Add WebrtcAudioModule

remoting/protocol/webrtc_audio_module.h

Index: remoting/protocol/webrtc_audio_module.h
diff --git a/content/renderer/media/webrtc_audio_device_not_impl.h b/remoting/protocol/webrtc_audio_module.h
similarity index 58%
copy from content/renderer/media/webrtc_audio_device_not_impl.h
copy to remoting/protocol/webrtc_audio_module.h
index e9a98f0e71050265470f32009c8afa2ee33a1a50..e341c46346053312c32bffe5f6733478acb49a7c 100644
--- a/content/renderer/media/webrtc_audio_device_not_impl.h
+++ b/remoting/protocol/webrtc_audio_module.h
@@ -1,46 +1,50 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
+// Copyright 2016 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_NOT_IMPL_H_
-#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_NOT_IMPL_H_
+#ifndef REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_
+#define REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_
 
-#include <stdint.h>
-
-#include "base/compiler_specific.h"
-#include "base/macros.h"
-#include "base/time/time.h"
-#include "content/common/content_export.h"
+#include "base/memory/ref_counted.h"
+#include "base/synchronization/lock.h"
+#include "base/timer/timer.h"
 #include "third_party/webrtc/modules/audio_device/include/audio_device.h"
 
-namespace content {
+namespace base {
+class SingleThreadTaskRunner;
+}  // namespace base
+
+namespace remoting {
+namespace protocol {
+
+class AudioStub;
 
-// WebRtcAudioDeviceNotImpl contains default implementations of all methods
-// in the webrtc::AudioDeviceModule which are currently not supported in Chrome.
-// The real implementation is in WebRtcAudioDeviceImpl and it derives from
-// this class. The main purpose of breaking out non-implemented methods into
-// a separate unit is to make WebRtcAudioDeviceImpl more readable and easier
-// to maintain.
-class CONTENT_EXPORT WebRtcAudioDeviceNotImpl
-    : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule) {
+// Audio module passed to WebRTC. It doesn't access actual audio devices, but it
+// provides all functionality we need to ensure that audio streaming works
+// properly in WebRTC. Particularly it's responsible for calling AudioTransport
+// on regular intervals when playback is active. This ensures that all incoming
+// audio data is processed and passed to webrtc::AudioTrackSinkInterface
+// connected to the audio track.
+class WebrtcAudioModule : public webrtc::AudioDeviceModule {

[nicholss, 2016/10/04 21:42:00]

For this to tie into the current host/client, it seems like it needs to be an
audio_consumer or audio_provider? Or is this circumventing that connection in
the current client?

[Sergey Ulanov, 2016/10/04 22:32:32]

I'm going to add a separate adapters that use AudioTrackSinkInterface on host
and client to send and receive audio, but they won't have direct dependency on
this class.

  public:
-  WebRtcAudioDeviceNotImpl();
+  WebrtcAudioModule();
+  ~WebrtcAudioModule() override;
 
-  // webrtc::Module implementation.
-  // TODO(henrika): it is possible to add functionality in these methods.
-  // Only adding very basic support for now without triggering any callback
-  // in the webrtc::AudioDeviceObserver interface.
+  void Initialize(
+      scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner);
+
+  // webrtc::AudioDeviceModule implementation.
   int64_t TimeUntilNextProcess() override;
   void Process() override;
-
-  // Methods in webrtc::AudioDeviceModule which are not yet implemented.
-  // The idea is that we can move methods from this class to the real
-  // implementation in WebRtcAudioDeviceImpl when needed.
-
+  int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
+  ErrorCode LastError() const override;
   int32_t RegisterEventObserver(
       webrtc::AudioDeviceObserver* event_callback) override;
-  int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
-  webrtc::AudioDeviceModule::ErrorCode LastError() const override;
+  int32_t RegisterAudioCallback(
+      webrtc::AudioTransport* audio_callback) override;
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
   int16_t PlayoutDevices() override;
   int16_t RecordingDevices() override;
   int32_t PlayoutDeviceName(uint16_t index,
@@ -53,8 +57,20 @@ class CONTENT_EXPORT WebRtcAudioDeviceNotImpl
   int32_t SetPlayoutDevice(WindowsDeviceType device) override;
   int32_t SetRecordingDevice(uint16_t index) override;
   int32_t SetRecordingDevice(WindowsDeviceType device) override;
+  int32_t PlayoutIsAvailable(bool* available) override;
   int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
   int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+  int32_t SetAGC(bool enable) override;
+  bool AGC() const override;
   int32_t SetWaveOutVolume(uint16_t volume_left,
                            uint16_t volume_right) override;
   int32_t WaveOutVolume(uint16_t* volume_left,
@@ -70,6 +86,10 @@ class CONTENT_EXPORT WebRtcAudioDeviceNotImpl
   int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
   int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
   int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
   int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
   int32_t SpeakerMuteIsAvailable(bool* available) override;
   int32_t SetSpeakerMute(bool enable) override;
@@ -80,14 +100,18 @@ class CONTENT_EXPORT WebRtcAudioDeviceNotImpl
   int32_t MicrophoneBoostIsAvailable(bool* available) override;
   int32_t SetMicrophoneBoost(bool enable) override;
   int32_t MicrophoneBoost(bool* enabled) const override;
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
   int32_t SetStereoPlayout(bool enable) override;
   int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
   int32_t SetStereoRecording(bool enable) override;
   int32_t StereoRecording(bool* enabled) const override;
   int32_t SetRecordingChannel(const ChannelType channel) override;
   int32_t RecordingChannel(ChannelType* channel) const override;
   int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms) override;
   int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
+  int32_t PlayoutDelay(uint16_t* delay_ms) const override;
+  int32_t RecordingDelay(uint16_t* delay_ms) const override;
   int32_t CPULoad(uint16_t* load) const override;
   int32_t StartRawOutputFileRecording(
       const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
@@ -96,31 +120,47 @@ class CONTENT_EXPORT WebRtcAudioDeviceNotImpl
       const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
   int32_t StopRawInputFileRecording() override;
   int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
+  int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
   int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
+  int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
   int32_t ResetAudioDevice() override;
   int32_t SetLoudspeakerStatus(bool enable) override;
   int32_t GetLoudspeakerStatus(bool* enabled) const override;
-  int32_t SetAGC(bool enable) override;
-  bool AGC() const override;
   bool BuiltInAECIsAvailable() const override;
   bool BuiltInAGCIsAvailable() const override;
   bool BuiltInNSIsAvailable() const override;
   int32_t EnableBuiltInAEC(bool enable) override;
   int32_t EnableBuiltInAGC(bool enable) override;
   int32_t EnableBuiltInNS(bool enable) override;
-#if defined(OS_IOS)
+
+// Only supported on iOS.
+#if defined(WEBRTC_IOS)
   int GetPlayoutAudioParameters(AudioParameters* params) const override;
   int GetRecordAudioParameters(AudioParameters* params) const override;
-#endif  // OS_IOS
-
- protected:
-  ~WebRtcAudioDeviceNotImpl() override{};
+#endif  // WEBRTC_IOS
 
  private:
-  base::TimeTicks last_process_time_;
-  DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceNotImpl);
+  void StartPlayoutOnAudioThread();
+  void StopPlayoutOnAudioThread();
+
+  void PollFromSource();
+
+  scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner_;
+
+  // |lock_| must be locked when accessing |initialized_|, |playing_| and
+  // |audio_transport_|.
+  mutable base::Lock lock_;
+
+  bool initialized_ = false;
+  bool playing_ = false;
+  webrtc::AudioTransport* audio_transport_ = nullptr;
+
+  // Timer running on the |audio_task_runner_| that polls audio from
+  // |audio_transport_|.
+  base::RepeatingTimer poll_timer_;
 };
 
-}  // namespace content
+}  // namespace protocol
+}  // namespace remoting
 
-#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_NOT_IMPL_H_
+#endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_