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
21namespace media {
22class AudioBus;
23class AudioConverter;
24class AudioFifo;
25class AudioParameters;
26}
27
28namespace blink {
29class WebAudioSourceProviderClient;
30}
31
32namespace 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.
44class 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_
114