Repo created

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

@@ -0,0 +1,216 @@
+/*
+ *  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

@@ -0,0 +1,141 @@
+/*
+ *  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_