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/WebVector.h"
  19
  20 namespace media {
  21 class AudioBus;
  22 class AudioConverter;
  23 class AudioFifo;
  24 class AudioParameters;
  25 }
  26
  27 namespace blink {
  28 class WebAudioSourceProviderClient;
  29 }
  30
  31 namespace content {
  32
  33 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
  34 //     WebRtcAudioCapturer ---> blink::WebAudioSourceProvider
  35 //
  36 // WebRtcLocalAudioSourceProvider works as a sink to the WebRtcAudiocapturer
  37 // and store the capture data to a FIFO. When the media stream is connected to
  38 // WebAudio as a source provider, WebAudio will periodically call
  39 // provideInput() to get the data from the FIFO.
  40 //
  41 // All calls are protected by a lock.
  42 class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
  43     :  NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
  44        NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
  45        NON_EXPORTED_BASE(public MediaStreamAudioSink) {
  46  public:
  47   static const size_t kWebAudioRenderBufferSize;
  48
  49   WebRtcLocalAudioSourceProvider();
  50   virtual ~WebRtcLocalAudioSourceProvider();
  51
  52   // MediaStreamAudioSink implementation.
  53   virtual void OnData(const int16* audio_data,
  54                       int sample_rate,
  55                       int number_of_channels,
  56                       int number_of_frames) OVERRIDE;
  57   virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
  58
  59   // blink::WebAudioSourceProvider implementation.
  60   virtual void setClient(blink::WebAudioSourceProviderClient* client) OVERRIDE;
  61   virtual void provideInput(const blink::WebVector<float*>& audio_data,
  62                             size_t number_of_frames) OVERRIDE;
  63
  64   // media::AudioConverter::Inputcallback implementation.
  65   // This function is triggered by provideInput()on the WebAudio audio thread,
  66   // so it has been under the protection of |lock_|.
  67   virtual double ProvideInput(media::AudioBus* audio_bus,
  68                               base::TimeDelta buffer_delay) OVERRIDE;
  69
  70   // Method to allow the unittests to inject its own sink parameters to avoid
  71   // query the hardware.
  72   // TODO(xians,tommi): Remove and instead offer a way to inject the sink
  73   // parameters so that the implementation doesn't rely on the global default
  74   // hardware config but instead gets the parameters directly from the sink
  75   // (WebAudio in this case). Ideally the unit test should be able to use that
  76   // same mechanism to inject the sink parameters for testing.
  77   void SetSinkParamsForTesting(const media::AudioParameters& sink_params);
  78
  79  private:
  80   // Used to DCHECK that some methods are called on the capture audio thread.
  81   base::ThreadChecker capture_thread_checker_;
  82
  83   scoped_ptr<media::AudioConverter> audio_converter_;
  84   scoped_ptr<media::AudioFifo> fifo_;
  85   scoped_ptr<media::AudioBus> input_bus_;
  86   scoped_ptr<media::AudioBus> output_wrapper_;
  87   bool is_enabled_;
  88   media::AudioParameters source_params_;
  89   media::AudioParameters sink_params_;
  90
  91   // Protects all the member variables above.
  92   base::Lock lock_;
  93
  94   // Used to report the correct delay to |webaudio_source_|.
  95   base::TimeTicks last_fill_;
  96
  97   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
  98 };
  99
 100 }  // namespace content
 101
 102 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_