Changes for fake audio on headless platform

media/engine/webrtc_voice_engine.cc

@@ -33,6 +33,7 @@
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/async_audio_processing/async_audio_processing.h"
 #include "modules/audio_device/audio_device_impl.h"
+#include "modules/audio_device/include/fake_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "modules/audio_processing/include/audio_processing.h"
@@ -358,6 +359,9 @@ void WebRtcVoiceEngine::Init() {
         webrtc::AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory_);
   }
 #endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
+  if (!adm_) {
+    adm_ = new rtc::RefCountedObject<webrtc::FakeAudioDeviceModule>();
+  }
   RTC_CHECK(adm());
   webrtc::adm_helpers::Init(adm());
 

modules/audio_device/include/audio_device_default.h

@@ -11,7 +11,14 @@
 #ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
 #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
 
+#include "modules/audio_device/audio_device_buffer.h"
 #include "modules/audio_device/include/audio_device.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/critical_section.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/sleep.h"
 
 namespace webrtc {
 namespace webrtc_impl {
@@ -22,11 +29,15 @@ namespace webrtc_impl {
 template <typename T>
 class AudioDeviceModuleDefault : public T {
  public:
-  AudioDeviceModuleDefault() {}
+  AudioDeviceModuleDefault() {
+    task_queue_factory = webrtc::CreateDefaultTaskQueueFactory();
+    audio_device_buffer.reset(new AudioDeviceBuffer(task_queue_factory.get()));
+    playout_frames_in_10ms = 48000 / 100;
+  }
   virtual ~AudioDeviceModuleDefault() {}
 
   int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
-    return 0;
+    return audio_device_buffer->RegisterAudioCallback(audioCallback);
   }
   int32_t Init() override { return 0; }
   int32_t InitSpeaker() override { return 0; }
@@ -36,7 +47,17 @@ class AudioDeviceModuleDefault : public T {
     return 0;
   }
   int32_t SetStereoPlayout(bool enable) override { return 0; }
-  int32_t StopPlayout() override { return 0; }
+  int32_t StopPlayout() override {
+    {
+      rtc::CritScope lock(&_critSect);
+      playing = false;
+    }
+    if (play_thread.get()) {
+      play_thread->Stop();
+      play_thread.reset();
+    }
+    return 0;
+  }
   int32_t InitMicrophone() override { return 0; }
   int32_t SetRecordingDevice(uint16_t index) override { return 0; }
   int32_t SetRecordingDevice(
@@ -65,16 +86,35 @@ class AudioDeviceModuleDefault : public T {
                               char guid[kAdmMaxGuidSize]) override {
     return 0;
   }
-  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
-  int32_t InitPlayout() override { return 0; }
+  int32_t PlayoutIsAvailable(bool* available) override {
+    *available = true;
+    return 0;
+  }
+  int32_t InitPlayout() override {
+    rtc::CritScope lock(&_critSect);
+    if (audio_device_buffer.get()) {
+      audio_device_buffer->SetPlayoutSampleRate(48000);
+      audio_device_buffer->SetPlayoutChannels(2);
+    }
+    return 0;
+  }
   bool PlayoutIsInitialized() const override { return true; }
   int32_t RecordingIsAvailable(bool* available) override { return 0; }
   int32_t InitRecording() override { return 0; }
   bool RecordingIsInitialized() const override { return true; }
-  int32_t StartPlayout() override { return 0; }
-  bool Playing() const override { return false; }
+  int32_t StartPlayout() override {
+    if (playing)
+      return 0;
+
+    playing = true;
+    play_thread.reset(new rtc::PlatformThread(PlayThreadFunc,
+	this, "fake_audio_play_thread", rtc::kRealtimePriority));
+    play_thread->Start();
+    return 0;
+  }
+  bool Playing() const override { return playing; }
   int32_t StartRecording() override { return 0; }
-  bool Recording() const override { return false; }
+  bool Recording() const override { return true; }
   bool SpeakerIsInitialized() const override { return true; }
   bool MicrophoneIsInitialized() const override { return true; }
   int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
@@ -94,12 +134,12 @@ class AudioDeviceModuleDefault : public T {
   int32_t SetMicrophoneMute(bool enable) override { return 0; }
   int32_t MicrophoneMute(bool* enabled) const override { return 0; }
   int32_t StereoPlayoutIsAvailable(bool* available) const override {
-    *available = false;
+    *available = true;
     return 0;
   }
   int32_t StereoPlayout(bool* enabled) const override { return 0; }
   int32_t StereoRecordingIsAvailable(bool* available) const override {
-    *available = false;
+    *available = true;
     return 0;
   }
   int32_t StereoRecording(bool* enabled) const override { return 0; }
@@ -116,6 +156,40 @@ class AudioDeviceModuleDefault : public T {
 
   int32_t GetPlayoutUnderrunCount() const override { return -1; }
 
+  bool PlayThreadProcess() {
+    if (!playing)
+      return false;
+    int64_t current_time = rtc::TimeMillis();
+
+    _critSect.Enter();
+    if (last_call_millis == 0 ||
+	current_time - last_call_millis >= 10) {
+      _critSect.Leave();
+      audio_device_buffer->RequestPlayoutData(playout_frames_in_10ms);
+      _critSect.Enter();
+      last_call_millis = current_time;
+    }
+    _critSect.Leave();
+    int64_t delta_time = rtc::TimeMillis() - current_time;
+    if (delta_time < 10) {
+      SleepMs(10 - delta_time);
+    }
+    return true;
+  }
+
+  static void PlayThreadFunc(void* pThis) {
+    AudioDeviceModuleDefault* device = static_cast<AudioDeviceModuleDefault*>(pThis);
+    while (device->PlayThreadProcess()) {
+    }
+  }
+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer;
+  std::unique_ptr<rtc::PlatformThread> play_thread;
+  size_t playout_frames_in_10ms;
+  rtc::CriticalSection _critSect;
+  bool playing = false;
+  int64_t last_call_millis = 0;
+  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory;
+
 #if defined(WEBRTC_IOS)
   int GetPlayoutAudioParameters(AudioParameters* params) const override {
     return -1;

modules/audio_processing/audio_processing_impl.cc

@@ -1790,6 +1790,9 @@ void AudioProcessingImpl::InitializeHighPassFilter(bool forced_reset) {
 }
 
 void AudioProcessingImpl::InitializeEchoController() {
+#if defined(WEBRTC_LINUX)
+  return;
+#else
   bool use_echo_controller =
       echo_control_factory_ ||
       (config_.echo_canceller.enabled && !config_.echo_canceller.mobile_mode);
@@ -1866,6 +1869,7 @@ void AudioProcessingImpl::InitializeEchoController() {
 
   submodules_.echo_control_mobile.reset();
   aecm_render_signal_queue_.reset();
+#endif
 }
 
 void AudioProcessingImpl::InitializeGainController1() {