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 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
6
7 #include "base/logging.h"
8 #include "content/renderer/render_thread_impl.h"
9 #include "media/audio/audio_parameters.h"
10 #include "media/base/audio_fifo.h"
11 #include "media/base/audio_hardware_config.h"
12 #include "third_party/WebKit/public/platform/WebAudioSourceProviderClient.h"
13
14 using blink::WebVector;
15
16 namespace content {
17
18 static const size_t kMaxNumberOfBuffers = 10;
19
20 // Size of the buffer that WebAudio processes each time, it is the same value
21 // as AudioNode::ProcessingSizeInFrames in WebKit.
22 // static
23 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
24
WebRtcLocalAudioSourceProvider(const blink::WebMediaStreamTrack & track)25 WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
26 const blink::WebMediaStreamTrack& track)
27 : is_enabled_(false),
28 track_(track),
29 track_stopped_(false) {
30 // Get the native audio output hardware sample-rate for the sink.
31 // We need to check if RenderThreadImpl is valid here since the unittests
32 // do not have one and they will inject their own |sink_params_| for testing.
33 if (RenderThreadImpl::current()) {
34 media::AudioHardwareConfig* hardware_config =
35 RenderThreadImpl::current()->GetAudioHardwareConfig();
36 int sample_rate = hardware_config->GetOutputSampleRate();
37 sink_params_.Reset(
38 media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
39 media::CHANNEL_LAYOUT_STEREO, 2, sample_rate, 16,
40 kWebAudioRenderBufferSize);
41 }
42
43 // Connect the source provider to the track as a sink.
44 MediaStreamAudioSink::AddToAudioTrack(this, track_);
45 }
46
~WebRtcLocalAudioSourceProvider()47 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
48 if (audio_converter_.get())
49 audio_converter_->RemoveInput(this);
50
51 // If the track is still active, it is necessary to notify the track before
52 // the source provider goes away.
53 if (!track_stopped_)
54 MediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
55 }
56
OnSetFormat(const media::AudioParameters & params)57 void WebRtcLocalAudioSourceProvider::OnSetFormat(
58 const media::AudioParameters& params) {
59 // We need detach the thread here because it will be a new capture thread
60 // calling OnSetFormat() and OnData() if the source is restarted.
61 capture_thread_checker_.DetachFromThread();
62 DCHECK(capture_thread_checker_.CalledOnValidThread());
63 DCHECK(params.IsValid());
64 DCHECK(sink_params_.IsValid());
65
66 base::AutoLock auto_lock(lock_);
67 source_params_ = params;
68 // Create the audio converter with |disable_fifo| as false so that the
69 // converter will request source_params.frames_per_buffer() each time.
70 // This will not increase the complexity as there is only one client to
71 // the converter.
72 audio_converter_.reset(
73 new media::AudioConverter(params, sink_params_, false));
74 audio_converter_->AddInput(this);
75 fifo_.reset(new media::AudioFifo(
76 params.channels(),
77 kMaxNumberOfBuffers * params.frames_per_buffer()));
78 input_bus_ = media::AudioBus::Create(params.channels(),
79 params.frames_per_buffer());
80 }
81
OnReadyStateChanged(blink::WebMediaStreamSource::ReadyState state)82 void WebRtcLocalAudioSourceProvider::OnReadyStateChanged(
83 blink::WebMediaStreamSource::ReadyState state) {
84 if (state == blink::WebMediaStreamSource::ReadyStateEnded)
85 track_stopped_ = true;
86 }
87
OnData(const int16 * audio_data,int sample_rate,int number_of_channels,int number_of_frames)88 void WebRtcLocalAudioSourceProvider::OnData(
89 const int16* audio_data,
90 int sample_rate,
91 int number_of_channels,
92 int number_of_frames) {
93 DCHECK(capture_thread_checker_.CalledOnValidThread());
94 base::AutoLock auto_lock(lock_);
95 if (!is_enabled_)
96 return;
97
98 DCHECK(fifo_.get());
99
100 // TODO(xians): A better way to handle the interleaved and deinterleaved
101 // format switching, see issue/317710.
102 DCHECK(input_bus_->frames() == number_of_frames);
103 DCHECK(input_bus_->channels() == number_of_channels);
104 input_bus_->FromInterleaved(audio_data, number_of_frames, 2);
105
106 if (fifo_->frames() + number_of_frames <= fifo_->max_frames()) {
107 fifo_->Push(input_bus_.get());
108 } else {
109 // This can happen if the data in FIFO is too slowly consumed or
110 // WebAudio stops consuming data.
111 DVLOG(3) << "Local source provicer FIFO is full" << fifo_->frames();
112 }
113 }
114
setClient(blink::WebAudioSourceProviderClient * client)115 void WebRtcLocalAudioSourceProvider::setClient(
116 blink::WebAudioSourceProviderClient* client) {
117 NOTREACHED();
118 }
119
provideInput(const WebVector<float * > & audio_data,size_t number_of_frames)120 void WebRtcLocalAudioSourceProvider::provideInput(
121 const WebVector<float*>& audio_data, size_t number_of_frames) {
122 DCHECK_EQ(number_of_frames, kWebAudioRenderBufferSize);
123 if (!output_wrapper_ ||
124 static_cast<size_t>(output_wrapper_->channels()) != audio_data.size()) {
125 output_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
126 }
127
128 output_wrapper_->set_frames(number_of_frames);
129 for (size_t i = 0; i < audio_data.size(); ++i)
130 output_wrapper_->SetChannelData(i, audio_data[i]);
131
132 base::AutoLock auto_lock(lock_);
133 if (!audio_converter_)
134 return;
135
136 is_enabled_ = true;
137 audio_converter_->Convert(output_wrapper_.get());
138 }
139
ProvideInput(media::AudioBus * audio_bus,base::TimeDelta buffer_delay)140 double WebRtcLocalAudioSourceProvider::ProvideInput(
141 media::AudioBus* audio_bus, base::TimeDelta buffer_delay) {
142 if (fifo_->frames() >= audio_bus->frames()) {
143 fifo_->Consume(audio_bus, 0, audio_bus->frames());
144 } else {
145 audio_bus->Zero();
146 DVLOG(1) << "WARNING: Underrun, FIFO has data " << fifo_->frames()
147 << " samples but " << audio_bus->frames()
148 << " samples are needed";
149 }
150
151 return 1.0;
152 }
153
SetSinkParamsForTesting(const media::AudioParameters & sink_params)154 void WebRtcLocalAudioSourceProvider::SetSinkParamsForTesting(
155 const media::AudioParameters& sink_params) {
156 sink_params_ = sink_params;
157 }
158
159 } // namespace content
160