mixer_selem_is_active)(elem)) {
+      selemName = LATE(snd_mixer_selem_get_name)(elem);
+      RTC_LOG(LS_VERBOSE) << "snd_mixer_selem_get_name " << mixerIdx << ": "
+                          << selemName << " =" << elem;
+
+      // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
+      if (strcmp(selemName, "PCM") == 0) {
+        _outputMixerElement = elem;
+        RTC_LOG(LS_VERBOSE) << "PCM element set";
+      } else if (strcmp(selemName, "Master") == 0) {
+        masterElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Master element found";
+      } else if (strcmp(selemName, "Speaker") == 0) {
+        speakerElem = elem;
+        RTC_LOG(LS_VERBOSE) << "Speaker element found";
+      }
+    }
+
+    if (_outputMixerElement) {
+      // We have found the element we want
+      break;
+    }
+  }
+
+  // If we didn't find a PCM Handle, use Master or Speaker
+  if (_outputMixerElement == NULL) {
+    if (masterElem != NULL) {
+      _outputMixerElement = masterElem;
+      RTC_LOG(LS_VERBOSE) << "Using Master as output volume.";
+    } else if (speakerElem != NULL) {
+      _outputMixerElement = speakerElem;
+      RTC_LOG(LS_VERBOSE) << "Using Speaker as output volume.";
+    } else {
+      _outputMixerElement = NULL;
+      RTC_LOG(LS_ERROR) << "Could not find output volume in the mixer.";
+      return -1;
+    }
+  }
+
+  return 0;
+}
+
+void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
+                                                char* deviceName) const {
+  // Example
+  // deviceName: "front:CARD=Intel,DEV=0"
+  // controlName: "hw:CARD=Intel"
+  char* pos1 = strchr(deviceName, ':');
+  char* pos2 = strchr(deviceName, ',');
+  if (!pos2) {
+    // Can also be default:CARD=Intel
+    pos2 = &deviceName[strlen(deviceName)];
+  }
+  if (pos1 && pos2) {
+    strcpy(controlName, "hw");
+    int nChar = (int)(pos2 - pos1);
+    strncpy(&controlName[2], pos1, nChar);
+    controlName[2 + nChar] = '\0';
+  } else {
+    strcpy(controlName, deviceName);
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h

@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
+
+#include <alsa/asoundlib.h>
+
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/linux/alsasymboltable_linux.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxALSA {
+ public:
+  int32_t OpenSpeaker(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t OpenMicrophone(char* deviceName) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SetSpeakerVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume) RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t Close() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
+  int32_t CloseMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxALSA();
+  ~AudioMixerManagerLinuxALSA();
+
+ private:
+  int32_t CloseSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t CloseMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  int32_t LoadMicMixerElement() const;
+  int32_t LoadSpeakerMixerElement() const;
+  void GetControlName(char* controlName, char* deviceName) const;
+
+ private:
+  Mutex mutex_;
+  mutable snd_mixer_t* _outputMixerHandle;
+  char _outputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_t* _inputMixerHandle;
+  char _inputMixerStr[kAdmMaxDeviceNameSize];
+  mutable snd_mixer_elem_t* _outputMixerElement;
+  mutable snd_mixer_elem_t* _inputMixerElement;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.cc

@@ -0,0 +1,844 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/audio_mixer_manager_pulse_linux.h"
+
+#include <stddef.h>
+
+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Accesses Pulse functions through our late-binding symbol table instead of
+// directly. This way we don't have to link to libpulse, which means our binary
+// will work on systems that don't have it.
+#define LATE(sym)                                             \
+  LATESYM_GET(webrtc::adm_linux_pulse::PulseAudioSymbolTable, \
+              GetPulseSymbolTable(), sym)
+
+namespace webrtc {
+
+class AutoPulseLock {
+ public:
+  explicit AutoPulseLock(pa_threaded_mainloop* pa_mainloop)
+      : pa_mainloop_(pa_mainloop) {
+    LATE(pa_threaded_mainloop_lock)(pa_mainloop_);
+  }
+
+  ~AutoPulseLock() { LATE(pa_threaded_mainloop_unlock)(pa_mainloop_); }
+
+ private:
+  pa_threaded_mainloop* const pa_mainloop_;
+};
+
+AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse()
+    : _paOutputDeviceIndex(-1),
+      _paInputDeviceIndex(-1),
+      _paPlayStream(NULL),
+      _paRecStream(NULL),
+      _paMainloop(NULL),
+      _paContext(NULL),
+      _paVolume(0),
+      _paMute(0),
+      _paVolSteps(0),
+      _paSpeakerMute(false),
+      _paSpeakerVolume(PA_VOLUME_NORM),
+      _paChannels(0),
+      _paObjectsSet(false) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " created";
+}
+
+AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__ << " destroyed";
+
+  Close();
+}
+
+// ===========================================================================
+//                                    PUBLIC METHODS
+// ===========================================================================
+
+int32_t AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
+    pa_threaded_mainloop* mainloop,
+    pa_context* context) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  if (!mainloop || !context) {
+    RTC_LOG(LS_ERROR) << "could not set PulseAudio objects for mixer";
+    return -1;
+  }
+
+  _paMainloop = mainloop;
+  _paContext = context;
+  _paObjectsSet = true;
+
+  RTC_LOG(LS_VERBOSE) << "the PulseAudio objects for the mixer has been set";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::Close() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  CloseSpeaker();
+  CloseMicrophone();
+
+  _paMainloop = NULL;
+  _paContext = NULL;
+  _paObjectsSet = false;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseSpeaker() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paOutputDeviceIndex = -1;
+  _paPlayStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::CloseMicrophone() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_VERBOSE) << __FUNCTION__;
+
+  // Reset the index to -1
+  _paInputDeviceIndex = -1;
+  _paRecStream = NULL;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)";
+
+  _paPlayStream = playStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetRecStream(recStream)";
+
+  _paRecStream = recStream;
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenSpeaker(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex="
+                      << deviceIndex << ")";
+
+  // No point in opening the speaker
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects has not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // output device to control
+  _paOutputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the output mixer device is now open";
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::OpenMicrophone(uint16_t deviceIndex) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex="
+      << deviceIndex << ")";
+
+  // No point in opening the microphone
+  // if PA objects have not been set
+  if (!_paObjectsSet) {
+    RTC_LOG(LS_ERROR) << "PulseAudio objects have not been set";
+    return -1;
+  }
+
+  // Set the index for the PulseAudio
+  // input device to control
+  _paInputDeviceIndex = deviceIndex;
+
+  RTC_LOG(LS_VERBOSE) << "the input mixer device is now open";
+
+  return 0;
+}
+
+bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paOutputDeviceIndex != -1);
+}
+
+bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+
+  return (_paInputDeviceIndex != -1);
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume="
+                      << volume << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really set the volume if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the number of channels from the sample specification
+    const pa_sample_spec* spec = LATE(pa_stream_get_sample_spec)(_paPlayStream);
+    if (!spec) {
+      RTC_LOG(LS_ERROR) << "could not get sample specification";
+      return -1;
+    }
+
+    // Set the same volume for all channels
+    pa_cvolume cVolumes;
+    LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_volume)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), &cVolumes,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the volume to be set at connection
+    _paSpeakerVolume = volume;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set speaker volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolume(uint32_t& volume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the volume if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  } else {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = _paSpeakerVolume;
+  }
+
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SpeakerVolume() => vol="
+                      << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxSpeakerVolume(
+    uint32_t& maxVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // but PA allows up to 150 db amplification
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinSpeakerVolume(
+    uint32_t& minVolume) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE) << "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable="
+                      << enable << ")";
+
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only really mute if we have a connected stream
+    AutoPulseLock auto_lock(_paMainloop);
+
+    pa_operation* paOperation = NULL;
+    paOperation = LATE(pa_context_set_sink_input_mute)(
+        _paContext, LATE(pa_stream_get_index)(_paPlayStream), (int)enable,
+        PaSetVolumeCallback, NULL);
+    if (!paOperation) {
+      setFailed = true;
+    }
+
+    // Don't need to wait for the completion
+    LATE(pa_operation_unref)(paOperation);
+  } else {
+    // We have not created a stream or it's not connected to the sink
+    // Save the mute status to be set at connection
+    _paSpeakerMute = enable;
+  }
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute speaker, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const {
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  if (_paPlayStream &&
+      (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+    // We can only get the mute status if we have a connected stream
+    if (!GetSinkInputInfo())
+      return -1;
+
+    enabled = static_cast<bool>(_paMute);
+  } else {
+    enabled = _paSpeakerMute;
+  }
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SpeakerMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paOutputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "output device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paOutputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paPlayStream &&
+        (LATE(pa_stream_get_state)(_paPlayStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
+    }
+  }
+
+  if (!GetSinkInfoByIndex(deviceIndex))
+    return -1;
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  pa_operation* paOperation = NULL;
+
+  // Get info for this source
+  // We want to know if the actual device can record in stereo
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  available = static_cast<bool>(_paChannels == 2);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
+         " => available="
+      << available;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=" << enable
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  AutoPulseLock auto_lock(_paMainloop);
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  // Set mute switch for the source
+  paOperation = LATE(pa_context_set_source_mute_by_index)(
+      _paContext, deviceIndex, enable, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not mute microphone, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream
+    // The device used by the stream can be changed
+    // during the call
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  enabled = static_cast<bool>(_paMute);
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneMute() => enabled=" << enabled;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
+    bool& available) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Always available in Pulse Audio
+  available = true;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::SetMicrophoneVolume(uint32_t volume) {
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=" << volume
+      << ")";
+
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // Unlike output streams, input streams have no concept of a stream
+  // volume, only a device volume. So we have to change the volume of the
+  // device itself.
+
+  // The device may have a different number of channels than the stream and
+  // their mapping may be different, so we don't want to use the channel
+  // count from our sample spec. We could use PA_CHANNELS_MAX to cover our
+  // bases, and the server allows that even if the device's channel count
+  // is lower, but some buggy PA clients don't like that (the pavucontrol
+  // on Hardy dies in an assert if the channel count is different). So
+  // instead we look up the actual number of channels that the device has.
+  AutoPulseLock auto_lock(_paMainloop);
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  // Get the actual stream device index if we have a connected stream
+  // The device used by the stream can be changed
+  // during the call
+  if (_paRecStream &&
+      (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+    deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+  }
+
+  bool setFailed(false);
+  pa_operation* paOperation = NULL;
+
+  // Get the number of channels for this source
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, deviceIndex, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+
+  uint8_t channels = _paChannels;
+  pa_cvolume cVolumes;
+  LATE(pa_cvolume_set)(&cVolumes, channels, volume);
+
+  // Set the volume for the source
+  paOperation = LATE(pa_context_set_source_volume_by_index)(
+      _paContext, deviceIndex, &cVolumes, PaSetVolumeCallback, NULL);
+
+  if (!paOperation) {
+    setFailed = true;
+  }
+
+  // Don't need to wait for this to complete.
+  LATE(pa_operation_unref)(paOperation);
+
+  if (setFailed) {
+    RTC_LOG(LS_WARNING) << "could not set microphone volume, error="
+                        << LATE(pa_context_errno)(_paContext);
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MicrophoneVolume(uint32_t& volume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  uint32_t deviceIndex = (uint32_t)_paInputDeviceIndex;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    // Get the actual stream device index if we have a connected stream.
+    // The device used by the stream can be changed during the call.
+    if (_paRecStream &&
+        (LATE(pa_stream_get_state)(_paRecStream) != PA_STREAM_UNCONNECTED)) {
+      deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
+    }
+  }
+
+  if (!GetSourceInfoByIndex(deviceIndex))
+    return -1;
+
+  {
+    AutoPulseLock auto_lock(_paMainloop);
+    volume = static_cast<uint32_t>(_paVolume);
+  }
+
+  RTC_LOG(LS_VERBOSE)
+      << "AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=" << volume;
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(
+    uint32_t& maxVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  // PA_VOLUME_NORM corresponds to 100% (0db)
+  // PA allows up to 150 db amplification (PA_VOLUME_MAX)
+  // but that doesn't work well for all sound cards
+  maxVolume = static_cast<uint32_t>(PA_VOLUME_NORM);
+
+  return 0;
+}
+
+int32_t AudioMixerManagerLinuxPulse::MinMicrophoneVolume(
+    uint32_t& minVolume) const {
+  if (_paInputDeviceIndex == -1) {
+    RTC_LOG(LS_WARNING) << "input device index has not been set";
+    return -1;
+  }
+
+  minVolume = static_cast<uint32_t>(PA_VOLUME_MUTED);
+
+  return 0;
+}
+
+// ===========================================================================
+//                                 Private Methods
+// ===========================================================================
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context* /*c*/,
+                                                     const pa_sink_info* i,
+                                                     int eol,
+                                                     void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSinkInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
+    pa_context* /*c*/,
+    const pa_sink_input_info* i,
+    int eol,
+    void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)
+      ->PaSinkInputInfoCallbackHandler(i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context* /*c*/,
+                                                       const pa_source_info* i,
+                                                       int eol,
+                                                       void* pThis) {
+  static_cast<AudioMixerManagerLinuxPulse*>(pThis)->PaSourceInfoCallbackHandler(
+      i, eol);
+}
+
+void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context* c,
+                                                      int success,
+                                                      void* /*pThis*/) {
+  if (!success) {
+    RTC_LOG(LS_ERROR) << "failed to set volume";
+  }
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
+    const pa_sink_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // get the max volume for any channel
+  _paMute = i->mute;     // get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
+    const pa_sink_input_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+}
+
+void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
+    const pa_source_info* i,
+    int eol) {
+  if (eol) {
+    // Signal that we are done
+    LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
+    return;
+  }
+
+  _paChannels = i->channel_map.channels;   // Get number of channels
+  pa_volume_t paVolume = PA_VOLUME_MUTED;  // Minimum possible value.
+  for (int j = 0; j < _paChannels; ++j) {
+    if (paVolume < i->volume.values[j]) {
+      paVolume = i->volume.values[j];
+    }
+  }
+  _paVolume = paVolume;  // Get the max volume for any channel
+  _paMute = i->mute;     // Get mute status
+
+  // supported since PA 0.9.15
+  //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
+  // default value is PA_VOLUME_NORM+1
+  _paVolSteps = PA_VOLUME_NORM + 1;
+}
+
+void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
+    pa_operation* paOperation) const {
+  while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING) {
+    LATE(pa_threaded_mainloop_wait)(_paMainloop);
+  }
+
+  LATE(pa_operation_unref)(paOperation);
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  // Get info for this stream (sink input).
+  paOperation = LATE(pa_context_get_sink_input_info)(
+      _paContext, LATE(pa_stream_get_index)(_paPlayStream),
+      PaSinkInputInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_sink_info_by_index)(
+      _paContext, device_index, PaSinkInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(int device_index) const {
+  pa_operation* paOperation = NULL;
+
+  AutoPulseLock auto_lock(_paMainloop);
+  paOperation = LATE(pa_context_get_source_info_by_index)(
+      _paContext, device_index, PaSourceInfoCallback, (void*)this);
+
+  WaitForOperationCompletion(paOperation);
+  return true;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h

@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+#define AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
+
+#include <pulse/pulseaudio.h>
+#include <stdint.h>
+
+#include "api/sequence_checker.h"
+
+#ifndef UINT32_MAX
+#define UINT32_MAX ((uint32_t)-1)
+#endif
+
+namespace webrtc {
+
+class AudioMixerManagerLinuxPulse {
+ public:
+  int32_t SetPlayStream(pa_stream* playStream);
+  int32_t SetRecStream(pa_stream* recStream);
+  int32_t OpenSpeaker(uint16_t deviceIndex);
+  int32_t OpenMicrophone(uint16_t deviceIndex);
+  int32_t SetSpeakerVolume(uint32_t volume);
+  int32_t SpeakerVolume(uint32_t& volume) const;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+  int32_t SpeakerVolumeIsAvailable(bool& available);
+  int32_t SpeakerMuteIsAvailable(bool& available);
+  int32_t SetSpeakerMute(bool enable);
+  int32_t StereoPlayoutIsAvailable(bool& available);
+  int32_t StereoRecordingIsAvailable(bool& available);
+  int32_t SpeakerMute(bool& enabled) const;
+  int32_t MicrophoneMuteIsAvailable(bool& available);
+  int32_t SetMicrophoneMute(bool enable);
+  int32_t MicrophoneMute(bool& enabled) const;
+  int32_t MicrophoneVolumeIsAvailable(bool& available);
+  int32_t SetMicrophoneVolume(uint32_t volume);
+  int32_t MicrophoneVolume(uint32_t& volume) const;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+  int32_t SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
+                               pa_context* context);
+  int32_t Close();
+  int32_t CloseSpeaker();
+  int32_t CloseMicrophone();
+  bool SpeakerIsInitialized() const;
+  bool MicrophoneIsInitialized() const;
+
+ public:
+  AudioMixerManagerLinuxPulse();
+  ~AudioMixerManagerLinuxPulse();
+
+ private:
+  static void PaSinkInfoCallback(pa_context* c,
+                                 const pa_sink_info* i,
+                                 int eol,
+                                 void* pThis);
+  static void PaSinkInputInfoCallback(pa_context* c,
+                                      const pa_sink_input_info* i,
+                                      int eol,
+                                      void* pThis);
+  static void PaSourceInfoCallback(pa_context* c,
+                                   const pa_source_info* i,
+                                   int eol,
+                                   void* pThis);
+  static void PaSetVolumeCallback(pa_context* /*c*/,
+                                  int success,
+                                  void* /*pThis*/);
+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
+  void PaSinkInputInfoCallbackHandler(const pa_sink_input_info* i, int eol);
+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
+
+  void WaitForOperationCompletion(pa_operation* paOperation) const;
+
+  bool GetSinkInputInfo() const;
+  bool GetSinkInfoByIndex(int device_index) const;
+  bool GetSourceInfoByIndex(int device_index) const;
+
+ private:
+  int16_t _paOutputDeviceIndex;
+  int16_t _paInputDeviceIndex;
+
+  pa_stream* _paPlayStream;
+  pa_stream* _paRecStream;
+
+  pa_threaded_mainloop* _paMainloop;
+  pa_context* _paContext;
+
+  mutable uint32_t _paVolume;
+  mutable uint32_t _paMute;
+  mutable uint32_t _paVolSteps;
+  bool _paSpeakerMute;
+  mutable uint32_t _paSpeakerVolume;
+  mutable uint8_t _paChannels;
+  bool _paObjectsSet;
+
+  // Stores thread ID in constructor.
+  // We can then use RTC_DCHECK_RUN_ON(&worker_thread_checker_) to ensure that
+  // other methods are called from the same thread.
+  // Currently only does RTC_DCHECK(thread_checker_.IsCurrent()).
+  SequenceChecker thread_checker_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.cc

@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/logging.h"
+
+#ifdef WEBRTC_LINUX
+#include <dlfcn.h>
+#endif
+
+namespace webrtc {
+namespace adm_linux {
+
+inline static const char* GetDllError() {
+#ifdef WEBRTC_LINUX
+  char* err = dlerror();
+  if (err) {
+    return err;
+  } else {
+    return "No error";
+  }
+#else
+#error Not implemented
+#endif
+}
+
+DllHandle InternalLoadDll(absl::string_view dll_name) {
+#ifdef WEBRTC_LINUX
+  DllHandle handle = dlopen(std::string(dll_name).c_str(), RTLD_NOW);
+#else
+#error Not implemented
+#endif
+  if (handle == kInvalidDllHandle) {
+    RTC_LOG(LS_WARNING) << "Can't load " << dll_name << " : " << GetDllError();
+  }
+  return handle;
+}
+
+void InternalUnloadDll(DllHandle handle) {
+#ifdef WEBRTC_LINUX
+// TODO(pbos): Remove this dlclose() exclusion when leaks and suppressions from
+// here are gone (or AddressSanitizer can display them properly).
+//
+// Skip dlclose() on AddressSanitizer as leaks including this module in the
+// stack trace gets displayed as <unknown module> instead of the actual library
+// -> it can not be suppressed.
+// https://code.google.com/p/address-sanitizer/issues/detail?id=89
+#if !defined(ADDRESS_SANITIZER)
+  if (dlclose(handle) != 0) {
+    RTC_LOG(LS_ERROR) << GetDllError();
+  }
+#endif  // !defined(ADDRESS_SANITIZER)
+#else
+#error Not implemented
+#endif
+}
+
+static bool LoadSymbol(DllHandle handle,
+                       absl::string_view symbol_name,
+                       void** symbol) {
+#ifdef WEBRTC_LINUX
+  *symbol = dlsym(handle, std::string(symbol_name).c_str());
+  char* err = dlerror();
+  if (err) {
+    RTC_LOG(LS_ERROR) << "Error loading symbol " << symbol_name << " : " << err;
+    return false;
+  } else if (!*symbol) {
+    RTC_LOG(LS_ERROR) << "Symbol " << symbol_name << " is NULL";
+    return false;
+  }
+  return true;
+#else
+#error Not implemented
+#endif
+}
+
+// This routine MUST assign SOME value for every symbol, even if that value is
+// NULL, or else some symbols may be left with uninitialized data that the
+// caller may later interpret as a valid address.
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]) {
+#ifdef WEBRTC_LINUX
+  // Clear any old errors.
+  dlerror();
+#endif
+  for (int i = 0; i < num_symbols; ++i) {
+    if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
+      return false;
+    }
+  }
+  return true;
+}
+
+}  // namespace adm_linux
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h

@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2010 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H_
+
+#include <stddef.h>  // for NULL
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+// This file provides macros for creating "symbol table" classes to simplify the
+// dynamic loading of symbols from DLLs. Currently the implementation only
+// supports Linux and pure C symbols.
+// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
+
+namespace webrtc {
+namespace adm_linux {
+
+#ifdef WEBRTC_LINUX
+typedef void* DllHandle;
+
+const DllHandle kInvalidDllHandle = NULL;
+#else
+#error Not implemented
+#endif
+
+// These are helpers for use only by the class below.
+DllHandle InternalLoadDll(absl::string_view);
+
+void InternalUnloadDll(DllHandle handle);
+
+bool InternalLoadSymbols(DllHandle handle,
+                         int num_symbols,
+                         const char* const symbol_names[],
+                         void* symbols[]);
+
+template <int SYMBOL_TABLE_SIZE,
+          const char kDllName[],
+          const char* const kSymbolNames[]>
+class LateBindingSymbolTable {
+ public:
+  LateBindingSymbolTable()
+      : handle_(kInvalidDllHandle), undefined_symbols_(false) {
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  ~LateBindingSymbolTable() { Unload(); }
+
+  LateBindingSymbolTable(const LateBindingSymbolTable&) = delete;
+  LateBindingSymbolTable& operator=(LateBindingSymbolTable&) = delete;
+
+  static int NumSymbols() { return SYMBOL_TABLE_SIZE; }
+
+  // We do not use this, but we offer it for theoretical convenience.
+  static const char* GetSymbolName(int index) {
+    RTC_DCHECK_LT(index, NumSymbols());
+    return kSymbolNames[index];
+  }
+
+  bool IsLoaded() const { return handle_ != kInvalidDllHandle; }
+
+  // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
+  // table loaded successfully.
+  bool Load() {
+    if (IsLoaded()) {
+      return true;
+    }
+    if (undefined_symbols_) {
+      // We do not attempt to load again because repeated attempts are not
+      // likely to succeed and DLL loading is costly.
+      return false;
+    }
+    handle_ = InternalLoadDll(kDllName);
+    if (!IsLoaded()) {
+      return false;
+    }
+    if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
+      undefined_symbols_ = true;
+      Unload();
+      return false;
+    }
+    return true;
+  }
+
+  void Unload() {
+    if (!IsLoaded()) {
+      return;
+    }
+    InternalUnloadDll(handle_);
+    handle_ = kInvalidDllHandle;
+    memset(symbols_, 0, sizeof(symbols_));
+  }
+
+  // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
+  // instead of this.
+  void* GetSymbol(int index) const {
+    RTC_DCHECK(IsLoaded());
+    RTC_DCHECK_LT(index, NumSymbols());
+    return symbols_[index];
+  }
+
+ private:
+  DllHandle handle_;
+  bool undefined_symbols_;
+  void* symbols_[SYMBOL_TABLE_SIZE];
+};
+
+// This macro must be invoked in a header to declare a symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) enum {
+// This macro must be invoked in the header declaration once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This macro defines an enum with names built from the symbols, which
+// essentially creates a hash table in the compiler from symbol names to their
+// indices in the symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
+  ClassName##_SYMBOL_TABLE_INDEX_##sym,
+
+// This macro completes the header declaration.
+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName)       \
+  ClassName##_SYMBOL_TABLE_SIZE                                \
+  }                                                            \
+  ;                                                            \
+                                                               \
+  extern const char ClassName##_kDllName[];                    \
+  extern const char* const                                     \
+      ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
+                                                               \
+  typedef ::webrtc::adm_linux::LateBindingSymbolTable<         \
+      ClassName##_SYMBOL_TABLE_SIZE, ClassName##_kDllName,     \
+      ClassName##_kSymbolNames>                                \
+      ClassName;
+
+// This macro must be invoked in a .cc file to define a previously-declared
+// symbol table class.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
+  const char ClassName##_kDllName[] = dllName;                     \
+  const char* const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
+// This macro must be invoked in the .cc definition once for each symbol
+// (recommended to use an X-Macro to avoid duplication).
+// This would have to use the mangled name if we were to ever support C++
+// symbols.
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) #sym,
+
+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
+  }                                                     \
+  ;
+
+// Index of a given symbol in the given symbol table class.
+#define LATESYM_INDEXOF(ClassName, sym) (ClassName##_SYMBOL_TABLE_INDEX_##sym)
+
+// Returns a reference to the given late-binded symbol, with the correct type.
+#define LATESYM_GET(ClassName, inst, sym) \
+  (*reinterpret_cast<__typeof__(&sym)>(   \
+      (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
+
+}  // namespace adm_linux
+}  // namespace webrtc
+
+#endif  // ADM_LATEBINDINGSYMBOLTABLE_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc

@@ -0,0 +1,41 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h

@@ -0,0 +1,106 @@
+/*
+ * libjingle
+ * Copyright 2004--2010, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+#define AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_
+
+#include "modules/audio_device/linux/latebindingsymboltable_linux.h"
+
+namespace webrtc {
+namespace adm_linux_pulse {
+
+// The PulseAudio symbols we need, as an X-Macro list.
+// This list must contain precisely every libpulse function that is used in
+// the ADM LINUX PULSE Device and Mixer classes
+#define PULSE_AUDIO_SYMBOLS_LIST           \
+  X(pa_bytes_per_second)                   \
+  X(pa_context_connect)                    \
+  X(pa_context_disconnect)                 \
+  X(pa_context_errno)                      \
+  X(pa_context_get_protocol_version)       \
+  X(pa_context_get_server_info)            \
+  X(pa_context_get_sink_info_list)         \
+  X(pa_context_get_sink_info_by_index)     \
+  X(pa_context_get_sink_info_by_name)      \
+  X(pa_context_get_sink_input_info)        \
+  X(pa_context_get_source_info_by_index)   \
+  X(pa_context_get_source_info_by_name)    \
+  X(pa_context_get_source_info_list)       \
+  X(pa_context_get_state)                  \
+  X(pa_context_new)                        \
+  X(pa_context_set_sink_input_volume)      \
+  X(pa_context_set_sink_input_mute)        \
+  X(pa_context_set_source_volume_by_index) \
+  X(pa_context_set_source_mute_by_index)   \
+  X(pa_context_set_state_callback)         \
+  X(pa_context_unref)                      \
+  X(pa_cvolume_set)                        \
+  X(pa_operation_get_state)                \
+  X(pa_operation_unref)                    \
+  X(pa_stream_connect_playback)            \
+  X(pa_stream_connect_record)              \
+  X(pa_stream_disconnect)                  \
+  X(pa_stream_drop)                        \
+  X(pa_stream_get_device_index)            \
+  X(pa_stream_get_index)                   \
+  X(pa_stream_get_latency)                 \
+  X(pa_stream_get_sample_spec)             \
+  X(pa_stream_get_state)                   \
+  X(pa_stream_new)                         \
+  X(pa_stream_peek)                        \
+  X(pa_stream_readable_size)               \
+  X(pa_stream_set_buffer_attr)             \
+  X(pa_stream_set_overflow_callback)       \
+  X(pa_stream_set_read_callback)           \
+  X(pa_stream_set_state_callback)          \
+  X(pa_stream_set_underflow_callback)      \
+  X(pa_stream_set_write_callback)          \
+  X(pa_stream_unref)                       \
+  X(pa_stream_writable_size)               \
+  X(pa_stream_write)                       \
+  X(pa_strerror)                           \
+  X(pa_threaded_mainloop_free)             \
+  X(pa_threaded_mainloop_get_api)          \
+  X(pa_threaded_mainloop_lock)             \
+  X(pa_threaded_mainloop_new)              \
+  X(pa_threaded_mainloop_signal)           \
+  X(pa_threaded_mainloop_start)            \
+  X(pa_threaded_mainloop_stop)             \
+  X(pa_threaded_mainloop_unlock)           \
+  X(pa_threaded_mainloop_wait)
+
+LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
+#define X(sym) \
+  LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
+PULSE_AUDIO_SYMBOLS_LIST
+#undef X
+LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
+
+}  // namespace adm_linux_pulse
+}  // namespace webrtc
+
+#endif  // AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/mock_audio_device_buffer.h

@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
+#define MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioDeviceBuffer : public AudioDeviceBuffer {
+ public:
+  using AudioDeviceBuffer::AudioDeviceBuffer;
+  virtual ~MockAudioDeviceBuffer() {}
+  MOCK_METHOD(int32_t, RequestPlayoutData, (size_t nSamples), (override));
+  MOCK_METHOD(int32_t, GetPlayoutData, (void* audioBuffer), (override));
+  MOCK_METHOD(int32_t,
+              SetRecordedBuffer,
+              (const void* audioBuffer,
+               size_t nSamples,
+               absl::optional<int64_t> capture_time_ns),
+              (override));
+  MOCK_METHOD(void, SetVQEData, (int playDelayMS, int recDelayMS), (override));
+  MOCK_METHOD(int32_t, DeliverRecordedData, (), (override));
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/test_audio_device_impl.cc

@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_device/test_audio_device_impl.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_utils/repeating_task.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kFrameLengthUs = 10000;
+
+}
+
+TestAudioDevice::TestAudioDevice(
+    TaskQueueFactory* task_queue_factory,
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+    float speed)
+    : task_queue_factory_(task_queue_factory),
+      capturer_(std::move(capturer)),
+      renderer_(std::move(renderer)),
+      process_interval_us_(kFrameLengthUs / speed),
+      audio_buffer_(nullptr),
+      rendering_(false),
+      capturing_(false) {
+  auto good_sample_rate = [](int sr) {
+    return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 ||
+           sr == 48000;
+  };
+
+  if (renderer_) {
+    const int sample_rate = renderer_->SamplingFrequency();
+    playout_buffer_.resize(TestAudioDeviceModule::SamplesPerFrame(sample_rate) *
+                               renderer_->NumChannels(),
+                           0);
+    RTC_CHECK(good_sample_rate(sample_rate));
+  }
+  if (capturer_) {
+    RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency()));
+  }
+}
+
+AudioDeviceGeneric::InitStatus TestAudioDevice::Init() {
+  task_queue_ = task_queue_factory_->CreateTaskQueue(
+      "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL);
+
+  RepeatingTaskHandle::Start(task_queue_.get(), [this]() {
+    ProcessAudio();
+    return TimeDelta::Micros(process_interval_us_);
+  });
+  return InitStatus::OK;
+}
+
+int32_t TestAudioDevice::PlayoutIsAvailable(bool& available) {
+  MutexLock lock(&lock_);
+  available = renderer_ != nullptr;
+  return 0;
+}
+
+int32_t TestAudioDevice::InitPlayout() {
+  MutexLock lock(&lock_);
+
+  if (rendering_) {
+    return -1;
+  }
+
+  if (audio_buffer_ != nullptr && renderer_ != nullptr) {
+    // Update webrtc audio buffer with the selected parameters
+    audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency());
+    audio_buffer_->SetPlayoutChannels(renderer_->NumChannels());
+  }
+  rendering_initialized_ = true;
+  return 0;
+}
+
+bool TestAudioDevice::PlayoutIsInitialized() const {
+  MutexLock lock(&lock_);
+  return rendering_initialized_;
+}
+
+int32_t TestAudioDevice::StartPlayout() {
+  MutexLock lock(&lock_);
+  RTC_CHECK(renderer_);
+  rendering_ = true;
+  return 0;
+}
+
+int32_t TestAudioDevice::StopPlayout() {
+  MutexLock lock(&lock_);
+  rendering_ = false;
+  return 0;
+}
+
+int32_t TestAudioDevice::RecordingIsAvailable(bool& available) {
+  MutexLock lock(&lock_);
+  available = capturer_ != nullptr;
+  return 0;
+}
+
+int32_t TestAudioDevice::InitRecording() {
+  MutexLock lock(&lock_);
+
+  if (capturing_) {
+    return -1;
+  }
+
+  if (audio_buffer_ != nullptr && capturer_ != nullptr) {
+    // Update webrtc audio buffer with the selected parameters
+    audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency());
+    audio_buffer_->SetRecordingChannels(capturer_->NumChannels());
+  }
+  capturing_initialized_ = true;
+  return 0;
+}
+
+bool TestAudioDevice::RecordingIsInitialized() const {
+  MutexLock lock(&lock_);
+  return capturing_initialized_;
+}
+
+int32_t TestAudioDevice::StartRecording() {
+  MutexLock lock(&lock_);
+  capturing_ = true;
+  return 0;
+}
+
+int32_t TestAudioDevice::StopRecording() {
+  MutexLock lock(&lock_);
+  capturing_ = false;
+  return 0;
+}
+
+bool TestAudioDevice::Playing() const {
+  MutexLock lock(&lock_);
+  return rendering_;
+}
+
+bool TestAudioDevice::Recording() const {
+  MutexLock lock(&lock_);
+  return capturing_;
+}
+
+void TestAudioDevice::ProcessAudio() {
+  MutexLock lock(&lock_);
+  if (audio_buffer_ == nullptr) {
+    return;
+  }
+  if (capturing_ && capturer_ != nullptr) {
+    // Capture 10ms of audio. 2 bytes per sample.
+    const bool keep_capturing = capturer_->Capture(&recording_buffer_);
+    if (recording_buffer_.size() > 0) {
+      audio_buffer_->SetRecordedBuffer(
+          recording_buffer_.data(),
+          recording_buffer_.size() / capturer_->NumChannels(),
+          absl::make_optional(rtc::TimeNanos()));
+      audio_buffer_->DeliverRecordedData();
+    }
+    if (!keep_capturing) {
+      capturing_ = false;
+    }
+  }
+  if (rendering_) {
+    const int sampling_frequency = renderer_->SamplingFrequency();
+    int32_t samples_per_channel = audio_buffer_->RequestPlayoutData(
+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency));
+    audio_buffer_->GetPlayoutData(playout_buffer_.data());
+    size_t samples_out = samples_per_channel * renderer_->NumChannels();
+    RTC_CHECK_LE(samples_out, playout_buffer_.size());
+    const bool keep_rendering = renderer_->Render(
+        rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
+    if (!keep_rendering) {
+      rendering_ = false;
+    }
+  }
+}
+
+void TestAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
+  MutexLock lock(&lock_);
+  RTC_DCHECK(audio_buffer || audio_buffer_);
+  audio_buffer_ = audio_buffer;
+
+  if (renderer_ != nullptr) {
+    audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency());
+    audio_buffer_->SetPlayoutChannels(renderer_->NumChannels());
+  }
+  if (capturer_ != nullptr) {
+    audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency());
+    audio_buffer_->SetRecordingChannels(capturer_->NumChannels());
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_device/test_audio_device_impl.h

@@ -0,0 +1,198 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_DEVICE_TEST_AUDIO_DEVICE_IMPL_H_
+#define MODULES_AUDIO_DEVICE_TEST_AUDIO_DEVICE_IMPL_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+class TestAudioDevice : public AudioDeviceGeneric {
+ public:
+  // Creates a new TestAudioDevice. When capturing or playing, 10 ms audio
+  // frames will be processed every 10ms / `speed`.
+  // `capturer` is an object that produces audio data. Can be nullptr if this
+  // device is never used for recording.
+  // `renderer` is an object that receives audio data that would have been
+  // played out. Can be nullptr if this device is never used for playing.
+  TestAudioDevice(TaskQueueFactory* task_queue_factory,
+                  std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+                  std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+                  float speed = 1);
+  TestAudioDevice(const TestAudioDevice&) = delete;
+  TestAudioDevice& operator=(const TestAudioDevice&) = delete;
+  ~TestAudioDevice() override = default;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override {
+    return 0;
+  }
+
+  // Main initializaton and termination
+  InitStatus Init() override;
+  int32_t Terminate() override { return 0; }
+  bool Initialized() const override { return true; }
+
+  // Device enumeration
+  int16_t PlayoutDevices() override { return 0; }
+  int16_t RecordingDevices() override { return 0; }
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override {
+    return 0;
+  }
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override {
+    return 0;
+  }
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override { return 0; }
+  bool SpeakerIsInitialized() const override { return true; }
+  int32_t InitMicrophone() override { return 0; }
+  bool MicrophoneIsInitialized() const override { return true; }
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool& available) override { return 0; }
+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
+  int32_t SpeakerVolume(uint32_t& volume) const override { return 0; }
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override { return 0; }
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override { return 0; }
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override { return 0; }
+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
+  int32_t MicrophoneVolume(uint32_t& volume) const override { return 0; }
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override { return 0; }
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override { return 0; }
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool& available) override { return 0; }
+  int32_t SetSpeakerMute(bool enable) override { return 0; }
+  int32_t SpeakerMute(bool& enabled) const override { return 0; }
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool& available) override { return 0; }
+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
+  int32_t MicrophoneMute(bool& enabled) const override { return 0; }
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool& available) override {
+    available = false;
+    return 0;
+  }
+  int32_t SetStereoPlayout(bool enable) override { return 0; }
+  int32_t StereoPlayout(bool& enabled) const override { return 0; }
+  int32_t StereoRecordingIsAvailable(bool& available) override {
+    available = false;
+    return 0;
+  }
+  int32_t SetStereoRecording(bool enable) override { return 0; }
+  int32_t StereoRecording(bool& enabled) const override { return 0; }
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t& delayMS) const override {
+    delayMS = 0;
+    return 0;
+  }
+
+  // Android only
+  bool BuiltInAECIsAvailable() const override { return false; }
+  bool BuiltInAGCIsAvailable() const override { return false; }
+  bool BuiltInNSIsAvailable() const override { return false; }
+
+  // Windows Core Audio and Android only.
+  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
+  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
+  int32_t EnableBuiltInNS(bool enable) override { return -1; }
+
+  // Play underrun count.
+  int32_t GetPlayoutUnderrunCount() const override { return -1; }
+
+// iOS only.
+// TODO(henrika): add Android support.
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+  int GetRecordAudioParameters(AudioParameters* params) const override {
+    return -1;
+  }
+#endif  // WEBRTC_IOS
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
+
+ private:
+  void ProcessAudio();
+
+  TaskQueueFactory* const task_queue_factory_;
+  const std::unique_ptr<TestAudioDeviceModule::Capturer> capturer_
+      RTC_GUARDED_BY(lock_);
+  const std::unique_ptr<TestAudioDeviceModule::Renderer> renderer_
+      RTC_GUARDED_BY(lock_);
+  const int64_t process_interval_us_;
+
+  mutable Mutex lock_;
+  AudioDeviceBuffer* audio_buffer_ RTC_GUARDED_BY(lock_) = nullptr;
+  bool rendering_ RTC_GUARDED_BY(lock_) = false;
+  bool capturing_ RTC_GUARDED_BY(lock_) = false;
+  bool rendering_initialized_ RTC_GUARDED_BY(lock_) = false;
+  bool capturing_initialized_ RTC_GUARDED_BY(lock_) = false;
+
+  std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
+  rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_DEVICE_TEST_AUDIO_DEVICE_IMPL_H_

TMessagesProj/jni/voip/webrtc/modules/audio_device/test_audio_device_impl_test.cc

@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_device/test_audio_device_impl.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/audio_device_defines.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+
+constexpr Timestamp kStartTime = Timestamp::Millis(10000);
+
+class TestAudioTransport : public AudioTransport {
+ public:
+  enum class Mode { kPlaying, kRecording };
+
+  explicit TestAudioTransport(Mode mode) : mode_(mode) {}
+  ~TestAudioTransport() override = default;
+
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t samples_per_channel,
+      size_t bytes_per_sample,
+      size_t number_of_channels,
+      uint32_t samples_per_second,
+      uint32_t total_delay_ms,
+      int32_t clock_drift,
+      uint32_t current_mic_level,
+      bool key_pressed,
+      uint32_t& new_mic_level,
+      absl::optional<int64_t> estimated_capture_time_ns) override {
+    new_mic_level = 1;
+
+    if (mode_ != Mode::kRecording) {
+      EXPECT_TRUE(false) << "RecordedDataIsAvailable mustn't be called when "
+                            "mode isn't kRecording";
+      return -1;
+    }
+
+    MutexLock lock(&mutex_);
+    samples_per_channel_.push_back(samples_per_channel);
+    number_of_channels_.push_back(number_of_channels);
+    bytes_per_sample_.push_back(bytes_per_sample);
+    samples_per_second_.push_back(samples_per_second);
+    return 0;
+  }
+
+  int32_t NeedMorePlayData(size_t samples_per_channel,
+                           size_t bytes_per_sample,
+                           size_t number_of_channels,
+                           uint32_t samples_per_second,
+                           void* audio_samples,
+                           size_t& samples_out,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
+    const size_t num_bytes = samples_per_channel * number_of_channels;
+    std::memset(audio_samples, 1, num_bytes);
+    samples_out = samples_per_channel * number_of_channels;
+    *elapsed_time_ms = 0;
+    *ntp_time_ms = 0;
+
+    if (mode_ != Mode::kPlaying) {
+      EXPECT_TRUE(false)
+          << "NeedMorePlayData mustn't be called when mode isn't kPlaying";
+      return -1;
+    }
+
+    MutexLock lock(&mutex_);
+    samples_per_channel_.push_back(samples_per_channel);
+    number_of_channels_.push_back(number_of_channels);
+    bytes_per_sample_.push_back(bytes_per_sample);
+    samples_per_second_.push_back(samples_per_second);
+    return 0;
+  }
+
+  int32_t RecordedDataIsAvailable(const void* audio_samples,
+                                  size_t samples_per_channel,
+                                  size_t bytes_per_sample,
+                                  size_t number_of_channels,
+                                  uint32_t samples_per_second,
+                                  uint32_t total_delay_ms,
+                                  int32_t clockDrift,
+                                  uint32_t current_mic_level,
+                                  bool key_pressed,
+                                  uint32_t& new_mic_level) override {
+    RTC_CHECK(false) << "This methods should be never executed";
+  }
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {
+    RTC_CHECK(false) << "This methods should be never executed";
+  }
+
+  std::vector<size_t> samples_per_channel() const {
+    MutexLock lock(&mutex_);
+    return samples_per_channel_;
+  }
+  std::vector<size_t> number_of_channels() const {
+    MutexLock lock(&mutex_);
+    return number_of_channels_;
+  }
+  std::vector<size_t> bytes_per_sample() const {
+    MutexLock lock(&mutex_);
+    return bytes_per_sample_;
+  }
+  std::vector<size_t> samples_per_second() const {
+    MutexLock lock(&mutex_);
+    return samples_per_second_;
+  }
+
+ private:
+  const Mode mode_;
+
+  mutable Mutex mutex_;
+  std::vector<size_t> samples_per_channel_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> number_of_channels_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> bytes_per_sample_ RTC_GUARDED_BY(mutex_);
+  std::vector<size_t> samples_per_second_ RTC_GUARDED_BY(mutex_);
+};
+
+TEST(TestAudioDeviceTest, EnablingRecordingProducesAudio) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioTransport audio_transport(TestAudioTransport::Mode::kRecording);
+  AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
+  ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          /*max_amplitude=*/1000,
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               std::move(capturer),
+                               /*renderer=*/nullptr);
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+  audio_device.AttachAudioBuffer(&audio_buffer);
+
+  EXPECT_FALSE(audio_device.RecordingIsInitialized());
+  ASSERT_EQ(audio_device.InitRecording(), 0);
+  EXPECT_TRUE(audio_device.RecordingIsInitialized());
+  audio_buffer.StartRecording();
+  ASSERT_EQ(audio_device.StartRecording(), 0);
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_TRUE(audio_device.Recording());
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_EQ(audio_device.StopRecording(), 0);
+  audio_buffer.StopRecording();
+
+  EXPECT_THAT(audio_transport.samples_per_channel(),
+              ElementsAre(480, 480, 480));
+  EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
+  EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
+  EXPECT_THAT(audio_transport.samples_per_second(),
+              ElementsAre(48000, 48000, 48000));
+}
+
+TEST(TestAudioDeviceTest, RecordingIsAvailableWhenCapturerIsSet) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          /*max_amplitude=*/1000,
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               std::move(capturer),
+                               /*renderer=*/nullptr);
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+
+  bool available;
+  EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
+  EXPECT_TRUE(available);
+}
+
+TEST(TestAudioDeviceTest, RecordingIsNotAvailableWhenCapturerIsNotSet) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               /*capturer=*/nullptr,
+                               /*renderer=*/nullptr);
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+
+  bool available;
+  EXPECT_EQ(audio_device.RecordingIsAvailable(available), 0);
+  EXPECT_FALSE(available);
+}
+
+TEST(TestAudioDeviceTest, EnablingPlayoutProducesAudio) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioTransport audio_transport(TestAudioTransport::Mode::kPlaying);
+  AudioDeviceBuffer audio_buffer(time_controller.GetTaskQueueFactory());
+  ASSERT_EQ(audio_buffer.RegisterAudioCallback(&audio_transport), 0);
+  std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
+      TestAudioDeviceModule::CreateDiscardRenderer(
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               /*capturer=*/nullptr, std::move(renderer));
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+  audio_device.AttachAudioBuffer(&audio_buffer);
+
+  EXPECT_FALSE(audio_device.PlayoutIsInitialized());
+  ASSERT_EQ(audio_device.InitPlayout(), 0);
+  EXPECT_TRUE(audio_device.PlayoutIsInitialized());
+  audio_buffer.StartPlayout();
+  ASSERT_EQ(audio_device.StartPlayout(), 0);
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_TRUE(audio_device.Playing());
+  time_controller.AdvanceTime(TimeDelta::Millis(10));
+  ASSERT_EQ(audio_device.StopPlayout(), 0);
+  audio_buffer.StopPlayout();
+
+  EXPECT_THAT(audio_transport.samples_per_channel(),
+              ElementsAre(480, 480, 480));
+  EXPECT_THAT(audio_transport.number_of_channels(), ElementsAre(2, 2, 2));
+  EXPECT_THAT(audio_transport.bytes_per_sample(), ElementsAre(4, 4, 4));
+  EXPECT_THAT(audio_transport.samples_per_second(),
+              ElementsAre(48000, 48000, 48000));
+}
+
+TEST(TestAudioDeviceTest, PlayoutIsAvailableWhenRendererIsSet) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
+      TestAudioDeviceModule::CreateDiscardRenderer(
+          /*sampling_frequency_in_hz=*/48000, /*num_channels=*/2);
+
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               /*capturer=*/nullptr, std::move(renderer));
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+
+  bool available;
+  EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
+  EXPECT_TRUE(available);
+}
+
+TEST(TestAudioDeviceTest, PlayoutIsNotAvailableWhenRendererIsNotSet) {
+  GlobalSimulatedTimeController time_controller(kStartTime);
+  TestAudioDevice audio_device(time_controller.GetTaskQueueFactory(),
+                               /*capturer=*/nullptr,
+                               /*renderer=*/nullptr);
+  ASSERT_EQ(audio_device.Init(), AudioDeviceGeneric::InitStatus::OK);
+
+  bool available;
+  EXPECT_EQ(audio_device.PlayoutIsAvailable(available), 0);
+  EXPECT_FALSE(available);
+}
+
+}  // namespace
+}  // namespace webrtc