1 // Copyright (c) 2012 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/webaudio_capturer_source.h"
6
7 #include "base/logging.h"
8 #include "base/time/time.h"
9 #include "content/renderer/media/webrtc_audio_capturer.h"
10 #include "content/renderer/media/webrtc_local_audio_track.h"
11
12 using media::AudioBus;
13 using media::AudioFifo;
14 using media::AudioParameters;
15 using media::ChannelLayout;
16 using media::CHANNEL_LAYOUT_MONO;
17 using media::CHANNEL_LAYOUT_STEREO;
18
19 static const int kMaxNumberOfBuffersInFifo = 5;
20
21 namespace content {
22
WebAudioCapturerSource()23 WebAudioCapturerSource::WebAudioCapturerSource()
24 : track_(NULL),
25 capturer_(NULL),
26 audio_format_changed_(false) {
27 }
28
~WebAudioCapturerSource()29 WebAudioCapturerSource::~WebAudioCapturerSource() {
30 }
31
setFormat(size_t number_of_channels,float sample_rate)32 void WebAudioCapturerSource::setFormat(
33 size_t number_of_channels, float sample_rate) {
34 DCHECK(thread_checker_.CalledOnValidThread());
35 DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
36 << sample_rate << ")";
37 if (number_of_channels > 2) {
38 // TODO(xians): Handle more than just the mono and stereo cases.
39 LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format.";
40 return;
41 }
42
43 ChannelLayout channel_layout =
44 number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
45
46 base::AutoLock auto_lock(lock_);
47 // Set the format used by this WebAudioCapturerSource. We are using 10ms data
48 // as buffer size since that is the native buffer size of WebRtc packet
49 // running on.
50 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
51 channel_layout, number_of_channels, 0, sample_rate, 16,
52 sample_rate / 100);
53 audio_format_changed_ = true;
54
55 wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
56 capture_bus_ = AudioBus::Create(params_);
57 fifo_.reset(new AudioFifo(
58 params_.channels(),
59 kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
60 }
61
Start(WebRtcLocalAudioTrack * track,WebRtcAudioCapturer * capturer)62 void WebAudioCapturerSource::Start(
63 WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) {
64 DCHECK(thread_checker_.CalledOnValidThread());
65 DCHECK(track);
66 base::AutoLock auto_lock(lock_);
67 track_ = track;
68 capturer_ = capturer;
69 }
70
Stop()71 void WebAudioCapturerSource::Stop() {
72 DCHECK(thread_checker_.CalledOnValidThread());
73 base::AutoLock auto_lock(lock_);
74 track_ = NULL;
75 capturer_ = NULL;
76 }
77
consumeAudio(const blink::WebVector<const float * > & audio_data,size_t number_of_frames)78 void WebAudioCapturerSource::consumeAudio(
79 const blink::WebVector<const float*>& audio_data,
80 size_t number_of_frames) {
81 base::AutoLock auto_lock(lock_);
82 if (!track_)
83 return;
84
85 // Update the downstream client if the audio format has been changed.
86 if (audio_format_changed_) {
87 track_->OnSetFormat(params_);
88 audio_format_changed_ = false;
89 }
90
91 wrapper_bus_->set_frames(number_of_frames);
92
93 // Make sure WebKit is honoring what it told us up front
94 // about the channels.
95 DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
96
97 for (size_t i = 0; i < audio_data.size(); ++i)
98 wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
99
100 // Handle mismatch between WebAudio buffer-size and WebRTC.
101 int available = fifo_->max_frames() - fifo_->frames();
102 if (available < static_cast<int>(number_of_frames)) {
103 NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
104 return;
105 }
106
107 fifo_->Push(wrapper_bus_.get());
108 int capture_frames = params_.frames_per_buffer();
109 base::TimeDelta delay;
110 int volume = 0;
111 bool key_pressed = false;
112 if (capturer_) {
113 capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed);
114 }
115 while (fifo_->frames() >= capture_frames) {
116 fifo_->Consume(capture_bus_.get(), 0, capture_frames);
117 track_->Capture(capture_bus_.get(), delay.InMilliseconds(),
118 volume, key_pressed);
119 }
120 }
121
122 } // namespace content
123