Repo created

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_
+
+#include <memory>
+
+#include "absl/base/nullability.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RTC_EXPORT AecDumpFactory {
+ public:
+  // The `worker_queue` must outlive the created AecDump instance.
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  // The AecDump takes responsibility for `handle` and closes it in the
+  // destructor. A non-null return value indicates that the file has been
+  // sucessfully opened.
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      FileWrapper file,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::string_view file_name,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::Nonnull<FILE*> handle,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_FACTORY_H_

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc

@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec_dump/aec_dump_impl.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/base/nullability.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+
+namespace webrtc {
+
+namespace {
+void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
+                           webrtc::audioproc::Config* pb_cfg) {
+  pb_cfg->set_aec_enabled(config.aec_enabled);
+  pb_cfg->set_aec_delay_agnostic_enabled(config.aec_delay_agnostic_enabled);
+  pb_cfg->set_aec_drift_compensation_enabled(
+      config.aec_drift_compensation_enabled);
+  pb_cfg->set_aec_extended_filter_enabled(config.aec_extended_filter_enabled);
+  pb_cfg->set_aec_suppression_level(config.aec_suppression_level);
+
+  pb_cfg->set_aecm_enabled(config.aecm_enabled);
+  pb_cfg->set_aecm_comfort_noise_enabled(config.aecm_comfort_noise_enabled);
+  pb_cfg->set_aecm_routing_mode(config.aecm_routing_mode);
+
+  pb_cfg->set_agc_enabled(config.agc_enabled);
+  pb_cfg->set_agc_mode(config.agc_mode);
+  pb_cfg->set_agc_limiter_enabled(config.agc_limiter_enabled);
+  pb_cfg->set_noise_robust_agc_enabled(config.noise_robust_agc_enabled);
+
+  pb_cfg->set_hpf_enabled(config.hpf_enabled);
+
+  pb_cfg->set_ns_enabled(config.ns_enabled);
+  pb_cfg->set_ns_level(config.ns_level);
+
+  pb_cfg->set_transient_suppression_enabled(
+      config.transient_suppression_enabled);
+
+  pb_cfg->set_pre_amplifier_enabled(config.pre_amplifier_enabled);
+  pb_cfg->set_pre_amplifier_fixed_gain_factor(
+      config.pre_amplifier_fixed_gain_factor);
+
+  pb_cfg->set_experiments_description(config.experiments_description);
+}
+
+}  // namespace
+
+AecDumpImpl::AecDumpImpl(FileWrapper debug_file,
+                         int64_t max_log_size_bytes,
+                         absl::Nonnull<TaskQueueBase*> worker_queue)
+    : debug_file_(std::move(debug_file)),
+      num_bytes_left_for_log_(max_log_size_bytes),
+      worker_queue_(worker_queue) {}
+
+AecDumpImpl::~AecDumpImpl() {
+  // Block until all tasks have finished running.
+  rtc::Event thread_sync_event;
+  worker_queue_->PostTask([&thread_sync_event] { thread_sync_event.Set(); });
+  // Wait until the event has been signaled with .Set(). By then all
+  // pending tasks will have finished.
+  thread_sync_event.Wait(rtc::Event::kForever);
+}
+
+void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format,
+                                   int64_t time_now_ms) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::INIT);
+  audioproc::Init* msg = event->mutable_init();
+
+  msg->set_sample_rate(api_format.input_stream().sample_rate_hz());
+  msg->set_output_sample_rate(api_format.output_stream().sample_rate_hz());
+  msg->set_reverse_sample_rate(
+      api_format.reverse_input_stream().sample_rate_hz());
+  msg->set_reverse_output_sample_rate(
+      api_format.reverse_output_stream().sample_rate_hz());
+
+  msg->set_num_input_channels(
+      static_cast<int32_t>(api_format.input_stream().num_channels()));
+  msg->set_num_output_channels(
+      static_cast<int32_t>(api_format.output_stream().num_channels()));
+  msg->set_num_reverse_channels(
+      static_cast<int32_t>(api_format.reverse_input_stream().num_channels()));
+  msg->set_num_reverse_output_channels(
+      api_format.reverse_output_stream().num_channels());
+  msg->set_timestamp_ms(time_now_ms);
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::AddCaptureStreamInput(
+    const AudioFrameView<const float>& src) {
+  capture_stream_info_.AddInput(src);
+}
+
+void AecDumpImpl::AddCaptureStreamOutput(
+    const AudioFrameView<const float>& src) {
+  capture_stream_info_.AddOutput(src);
+}
+
+void AecDumpImpl::AddCaptureStreamInput(const int16_t* const data,
+                                        int num_channels,
+                                        int samples_per_channel) {
+  capture_stream_info_.AddInput(data, num_channels, samples_per_channel);
+}
+
+void AecDumpImpl::AddCaptureStreamOutput(const int16_t* const data,
+                                         int num_channels,
+                                         int samples_per_channel) {
+  capture_stream_info_.AddOutput(data, num_channels, samples_per_channel);
+}
+
+void AecDumpImpl::AddAudioProcessingState(const AudioProcessingState& state) {
+  capture_stream_info_.AddAudioProcessingState(state);
+}
+
+void AecDumpImpl::WriteCaptureStreamMessage() {
+  PostWriteToFileTask(capture_stream_info_.FetchEvent());
+}
+
+void AecDumpImpl::WriteRenderStreamMessage(const int16_t* const data,
+                                           int num_channels,
+                                           int samples_per_channel) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::REVERSE_STREAM);
+  audioproc::ReverseStream* msg = event->mutable_reverse_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  msg->set_data(data, data_size);
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteRenderStreamMessage(
+    const AudioFrameView<const float>& src) {
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::REVERSE_STREAM);
+
+  audioproc::ReverseStream* msg = event->mutable_reverse_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size());
+  }
+
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::CONFIG);
+  CopyFromConfigToEvent(config, event->mutable_config());
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::WriteRuntimeSetting(
+    const AudioProcessing::RuntimeSetting& runtime_setting) {
+  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
+  auto event = std::make_unique<audioproc::Event>();
+  event->set_type(audioproc::Event::RUNTIME_SETTING);
+  audioproc::RuntimeSetting* setting = event->mutable_runtime_setting();
+  switch (runtime_setting.type()) {
+    case AudioProcessing::RuntimeSetting::Type::kCapturePreGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_pre_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCapturePostGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_post_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::
+        kCustomRenderProcessingRuntimeSetting: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_custom_render_processing_setting(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCaptureCompressionGain:
+      // Runtime AGC1 compression gain is ignored.
+      // TODO(http://bugs.webrtc.org/10432): Store compression gain in aecdumps.
+      break;
+    case AudioProcessing::RuntimeSetting::Type::kCaptureFixedPostGain: {
+      float x;
+      runtime_setting.GetFloat(&x);
+      setting->set_capture_fixed_post_gain(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kCaptureOutputUsed: {
+      bool x;
+      runtime_setting.GetBool(&x);
+      setting->set_capture_output_used(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kPlayoutVolumeChange: {
+      int x;
+      runtime_setting.GetInt(&x);
+      setting->set_playout_volume_change(x);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kPlayoutAudioDeviceChange: {
+      AudioProcessing::RuntimeSetting::PlayoutAudioDeviceInfo src;
+      runtime_setting.GetPlayoutAudioDeviceInfo(&src);
+      auto* dst = setting->mutable_playout_audio_device_change();
+      dst->set_id(src.id);
+      dst->set_max_volume(src.max_volume);
+      break;
+    }
+    case AudioProcessing::RuntimeSetting::Type::kNotSpecified:
+      RTC_DCHECK_NOTREACHED();
+      break;
+  }
+  PostWriteToFileTask(std::move(event));
+}
+
+void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) {
+  RTC_DCHECK(event);
+  worker_queue_->PostTask([event = std::move(event), this] {
+    std::string event_string = event->SerializeAsString();
+    const size_t event_byte_size = event_string.size();
+
+    if (num_bytes_left_for_log_ >= 0) {
+      const int64_t next_message_size = sizeof(int32_t) + event_byte_size;
+      if (num_bytes_left_for_log_ < next_message_size) {
+        // Ensure that no further events are written, even if they're smaller
+        // than the current event.
+        num_bytes_left_for_log_ = 0;
+        return;
+      }
+      num_bytes_left_for_log_ -= next_message_size;
+    }
+
+    // Write message preceded by its size.
+    if (!debug_file_.Write(&event_byte_size, sizeof(int32_t))) {
+      RTC_DCHECK_NOTREACHED();
+    }
+    if (!debug_file_.Write(event_string.data(), event_string.size())) {
+      RTC_DCHECK_NOTREACHED();
+    }
+  });
+}
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    FileWrapper file,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  RTC_DCHECK(worker_queue);
+  if (!file.is_open())
+    return nullptr;
+
+  return std::make_unique<AecDumpImpl>(std::move(file), max_log_size_bytes,
+                                       worker_queue);
+}
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::string_view file_name,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  return Create(FileWrapper::OpenWriteOnly(file_name), max_log_size_bytes,
+                worker_queue);
+}
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::Nonnull<FILE*> handle,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_processing/aec_dump/capture_stream_info.h"
+#include "modules/audio_processing/include/aec_dump.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "rtc_base/thread_annotations.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+
+// Task-queue based implementation of AecDump. It is thread safe by
+// relying on locks in TaskQueue.
+class AecDumpImpl : public AecDump {
+ public:
+  // `max_log_size_bytes` - maximum number of bytes to write to the debug file,
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  AecDumpImpl(FileWrapper debug_file,
+              int64_t max_log_size_bytes,
+              absl::Nonnull<TaskQueueBase*> worker_queue);
+  AecDumpImpl(const AecDumpImpl&) = delete;
+  AecDumpImpl& operator=(const AecDumpImpl&) = delete;
+  ~AecDumpImpl() override;
+
+  void WriteInitMessage(const ProcessingConfig& api_format,
+                        int64_t time_now_ms) override;
+  void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
+  void AddCaptureStreamInput(const int16_t* const data,
+                             int num_channels,
+                             int samples_per_channel) override;
+  void AddCaptureStreamOutput(const int16_t* const data,
+                              int num_channels,
+                              int samples_per_channel) override;
+  void AddAudioProcessingState(const AudioProcessingState& state) override;
+  void WriteCaptureStreamMessage() override;
+
+  void WriteRenderStreamMessage(const int16_t* const data,
+                                int num_channels,
+                                int samples_per_channel) override;
+  void WriteRenderStreamMessage(
+      const AudioFrameView<const float>& src) override;
+
+  void WriteConfig(const InternalAPMConfig& config) override;
+
+  void WriteRuntimeSetting(
+      const AudioProcessing::RuntimeSetting& runtime_setting) override;
+
+ private:
+  void PostWriteToFileTask(std::unique_ptr<audioproc::Event> event);
+
+  FileWrapper debug_file_;
+  int64_t num_bytes_left_for_log_ = 0;
+  rtc::RaceChecker race_checker_;
+  absl::Nonnull<TaskQueueBase*> worker_queue_;
+  CaptureStreamInfo capture_stream_info_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_AEC_DUMP_IMPL_H_

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_processing/aec_dump/capture_stream_info.h"
+
+namespace webrtc {
+
+void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
+  auto* stream = event_->mutable_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_input_channel(channel_view.begin(),
+                              sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
+  auto* stream = event_->mutable_stream();
+
+  for (int i = 0; i < src.num_channels(); ++i) {
+    const auto& channel_view = src.channel(i);
+    stream->add_output_channel(channel_view.begin(),
+                               sizeof(float) * channel_view.size());
+  }
+}
+
+void CaptureStreamInfo::AddInput(const int16_t* const data,
+                                 int num_channels,
+                                 int samples_per_channel) {
+  auto* stream = event_->mutable_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  stream->set_input_data(data, data_size);
+}
+
+void CaptureStreamInfo::AddOutput(const int16_t* const data,
+                                  int num_channels,
+                                  int samples_per_channel) {
+  auto* stream = event_->mutable_stream();
+  const size_t data_size = sizeof(int16_t) * samples_per_channel * num_channels;
+  stream->set_output_data(data, data_size);
+}
+
+void CaptureStreamInfo::AddAudioProcessingState(
+    const AecDump::AudioProcessingState& state) {
+  auto* stream = event_->mutable_stream();
+  stream->set_delay(state.delay);
+  stream->set_drift(state.drift);
+  if (state.applied_input_volume.has_value()) {
+    stream->set_applied_input_volume(*state.applied_input_volume);
+  }
+  stream->set_keypress(state.keypress);
+}
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h

@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_processing/include/aec_dump.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
+#else
+#include "modules/audio_processing/debug.pb.h"
+#endif
+
+namespace webrtc {
+
+class CaptureStreamInfo {
+ public:
+  CaptureStreamInfo() { CreateNewEvent(); }
+  CaptureStreamInfo(const CaptureStreamInfo&) = delete;
+  CaptureStreamInfo& operator=(const CaptureStreamInfo&) = delete;
+  ~CaptureStreamInfo() = default;
+
+  void AddInput(const AudioFrameView<const float>& src);
+  void AddOutput(const AudioFrameView<const float>& src);
+
+  void AddInput(const int16_t* const data,
+                int num_channels,
+                int samples_per_channel);
+  void AddOutput(const int16_t* const data,
+                 int num_channels,
+                 int samples_per_channel);
+
+  void AddAudioProcessingState(const AecDump::AudioProcessingState& state);
+
+  std::unique_ptr<audioproc::Event> FetchEvent() {
+    std::unique_ptr<audioproc::Event> result = std::move(event_);
+    CreateNewEvent();
+    return result;
+  }
+
+ private:
+  void CreateNewEvent() {
+    event_ = std::make_unique<audioproc::Event>();
+    event_->set_type(audioproc::Event::STREAM);
+  }
+  std::unique_ptr<audioproc::Event> event_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_CAPTURE_STREAM_INFO_H_

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc

@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
+
+namespace webrtc {
+
+namespace test {
+
+MockAecDump::MockAecDump() = default;
+MockAecDump::~MockAecDump() = default;
+}  // namespace test
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h

@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
+#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
+
+#include <memory>
+
+#include "modules/audio_processing/include/aec_dump.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+namespace test {
+
+class MockAecDump : public AecDump {
+ public:
+  MockAecDump();
+  virtual ~MockAecDump();
+
+  MOCK_METHOD(void,
+              WriteInitMessage,
+              (const ProcessingConfig& api_format, int64_t time_now_ms),
+              (override));
+
+  MOCK_METHOD(void,
+              AddCaptureStreamInput,
+              (const AudioFrameView<const float>& src),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamOutput,
+              (const AudioFrameView<const float>& src),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamInput,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              AddCaptureStreamOutput,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              AddAudioProcessingState,
+              (const AudioProcessingState& state),
+              (override));
+  MOCK_METHOD(void, WriteCaptureStreamMessage, (), (override));
+
+  MOCK_METHOD(void,
+              WriteRenderStreamMessage,
+              (const int16_t* const data,
+               int num_channels,
+               int samples_per_channel),
+              (override));
+  MOCK_METHOD(void,
+              WriteRenderStreamMessage,
+              (const AudioFrameView<const float>& src),
+              (override));
+
+  MOCK_METHOD(void, WriteConfig, (const InternalAPMConfig& config), (override));
+
+  MOCK_METHOD(void,
+              WriteRuntimeSetting,
+              (const AudioProcessing::RuntimeSetting& config),
+              (override));
+};
+
+}  // namespace test
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_

TMessagesProj/jni/voip/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/base/nullability.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/include/aec_dump.h"
+
+namespace webrtc {
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    FileWrapper file,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  return nullptr;
+}
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::string_view file_name,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  return nullptr;
+}
+
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::Nonnull<FILE*> handle,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
+  return nullptr;
+}
+}  // namespace webrtc