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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/media_stream_audio_processor.h" | 5 #include "content/renderer/media/media_stream_audio_processor.h" |
| 6 | 6 |
| 7 #include "base/command_line.h" | 7 #include "base/command_line.h" |
| 8 #include "base/debug/trace_event.h" | 8 #include "base/debug/trace_event.h" |
| 9 #include "content/public/common/content_switches.h" | 9 #include "content/public/common/content_switches.h" |
| 10 #include "content/renderer/media/media_stream_audio_processor_options.h" | 10 #include "content/renderer/media/media_stream_audio_processor_options.h" |
| 11 #include "content/renderer/media/rtc_media_constraints.h" | 11 #include "content/renderer/media/rtc_media_constraints.h" |
| 12 #include "media/audio/audio_parameters.h" | 12 #include "media/audio/audio_parameters.h" |
| 13 #include "media/base/audio_converter.h" | 13 #include "media/base/audio_converter.h" |
| 14 #include "media/base/audio_fifo.h" | 14 #include "media/base/audio_fifo.h" |
| 15 #include "media/base/channel_layout.h" | 15 #include "media/base/channel_layout.h" |
| 16 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 16 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
| 17 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface
.h" | 17 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface
.h" |
| 18 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" |
| 18 | 19 |
| 19 namespace content { | 20 namespace content { |
| 20 | 21 |
| 21 namespace { | 22 namespace { |
| 22 | 23 |
| 23 using webrtc::AudioProcessing; | 24 using webrtc::AudioProcessing; |
| 24 using webrtc::MediaConstraintsInterface; | 25 using webrtc::MediaConstraintsInterface; |
| 25 | 26 |
| 26 #if defined(OS_ANDROID) | 27 #if defined(OS_ANDROID) |
| 27 const int kAudioProcessingSampleRate = 16000; | 28 const int kAudioProcessingSampleRate = 16000; |
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| 136 media::AudioConverter audio_converter_; | 137 media::AudioConverter audio_converter_; |
| 137 scoped_ptr<media::AudioBus> audio_wrapper_; | 138 scoped_ptr<media::AudioBus> audio_wrapper_; |
| 138 scoped_ptr<media::AudioFifo> fifo_; | 139 scoped_ptr<media::AudioFifo> fifo_; |
| 139 }; | 140 }; |
| 140 | 141 |
| 141 MediaStreamAudioProcessor::MediaStreamAudioProcessor( | 142 MediaStreamAudioProcessor::MediaStreamAudioProcessor( |
| 142 const media::AudioParameters& source_params, | 143 const media::AudioParameters& source_params, |
| 143 const blink::WebMediaConstraints& constraints, | 144 const blink::WebMediaConstraints& constraints, |
| 144 int effects) | 145 int effects) |
| 145 : render_delay_ms_(0), | 146 : render_delay_ms_(0), |
| 146 audio_mirroring_(false) { | 147 audio_mirroring_(false), |
| 148 typing_detected_(false) { |
| 147 capture_thread_checker_.DetachFromThread(); | 149 capture_thread_checker_.DetachFromThread(); |
| 148 render_thread_checker_.DetachFromThread(); | 150 render_thread_checker_.DetachFromThread(); |
| 149 InitializeAudioProcessingModule(constraints, effects); | 151 InitializeAudioProcessingModule(constraints, effects); |
| 150 InitializeCaptureConverter(source_params); | 152 InitializeCaptureConverter(source_params); |
| 151 } | 153 } |
| 152 | 154 |
| 153 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() { | 155 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() { |
| 154 DCHECK(main_thread_checker_.CalledOnValidThread()); | 156 DCHECK(main_thread_checker_.CalledOnValidThread()); |
| 155 StopAudioProcessing(); | 157 StopAudioProcessing(); |
| 156 } | 158 } |
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| 257 const bool enable_ns = GetPropertyFromConstraints( | 259 const bool enable_ns = GetPropertyFromConstraints( |
| 258 &native_constraints, MediaConstraintsInterface::kNoiseSuppression); | 260 &native_constraints, MediaConstraintsInterface::kNoiseSuppression); |
| 259 const bool enable_high_pass_filter = GetPropertyFromConstraints( | 261 const bool enable_high_pass_filter = GetPropertyFromConstraints( |
| 260 &native_constraints, MediaConstraintsInterface::kHighpassFilter); | 262 &native_constraints, MediaConstraintsInterface::kHighpassFilter); |
| 261 | 263 |
| 262 audio_mirroring_ = GetPropertyFromConstraints( | 264 audio_mirroring_ = GetPropertyFromConstraints( |
| 263 &native_constraints, webrtc::MediaConstraintsInterface::kAudioMirroring); | 265 &native_constraints, webrtc::MediaConstraintsInterface::kAudioMirroring); |
| 264 | 266 |
| 265 // Return immediately if no audio processing component is enabled. | 267 // Return immediately if no audio processing component is enabled. |
| 266 if (!enable_aec && !enable_experimental_aec && !enable_ns && | 268 if (!enable_aec && !enable_experimental_aec && !enable_ns && |
| 267 !enable_high_pass_filter && !enable_typing_detection && !enable_agc) { | 269 !enable_high_pass_filter && !enable_typing_detection && !enable_agc && |
| 270 !audio_mirroring_) { |
| 268 return; | 271 return; |
| 269 } | 272 } |
| 270 | 273 |
| 271 // Create and configure the webrtc::AudioProcessing. | 274 // Create and configure the webrtc::AudioProcessing. |
| 272 audio_processing_.reset(webrtc::AudioProcessing::Create(0)); | 275 audio_processing_.reset(webrtc::AudioProcessing::Create(0)); |
| 273 | 276 |
| 274 // Enable the audio processing components. | 277 // Enable the audio processing components. |
| 275 if (enable_aec) { | 278 if (enable_aec) { |
| 276 EnableEchoCancellation(audio_processing_.get()); | 279 EnableEchoCancellation(audio_processing_.get()); |
| 277 if (enable_experimental_aec) | 280 if (enable_experimental_aec) |
| 278 EnableExperimentalEchoCancellation(audio_processing_.get()); | 281 EnableExperimentalEchoCancellation(audio_processing_.get()); |
| 279 } | 282 } |
| 280 | 283 |
| 281 if (enable_ns) | 284 if (enable_ns) |
| 282 EnableNoiseSuppression(audio_processing_.get()); | 285 EnableNoiseSuppression(audio_processing_.get()); |
| 283 | 286 |
| 284 if (enable_high_pass_filter) | 287 if (enable_high_pass_filter) |
| 285 EnableHighPassFilter(audio_processing_.get()); | 288 EnableHighPassFilter(audio_processing_.get()); |
| 286 | 289 |
| 287 if (enable_typing_detection) | 290 if (enable_typing_detection) { |
| 288 EnableTypingDetection(audio_processing_.get()); | 291 // TODO(xians): Remove this |typing_detector_| after the typing suppression |
| 292 // is enabled by default. |
| 293 typing_detector_.reset(new webrtc::TypingDetection()); |
| 294 EnableTypingDetection(audio_processing_.get(), typing_detector_.get()); |
| 295 } |
| 289 | 296 |
| 290 if (enable_agc) | 297 if (enable_agc) |
| 291 EnableAutomaticGainControl(audio_processing_.get()); | 298 EnableAutomaticGainControl(audio_processing_.get()); |
| 292 | 299 |
| 293 // Configure the audio format the audio processing is running on. This | 300 // Configure the audio format the audio processing is running on. This |
| 294 // has to be done after all the needed components are enabled. | 301 // has to be done after all the needed components are enabled. |
| 295 CHECK_EQ(audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate), | 302 CHECK_EQ(audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate), |
| 296 0); | 303 0); |
| 297 CHECK_EQ(audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel, | 304 CHECK_EQ(audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel, |
| 298 kAudioProcessingNumberOfChannel), | 305 kAudioProcessingNumberOfChannel), |
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| 391 DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err; | 398 DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err; |
| 392 err = audio_processing_->ProcessStream(audio_frame); | 399 err = audio_processing_->ProcessStream(audio_frame); |
| 393 DCHECK_EQ(err, 0) << "ProcessStream() error: " << err; | 400 DCHECK_EQ(err, 0) << "ProcessStream() error: " << err; |
| 394 | 401 |
| 395 // TODO(xians): Add support for typing detection, audio level calculation. | 402 // TODO(xians): Add support for typing detection, audio level calculation. |
| 396 | 403 |
| 397 if (audio_mirroring_ && audio_frame->num_channels_ == 2) { | 404 if (audio_mirroring_ && audio_frame->num_channels_ == 2) { |
| 398 // TODO(xians): Swap the stereo channels after switching to media::AudioBus. | 405 // TODO(xians): Swap the stereo channels after switching to media::AudioBus. |
| 399 } | 406 } |
| 400 | 407 |
| 408 if (typing_detector_ && |
| 409 audio_frame->vad_activity_ != webrtc::AudioFrame::kVadUnknown) { |
| 410 bool vad_active = |
| 411 (audio_frame->vad_activity_ == webrtc::AudioFrame::kVadActive); |
| 412 // TODO(xians): Pass this |typing_detected_| to peer connection. |
| 413 typing_detected_ = typing_detector_->Process(key_pressed, vad_active); |
| 414 } |
| 415 |
| 401 // Return 0 if the volume has not been changed, otherwise return the new | 416 // Return 0 if the volume has not been changed, otherwise return the new |
| 402 // volume. | 417 // volume. |
| 403 return (agc->stream_analog_level() == volume) ? | 418 return (agc->stream_analog_level() == volume) ? |
| 404 0 : agc->stream_analog_level(); | 419 0 : agc->stream_analog_level(); |
| 405 } | 420 } |
| 406 | 421 |
| 407 void MediaStreamAudioProcessor::StopAudioProcessing() { | 422 void MediaStreamAudioProcessor::StopAudioProcessing() { |
| 408 if (!audio_processing_.get()) | 423 if (!audio_processing_.get()) |
| 409 return; | 424 return; |
| 410 | 425 |
| 411 audio_processing_.reset(); | 426 audio_processing_.reset(); |
| 412 } | 427 } |
| 413 | 428 |
| 414 } // namespace content | 429 } // namespace content |
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