content/renderer/media/webrtc_local_audio_source_provider.h

   1 // Copyright 2013 The Chromium Authors. All rights reserved.
   2 // Use of this source code is governed by a BSD-style license that can be
   3 // found in the LICENSE file.
   4
   5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
   6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
   7
   8 #include <vector>
   9
  10 #include "base/memory/scoped_ptr.h"
  11 #include "base/synchronization/lock.h"
  12 #include "base/threading/thread_checker.h"
  13 #include "base/time/time.h"
  14 #include "content/common/content_export.h"
  15 #include "content/public/renderer/media_stream_audio_sink.h"
  16 #include "media/base/audio_converter.h"
  17 #include "third_party/WebKit/public/platform/WebAudioSourceProvider.h"
  18 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
  19 #include "third_party/WebKit/public/platform/WebVector.h"
  20
  21 namespace media {
  22 class AudioBus;
  23 class AudioConverter;
  24 class AudioFifo;
  25 class AudioParameters;
  26 }
  27
  28 namespace blink {
  29 class WebAudioSourceProviderClient;
  30 }
  31
  32 namespace content {
  33
  34 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
  35 //     WebRtcLocalAudioTrack ---> blink::WebAudioSourceProvider
  36 //
  37 // WebRtcLocalAudioSourceProvider works as a sink to the WebRtcLocalAudioTrack
  38 // and store the capture data to a FIFO. When the media stream is connected to
  39 // WebAudio MediaStreamAudioSourceNode as a source provider,
  40 // MediaStreamAudioSourceNode will periodically call provideInput() to get the
  41 // data from the FIFO.
  42 //
  43 // All calls are protected by a lock.
  44 class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
  45     :  NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
  46        NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
  47        NON_EXPORTED_BASE(public MediaStreamAudioSink) {
  48  public:
  49   static const size_t kWebAudioRenderBufferSize;
  50
  51   explicit WebRtcLocalAudioSourceProvider(
  52       const blink::WebMediaStreamTrack& track);
  53   virtual ~WebRtcLocalAudioSourceProvider();
  54
  55   // MediaStreamAudioSink implementation.
  56   virtual void OnData(const int16* audio_data,
  57                       int sample_rate,
  58                       int number_of_channels,
  59                       int number_of_frames) OVERRIDE;
  60   virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
  61   virtual void OnReadyStateChanged(
  62       blink::WebMediaStreamSource::ReadyState state) OVERRIDE;
  63
  64   // blink::WebAudioSourceProvider implementation.
  65   virtual void setClient(blink::WebAudioSourceProviderClient* client) OVERRIDE;
  66   virtual void provideInput(const blink::WebVector<float*>& audio_data,
  67                             size_t number_of_frames) OVERRIDE;
  68
  69   // media::AudioConverter::Inputcallback implementation.
  70   // This function is triggered by provideInput()on the WebAudio audio thread,
  71   // so it has been under the protection of |lock_|.
  72   virtual double ProvideInput(media::AudioBus* audio_bus,
  73                               base::TimeDelta buffer_delay) OVERRIDE;
  74
  75   // Method to allow the unittests to inject its own sink parameters to avoid
  76   // query the hardware.
  77   // TODO(xians,tommi): Remove and instead offer a way to inject the sink
  78   // parameters so that the implementation doesn't rely on the global default
  79   // hardware config but instead gets the parameters directly from the sink
  80   // (WebAudio in this case). Ideally the unit test should be able to use that
  81   // same mechanism to inject the sink parameters for testing.
  82   void SetSinkParamsForTesting(const media::AudioParameters& sink_params);
  83
  84  private:
  85   // Used to DCHECK that some methods are called on the capture audio thread.
  86   base::ThreadChecker capture_thread_checker_;
  87
  88   scoped_ptr<media::AudioConverter> audio_converter_;
  89   scoped_ptr<media::AudioFifo> fifo_;
  90   scoped_ptr<media::AudioBus> input_bus_;
  91   scoped_ptr<media::AudioBus> output_wrapper_;
  92   bool is_enabled_;
  93   media::AudioParameters source_params_;
  94   media::AudioParameters sink_params_;
  95
  96   // Protects all the member variables above.
  97   base::Lock lock_;
  98
  99   // Used to report the correct delay to |webaudio_source_|.
 100   base::TimeTicks last_fill_;
 101
 102   // The audio track that this source provider is connected to.
 103   blink::WebMediaStreamTrack track_;
 104
 105   // Flag to tell if the track has been stopped or not.
 106   bool track_stopped_;
 107
 108   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
 109 };
 110
 111 }  // namespace content
 112
 113 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_