Adding audio support to the new webmediaplyer

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright (c) 2012 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_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 
 #include "base/basictypes.h"
 #include "base/compiler_specific.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop_proxy.h"
 #include "base/time.h"
 #include "content/common/content_export.h"
+#include "content/renderer/media/webrtc_audio_renderer.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_renderer_sink.h"
 #include "third_party/webrtc/modules/audio_device/include/audio_device.h"
 
 // A WebRtcAudioDeviceImpl instance implements the abstract interface
 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc::
 // VoiceEngine) to register this class as an external AudioDeviceModule (ADM).
 // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the
 // session id that tells which device to use. The user can either get the
 // session id from the MediaStream or use a value of 1 (AudioInputDeviceManager
@@ skipping 170 matching lines @@
 //  - AGC is only supported in combination with the WASAPI-based audio layer
 //    on Windows, i.e., it is not supported on Windows XP.
 //  - All volume levels required for the AGC scheme are transfered in a
 //    normalized range [0.0, 1.0]. Scaling takes place in both endpoints
 //    (WebRTC client a media layer). This approach ensures that we can avoid
 //    transferring maximum levels between the renderer and the browser.
 //
 
 namespace content {
 
+// TODO(xians): Move this interface to webrtc so that libjingle can own a
+// reference to the renderer object.
+class WebRtcAudioRendererSource {
+ public:
+  // Callback to get the rendered interleaved data.
+  virtual void RenderData(uint8* audio_data,
+                          int number_of_channels,
+                          int number_of_frames,
+                          int audio_delay_milliseconds) = 0;
+
+ // Set the format for the capture audio parameters.
+ virtual void SetRenderFormat(const media::AudioParameters& params) = 0;

[scherkus (not reviewing), 2012/10/25 17:47:39]

fix indent

[no longer working on chromium, 2012/10/26 09:35:29]

Done.

+
+ protected:
+   virtual ~WebRtcAudioRendererSource() {}
+ };
+
+class WebRtcAudioRenderer;
+
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule),
-      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureCallback),
-      NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureEventHandler) {
+      NON_EXPORTED_BASE(public media::AudioInputDevice::CaptureEventHandler),
+      public WebRtcAudioRendererSource {
  public:
   // Methods called on main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
-  // media::AudioRendererSink::RenderCallback implementation.
-  virtual int Render(media::AudioBus* audio_bus,
-                     int audio_delay_milliseconds) OVERRIDE;
-  virtual void OnRenderError() OVERRIDE;
+  // WebRtcAudioRendererSource implementation.
+  virtual void RenderData(uint8* audio_data,
+                          int number_of_channels,
+                          int number_of_frames,
+                          int audio_delay_milliseconds) OVERRIDE;
+  virtual void SetRenderFormat(const media::AudioParameters& params) OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(media::AudioBus* audio_bus,
                        int audio_delay_milliseconds,
                        double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // AudioInputDevice::CaptureEventHandler implementation.
   virtual void OnDeviceStarted(const std::string& device_id) OVERRIDE;
   virtual void OnDeviceStopped() OVERRIDE;
@@ skipping 117 matching lines @@
   virtual int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) OVERRIDE;
   virtual int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const OVERRIDE;
 
   virtual int32_t ResetAudioDevice() OVERRIDE;
   virtual int32_t SetLoudspeakerStatus(bool enable) OVERRIDE;
   virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
   // Sets the session id.
   void SetSessionId(int session_id);
 
+  // Sets the |renderer_|, returns false if |renderer_| has already existed.
+  bool SetRenderer(WebRtcAudioRenderer* renderer);
+
   // Accessors.
   int input_buffer_size() const {
     return input_audio_parameters_.frames_per_buffer();
   }
   int output_buffer_size() const {
     return output_audio_parameters_.frames_per_buffer();
   }
   int input_channels() const {
     return input_audio_parameters_.channels();
   }
@@ skipping 21 matching lines @@
 
   int ref_count_;
 
   // Gives access to the message loop of the render thread on which this
   // object is created.
   scoped_refptr<base::MessageLoopProxy> render_loop_;
 
   // Provides access to the native audio input layer in the browser process.
   scoped_refptr<media::AudioInputDevice> audio_input_device_;
 
-  // Provides access to the native audio output layer in the browser process.
-  scoped_refptr<media::AudioRendererSink> audio_output_device_;
+  // Provides access to the audio renderer in the browser process.
+  scoped_refptr<WebRtcAudioRenderer> renderer_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
   // RegisterAudioCallback().
   webrtc::AudioTransport* audio_transport_callback_;
 
   // Cached values of utilized audio parameters. Platform dependent.
   media::AudioParameters input_audio_parameters_;
   media::AudioParameters output_audio_parameters_;
 
   // Cached value of the current audio delay on the input/capture side.
   int input_delay_ms_;
 
   // Cached value of the current audio delay on the output/renderer side.
   int output_delay_ms_;
 
   // Buffers used for temporary storage during capture/render callbacks.
   // Allocated during initialization to save stack.
   scoped_array<int16> input_buffer_;
-  scoped_array<int16> output_buffer_;
 
   webrtc::AudioDeviceModule::ErrorCode last_error_;
 
   base::TimeTicks last_process_time_;
 
   // Id of the media session to be started, it tells which device to be used
   // on the input/capture side.
   int session_id_;
 
-  // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|.
+  // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|, |renderer_|.
   mutable base::Lock lock_;
 
   int bytes_per_sample_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
   // Local copy of the current Automatic Gain Control state.
   bool agc_is_enabled_;
 
   // Used for histograms of total recording and playout times.
   base::Time start_capture_time_;
   base::Time start_render_time_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_