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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 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 | 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/webrtc_audio_device_impl.h" | 5 #include "content/renderer/media/webrtc_audio_device_impl.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/metrics/histogram.h" | 8 #include "base/metrics/histogram.h" |
9 #include "base/strings/string_util.h" | 9 #include "base/strings/string_util.h" |
10 #include "base/win/windows_version.h" | 10 #include "base/win/windows_version.h" |
11 #include "content/renderer/media/webrtc_audio_capturer.h" | 11 #include "content/renderer/media/webrtc_audio_capturer.h" |
12 #include "content/renderer/media/webrtc_audio_renderer.h" | 12 #include "content/renderer/media/webrtc_audio_renderer.h" |
13 #include "content/renderer/render_thread_impl.h" | 13 #include "content/renderer/render_thread_impl.h" |
14 #include "media/audio/audio_parameters.h" | 14 #include "media/audio/audio_parameters.h" |
15 #include "media/audio/sample_rates.h" | 15 #include "media/audio/sample_rates.h" |
16 | 16 |
17 using media::AudioParameters; | 17 using media::AudioParameters; |
18 using media::ChannelLayout; | 18 using media::ChannelLayout; |
19 | 19 |
20 namespace content { | 20 namespace content { |
21 | 21 |
22 WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl() | 22 WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl() |
23 : ref_count_(0), | 23 : ref_count_(0), |
24 audio_transport_callback_(NULL), | 24 audio_transport_callback_(NULL), |
25 input_delay_ms_(0), | 25 input_delay_ms_(0), |
26 output_delay_ms_(0), | 26 output_delay_ms_(0), |
27 initialized_(false), | 27 initialized_(false), |
28 playing_(false), | 28 playing_(false), |
29 recording_(false), | 29 recording_(false), |
30 microphone_volume_(0) { | 30 microphone_volume_(0), |
31 render_buffer_size_(0) { | |
31 DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()"; | 32 DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()"; |
32 } | 33 } |
33 | 34 |
34 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { | 35 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() { |
35 DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()"; | 36 DVLOG(1) << "WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl()"; |
36 DCHECK(thread_checker_.CalledOnValidThread()); | 37 DCHECK(thread_checker_.CalledOnValidThread()); |
37 Terminate(); | 38 Terminate(); |
38 } | 39 } |
39 | 40 |
40 int32_t WebRtcAudioDeviceImpl::AddRef() { | 41 int32_t WebRtcAudioDeviceImpl::AddRef() { |
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114 } | 115 } |
115 | 116 |
116 return new_volume; | 117 return new_volume; |
117 } | 118 } |
118 | 119 |
119 void WebRtcAudioDeviceImpl::OnSetFormat( | 120 void WebRtcAudioDeviceImpl::OnSetFormat( |
120 const media::AudioParameters& params) { | 121 const media::AudioParameters& params) { |
121 DVLOG(1) << "WebRtcAudioDeviceImpl::OnSetFormat()"; | 122 DVLOG(1) << "WebRtcAudioDeviceImpl::OnSetFormat()"; |
122 } | 123 } |
123 | 124 |
124 void WebRtcAudioDeviceImpl::RenderData(uint8* audio_data, | 125 void WebRtcAudioDeviceImpl::RenderData(media::AudioBus* audio_bus, |
125 int number_of_channels, | 126 int sample_rate, |
126 int number_of_frames, | |
127 int audio_delay_milliseconds) { | 127 int audio_delay_milliseconds) { |
128 DCHECK_LE(number_of_frames, output_buffer_size()); | 128 if (!render_buffer_ || |
miu
2014/01/24 22:27:19
You could simplify a lot here if render_buffer_ we
| |
129 render_buffer_size_ != (audio_bus->frames() * audio_bus->channels())) { | |
130 render_buffer_size_ = audio_bus->frames() * audio_bus->channels(); | |
131 render_buffer_.reset(new int16[render_buffer_size_]); | |
132 } | |
133 | |
129 { | 134 { |
130 base::AutoLock auto_lock(lock_); | 135 base::AutoLock auto_lock(lock_); |
131 DCHECK(audio_transport_callback_); | 136 DCHECK(audio_transport_callback_); |
132 // Store the reported audio delay locally. | 137 // Store the reported audio delay locally. |
133 output_delay_ms_ = audio_delay_milliseconds; | 138 output_delay_ms_ = audio_delay_milliseconds; |
134 } | 139 } |
135 | 140 |
136 const int channels = number_of_channels; | 141 int samples_per_10_msec = (sample_rate / 100); |
137 DCHECK_LE(channels, output_channels()); | 142 int bytes_per_sample = 2; |
miu
2014/01/24 22:27:19
Replace 2 with sizeof(int16).
no longer working on chromium
2014/01/27 17:09:33
Done with changing it to sizeof(render_buffer_[0])
| |
138 | |
139 int samples_per_sec = output_sample_rate(); | |
140 int samples_per_10_msec = (samples_per_sec / 100); | |
141 int bytes_per_sample = output_audio_parameters_.bits_per_sample() / 8; | |
142 const int bytes_per_10_msec = | 143 const int bytes_per_10_msec = |
143 channels * samples_per_10_msec * bytes_per_sample; | 144 audio_bus->channels() * samples_per_10_msec * bytes_per_sample; |
144 | 145 DCHECK_EQ(audio_bus->frames() % samples_per_10_msec, 0); |
145 uint32_t num_audio_samples = 0; | |
146 int accumulated_audio_samples = 0; | |
147 | 146 |
148 // Get audio samples in blocks of 10 milliseconds from the registered | 147 // Get audio samples in blocks of 10 milliseconds from the registered |
149 // webrtc::AudioTransport source. Keep reading until our internal buffer | 148 // webrtc::AudioTransport source. Keep reading until our internal buffer |
150 // is full. | 149 // is full. |
151 while (accumulated_audio_samples < number_of_frames) { | 150 uint32_t num_audio_samples = 0; |
151 int accumulated_audio_samples = 0; | |
152 int16* audio_data = render_buffer_.get(); | |
153 while (accumulated_audio_samples < audio_bus->frames()) { | |
152 // Get 10ms and append output to temporary byte buffer. | 154 // Get 10ms and append output to temporary byte buffer. |
153 audio_transport_callback_->NeedMorePlayData(samples_per_10_msec, | 155 audio_transport_callback_->NeedMorePlayData(samples_per_10_msec, |
154 bytes_per_sample, | 156 bytes_per_sample, |
155 channels, | 157 audio_bus->channels(), |
156 samples_per_sec, | 158 sample_rate, |
157 audio_data, | 159 audio_data, |
158 num_audio_samples); | 160 num_audio_samples); |
159 accumulated_audio_samples += num_audio_samples; | 161 accumulated_audio_samples += num_audio_samples; |
160 audio_data += bytes_per_10_msec; | 162 audio_data += bytes_per_10_msec; |
161 } | 163 } |
162 } | |
163 | 164 |
164 void WebRtcAudioDeviceImpl::SetRenderFormat(const AudioParameters& params) { | 165 // De-interleave each channel and convert to 32-bit floating-point |
165 DCHECK(thread_checker_.CalledOnValidThread()); | 166 // with nominal range -1.0 -> +1.0 to match the callback format. |
166 output_audio_parameters_ = params; | 167 audio_bus->FromInterleaved(render_buffer_.get(), |
168 audio_bus->frames(), | |
169 bytes_per_sample); | |
170 | |
171 // Pass the render data to the observers. | |
172 base::AutoLock auto_lock(lock_); | |
173 for (RenderDataObservers::const_iterator it = render_data_observers_.begin(); | |
174 it != render_data_observers_.end(); ++it) { | |
175 (*it)->RenderData(audio_bus, sample_rate, audio_delay_milliseconds); | |
176 } | |
167 } | 177 } |
168 | 178 |
169 void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) { | 179 void WebRtcAudioDeviceImpl::RemoveAudioRenderer(WebRtcAudioRenderer* renderer) { |
170 DCHECK(thread_checker_.CalledOnValidThread()); | 180 DCHECK(thread_checker_.CalledOnValidThread()); |
171 DCHECK_EQ(renderer, renderer_); | 181 DCHECK_EQ(renderer, renderer_); |
172 base::AutoLock auto_lock(lock_); | 182 base::AutoLock auto_lock(lock_); |
173 renderer_ = NULL; | 183 renderer_ = NULL; |
174 playing_ = false; | 184 playing_ = false; |
175 } | 185 } |
176 | 186 |
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362 return 0; | 372 return 0; |
363 } | 373 } |
364 | 374 |
365 int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume(uint32_t* min_volume) const { | 375 int32_t WebRtcAudioDeviceImpl::MinMicrophoneVolume(uint32_t* min_volume) const { |
366 *min_volume = 0; | 376 *min_volume = 0; |
367 return 0; | 377 return 0; |
368 } | 378 } |
369 | 379 |
370 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const { | 380 int32_t WebRtcAudioDeviceImpl::StereoPlayoutIsAvailable(bool* available) const { |
371 DCHECK(initialized_); | 381 DCHECK(initialized_); |
372 *available = (output_channels() == 2); | 382 *available = renderer_ ? (renderer_->channels() == 2) : false; |
miu
2014/01/24 22:27:19
For my own curiosity: What if renderer_->channels(
no longer working on chromium
2014/01/27 17:09:33
The playout of the webrtc only supports mono and s
| |
373 return 0; | 383 return 0; |
374 } | 384 } |
375 | 385 |
376 int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable( | 386 int32_t WebRtcAudioDeviceImpl::StereoRecordingIsAvailable( |
377 bool* available) const { | 387 bool* available) const { |
378 DCHECK(initialized_); | 388 DCHECK(initialized_); |
379 // TODO(xians): These kind of hardware methods do not make much sense since we | 389 // TODO(xians): These kind of hardware methods do not make much sense since we |
380 // support multiple sources. Remove or figure out new APIs for such methods. | 390 // support multiple sources. Remove or figure out new APIs for such methods. |
381 scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer()); | 391 scoped_refptr<WebRtcAudioCapturer> capturer(GetDefaultCapturer()); |
382 if (!capturer.get()) | 392 if (!capturer.get()) |
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405 if (!capturer.get()) | 415 if (!capturer.get()) |
406 return -1; | 416 return -1; |
407 | 417 |
408 *samples_per_sec = static_cast<uint32_t>( | 418 *samples_per_sec = static_cast<uint32_t>( |
409 capturer->source_audio_parameters().sample_rate()); | 419 capturer->source_audio_parameters().sample_rate()); |
410 return 0; | 420 return 0; |
411 } | 421 } |
412 | 422 |
413 int32_t WebRtcAudioDeviceImpl::PlayoutSampleRate( | 423 int32_t WebRtcAudioDeviceImpl::PlayoutSampleRate( |
414 uint32_t* samples_per_sec) const { | 424 uint32_t* samples_per_sec) const { |
415 *samples_per_sec = static_cast<uint32_t>(output_sample_rate()); | 425 *samples_per_sec = renderer_ ? renderer_->sample_rate() : 0; |
416 return 0; | 426 return 0; |
417 } | 427 } |
418 | 428 |
419 bool WebRtcAudioDeviceImpl::SetAudioRenderer(WebRtcAudioRenderer* renderer) { | 429 bool WebRtcAudioDeviceImpl::SetAudioRenderer(WebRtcAudioRenderer* renderer) { |
420 DCHECK(thread_checker_.CalledOnValidThread()); | 430 DCHECK(thread_checker_.CalledOnValidThread()); |
421 DCHECK(renderer); | 431 DCHECK(renderer); |
422 | 432 |
423 base::AutoLock auto_lock(lock_); | 433 base::AutoLock auto_lock(lock_); |
424 if (renderer_.get()) | 434 if (renderer_.get()) |
425 return false; | 435 return false; |
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458 // Use the last |capturer| which is from the latest getUserMedia call as | 468 // Use the last |capturer| which is from the latest getUserMedia call as |
459 // the default capture device. | 469 // the default capture device. |
460 for (CapturerList::const_reverse_iterator iter = capturers_.rbegin(); | 470 for (CapturerList::const_reverse_iterator iter = capturers_.rbegin(); |
461 iter != capturers_.rend(); ++iter) { | 471 iter != capturers_.rend(); ++iter) { |
462 return *iter; | 472 return *iter; |
463 } | 473 } |
464 | 474 |
465 return NULL; | 475 return NULL; |
466 } | 476 } |
467 | 477 |
478 void WebRtcAudioDeviceImpl::AddRenderDataObserver( | |
479 WebRtcAudioRendererSource* observer) { | |
480 DCHECK(observer); | |
481 base::AutoLock auto_lock(lock_); | |
482 DCHECK(std::find(render_data_observers_.begin(), | |
483 render_data_observers_.end(), | |
484 observer) == render_data_observers_.end()); | |
485 render_data_observers_.push_back(observer); | |
486 } | |
487 | |
488 void WebRtcAudioDeviceImpl::RemoveRenderDataObserver( | |
489 WebRtcAudioRendererSource* observer) { | |
490 DCHECK(observer); | |
491 base::AutoLock auto_lock(lock_); | |
492 render_data_observers_.remove(observer); | |
493 } | |
494 | |
468 bool WebRtcAudioDeviceImpl::GetAuthorizedDeviceInfoForAudioRenderer( | 495 bool WebRtcAudioDeviceImpl::GetAuthorizedDeviceInfoForAudioRenderer( |
469 int* session_id, | 496 int* session_id, |
470 int* output_sample_rate, | 497 int* output_sample_rate, |
471 int* output_frames_per_buffer) { | 498 int* output_frames_per_buffer) { |
472 DCHECK(thread_checker_.CalledOnValidThread()); | 499 DCHECK(thread_checker_.CalledOnValidThread()); |
473 // If there is no capturer or there are more than one open capture devices, | 500 // If there is no capturer or there are more than one open capture devices, |
474 // return false. | 501 // return false. |
475 if (capturers_.empty() || capturers_.size() > 1) | 502 if (capturers_.empty() || capturers_.size() > 1) |
476 return false; | 503 return false; |
477 | 504 |
478 return GetDefaultCapturer()->GetPairedOutputParameters( | 505 return GetDefaultCapturer()->GetPairedOutputParameters( |
479 session_id, output_sample_rate, output_frames_per_buffer); | 506 session_id, output_sample_rate, output_frames_per_buffer); |
480 } | 507 } |
481 | 508 |
482 } // namespace content | 509 } // namespace content |
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