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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 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 | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/webaudio_capturer_source.h" | 5 #include "content/renderer/media/webaudio_capturer_source.h" |
6 | 6 |
7 #include "base/bind.h" | |
8 #include "base/bind_helpers.h" | |
7 #include "base/logging.h" | 9 #include "base/logging.h" |
8 #include "base/time/time.h" | 10 #include "base/time/time.h" |
9 #include "content/renderer/media/webrtc_local_audio_track.h" | 11 #include "content/renderer/media/webrtc_local_audio_track.h" |
10 | 12 |
11 using media::AudioBus; | 13 using media::AudioBus; |
12 using media::AudioFifo; | |
13 using media::AudioParameters; | 14 using media::AudioParameters; |
14 using media::ChannelLayout; | 15 using media::ChannelLayout; |
15 using media::CHANNEL_LAYOUT_MONO; | 16 using media::CHANNEL_LAYOUT_MONO; |
16 using media::CHANNEL_LAYOUT_STEREO; | 17 using media::CHANNEL_LAYOUT_STEREO; |
17 | 18 |
18 static const int kMaxNumberOfBuffersInFifo = 5; | |
19 | |
20 namespace content { | 19 namespace content { |
21 | 20 |
22 WebAudioCapturerSource::WebAudioCapturerSource( | 21 WebAudioCapturerSource::WebAudioCapturerSource( |
23 const blink::WebMediaStreamSource& blink_source) | 22 const blink::WebMediaStreamSource& blink_source) |
24 : track_(NULL), | 23 : track_(NULL), |
25 audio_format_changed_(false), | 24 audio_format_changed_(false), |
26 blink_source_(blink_source) { | 25 rechunker_(base::TimeDelta::FromMilliseconds(10), |
27 } | 26 base::Bind(&WebAudioCapturerSource::DeliverRechunkedAudio, |
27 base::Unretained(this))), | |
28 blink_source_(blink_source) {} | |
28 | 29 |
29 WebAudioCapturerSource::~WebAudioCapturerSource() { | 30 WebAudioCapturerSource::~WebAudioCapturerSource() { |
30 DCHECK(thread_checker_.CalledOnValidThread()); | 31 DCHECK(thread_checker_.CalledOnValidThread()); |
31 removeFromBlinkSource(); | 32 removeFromBlinkSource(); |
32 } | 33 } |
33 | 34 |
34 void WebAudioCapturerSource::setFormat( | 35 void WebAudioCapturerSource::setFormat( |
35 size_t number_of_channels, float sample_rate) { | 36 size_t number_of_channels, float sample_rate) { |
36 DCHECK(thread_checker_.CalledOnValidThread()); | 37 DCHECK(thread_checker_.CalledOnValidThread()); |
37 DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate=" | 38 DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate=" |
38 << sample_rate << ")"; | 39 << sample_rate << ")"; |
39 | 40 |
40 // If the channel count is greater than 8, use discrete layout. However, | 41 // If the channel count is greater than 8, use discrete layout. However, |
41 // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline. | 42 // anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline. |
42 ChannelLayout channel_layout = | 43 ChannelLayout channel_layout = |
43 number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE | 44 number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE |
44 : media::GuessChannelLayout(number_of_channels); | 45 : media::GuessChannelLayout(number_of_channels); |
45 | 46 |
46 base::AutoLock auto_lock(lock_); | 47 base::AutoLock auto_lock(lock_); |
47 | 48 |
48 // Set the format used by this WebAudioCapturerSource. We are using 10ms data | 49 // Set the format used by this WebAudioCapturerSource. We are using 10ms data |
49 // as buffer size since that is the native buffer size of WebRtc packet | 50 // as buffer size since that is the native buffer size of WebRtc packet |
50 // running on. | 51 // running on. |
52 rechunker_.SetSampleRate(sample_rate); | |
51 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, | 53 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout, |
52 sample_rate, 16, sample_rate / 100); | 54 sample_rate, 16, rechunker_.output_frames()); |
53 | 55 |
54 // Take care of the discrete channel layout case. | 56 // Take care of the discrete channel layout case. |
55 params_.set_channels_for_discrete(number_of_channels); | 57 params_.set_channels_for_discrete(number_of_channels); |
56 | 58 |
57 audio_format_changed_ = true; | 59 audio_format_changed_ = true; |
58 | 60 |
59 wrapper_bus_ = AudioBus::CreateWrapper(params_.channels()); | 61 if (!wrapper_bus_ || |
60 capture_bus_ = AudioBus::Create(params_); | 62 wrapper_bus_->channels() != static_cast<int>(number_of_channels)) { |
61 | 63 wrapper_bus_ = AudioBus::CreateWrapper(params_.channels()); |
62 fifo_.reset(new AudioFifo( | 64 } |
63 params_.channels(), | |
64 kMaxNumberOfBuffersInFifo * params_.frames_per_buffer())); | |
65 } | 65 } |
66 | 66 |
67 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) { | 67 void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) { |
68 DCHECK(thread_checker_.CalledOnValidThread()); | 68 DCHECK(thread_checker_.CalledOnValidThread()); |
69 DCHECK(track); | 69 DCHECK(track); |
70 base::AutoLock auto_lock(lock_); | 70 base::AutoLock auto_lock(lock_); |
71 track_ = track; | 71 track_ = track; |
72 } | 72 } |
73 | 73 |
74 void WebAudioCapturerSource::Stop() { | 74 void WebAudioCapturerSource::Stop() { |
75 DCHECK(thread_checker_.CalledOnValidThread()); | 75 DCHECK(thread_checker_.CalledOnValidThread()); |
76 { | 76 { |
77 base::AutoLock auto_lock(lock_); | 77 base::AutoLock auto_lock(lock_); |
78 track_ = NULL; | 78 track_ = NULL; |
79 } | 79 } |
80 // removeFromBlinkSource() should not be called while |lock_| is acquired, | 80 // removeFromBlinkSource() should not be called while |lock_| is acquired, |
81 // as it could result in a deadlock. | 81 // as it could result in a deadlock. |
82 removeFromBlinkSource(); | 82 removeFromBlinkSource(); |
83 } | 83 } |
84 | 84 |
85 void WebAudioCapturerSource::consumeAudio( | 85 void WebAudioCapturerSource::consumeAudio( |
86 const blink::WebVector<const float*>& audio_data, | 86 const blink::WebVector<const float*>& audio_data, |
87 size_t number_of_frames) { | 87 size_t number_of_frames) { |
88 // TODO(miu): Plumbing is needed to determine the actual capture timestamp | |
89 // of the audio, instead of just snapshotting TimeTicks::Now(), for proper | |
90 // audio/video sync. http://crbug.com/335335 | |
91 base::TimeTicks reference_time = base::TimeTicks::Now(); | |
92 | |
88 base::AutoLock auto_lock(lock_); | 93 base::AutoLock auto_lock(lock_); |
89 if (!track_) | 94 if (!track_) |
90 return; | 95 return; |
91 | 96 |
92 // Update the downstream client if the audio format has been changed. | 97 // Update the downstream client if the audio format has been changed. |
93 if (audio_format_changed_) { | 98 if (audio_format_changed_) { |
94 track_->OnSetFormat(params_); | 99 track_->OnSetFormat(params_); |
95 audio_format_changed_ = false; | 100 audio_format_changed_ = false; |
96 } | 101 } |
97 | 102 |
98 wrapper_bus_->set_frames(number_of_frames); | 103 wrapper_bus_->set_frames(number_of_frames); |
99 | |
100 // Make sure WebKit is honoring what it told us up front | |
101 // about the channels. | |
102 DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size())); | 104 DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size())); |
103 | |
104 for (size_t i = 0; i < audio_data.size(); ++i) | 105 for (size_t i = 0; i < audio_data.size(); ++i) |
105 wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i])); | 106 wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i])); |
106 | 107 |
107 // Handle mismatch between WebAudio buffer-size and WebRTC. | 108 // The following will result in zero, one, or multiple synchronous calls to |
108 int available = fifo_->max_frames() - fifo_->frames(); | 109 // DeliverRechunkedAudio(). |
109 if (available < static_cast<int>(number_of_frames)) { | 110 rechunker_.Push(*wrapper_bus_, reference_time - base::TimeTicks()); |
o1ka
2016/02/22 13:04:47
I like how neat it becomes!
miu
2016/02/23 04:27:41
Acknowledged.
| |
110 NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun."; | 111 } |
111 return; | |
112 } | |
113 | 112 |
114 // Compute the estimated capture time of the first sample frame of audio that | 113 void WebAudioCapturerSource::DeliverRechunkedAudio( |
115 // will be consumed from the FIFO in the loop below. | 114 const media::AudioBus& audio_bus, |
116 base::TimeTicks estimated_capture_time = base::TimeTicks::Now() - | 115 base::TimeDelta reference_timestamp) { |
117 fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate(); | 116 lock_.AssertAcquired(); |
118 | 117 track_->Capture(audio_bus, base::TimeTicks() + reference_timestamp, false); |
119 fifo_->Push(wrapper_bus_.get()); | |
120 while (fifo_->frames() >= capture_bus_->frames()) { | |
121 fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames()); | |
122 track_->Capture(*capture_bus_, estimated_capture_time, false); | |
123 | |
124 // Advance the estimated capture time for the next FIFO consume operation. | |
125 estimated_capture_time += | |
126 capture_bus_->frames() * base::TimeDelta::FromSeconds(1) / | |
127 params_.sample_rate(); | |
128 } | |
129 } | 118 } |
130 | 119 |
131 // If registered as audio consumer in |blink_source_|, deregister from | 120 // If registered as audio consumer in |blink_source_|, deregister from |
132 // |blink_source_| and stop keeping a reference to |blink_source_|. | 121 // |blink_source_| and stop keeping a reference to |blink_source_|. |
133 // Failure to call this method when stopping the track might leave an invalid | 122 // Failure to call this method when stopping the track might leave an invalid |
134 // WebAudioCapturerSource reference still registered as an audio consumer on | 123 // WebAudioCapturerSource reference still registered as an audio consumer on |
135 // the blink side. | 124 // the blink side. |
136 void WebAudioCapturerSource::removeFromBlinkSource() { | 125 void WebAudioCapturerSource::removeFromBlinkSource() { |
137 if (!blink_source_.isNull()) { | 126 if (!blink_source_.isNull()) { |
138 blink_source_.removeAudioConsumer(this); | 127 blink_source_.removeAudioConsumer(this); |
139 blink_source_.reset(); | 128 blink_source_.reset(); |
140 } | 129 } |
141 } | 130 } |
142 | 131 |
143 } // namespace content | 132 } // namespace content |
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