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| 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/media_stream_audio_processor.h" | |
| 6 | |
| 7 #include "base/command_line.h" | |
| 8 #include "base/debug/trace_event.h" | |
| 9 #include "content/public/common/content_switches.h" | |
| 10 #include "content/renderer/media/media_stream_audio_processor_options.h" | |
| 11 #include "media/audio/audio_parameters.h" | |
| 12 #include "media/base/audio_converter.h" | |
| 13 #include "media/base/audio_fifo.h" | |
| 14 #include "media/base/channel_layout.h" | |
| 15 | |
| 16 namespace content { | |
| 17 | |
| 18 namespace { | |
| 19 | |
| 20 using webrtc::AudioProcessing; | |
| 21 using webrtc::MediaConstraintsInterface; | |
| 22 | |
| 23 #if defined(ANDROID) | |
| 24 const int kAudioProcessingSampleRate = 16000; | |
| 25 #else | |
| 26 const int kAudioProcessingSampleRate = 32000; | |
| 27 #endif | |
| 28 const int kAudioProcessingNumberOfChannel = 1; | |
| 29 | |
| 30 const int kMaxNumberOfBuffersInFifo = 2; | |
| 31 | |
| 32 } // namespace | |
| 33 | |
| 34 class MediaStreamAudioProcessor::MediaStreamAudioConverter | |
| 35 : public media::AudioConverter::InputCallback { | |
| 36 public: | |
| 37 MediaStreamAudioConverter(const media::AudioParameters& source_params, | |
| 38 const media::AudioParameters& sink_params) | |
| 39 : source_params_(source_params), | |
| 40 sink_params_(sink_params), | |
| 41 audio_converter_(source_params, sink_params_, false) { | |
| 42 audio_converter_.AddInput(this); | |
| 43 // Create and initialize audio fifo and audio bus wrapper. | |
| 44 // The size of the FIFO should be at least twice of the source buffer size | |
| 45 // or twice of the sink buffer size. | |
| 46 int buffer_size = std::max( | |
| 47 kMaxNumberOfBuffersInFifo * source_params_.frames_per_buffer(), | |
| 48 kMaxNumberOfBuffersInFifo * sink_params_.frames_per_buffer()); | |
| 49 fifo_.reset(new media::AudioFifo(source_params_.channels(), buffer_size)); | |
| 50 // TODO(xians): Use CreateWrapper to save one memcpy. | |
| 51 audio_wrapper_ = media::AudioBus::Create(sink_params_.channels(), | |
| 52 sink_params_.frames_per_buffer()); | |
| 53 } | |
| 54 | |
| 55 virtual ~MediaStreamAudioConverter() { | |
| 56 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 57 audio_converter_.RemoveInput(this); | |
| 58 } | |
| 59 | |
| 60 void Push(media::AudioBus* audio_source) { | |
| 61 // Called on the audio thread, which is the capture audio thread for | |
| 62 // |MediaStreamAudioProcessor::capture_converter_|, and render audio thread | |
| 63 // for |MediaStreamAudioProcessor::render_converter_|. | |
| 64 // And it must be the same thread as calling Convert(). | |
| 65 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 66 fifo_->Push(audio_source); | |
| 67 } | |
| 68 | |
| 69 bool Convert(webrtc::AudioFrame* out) { | |
| 70 // Called on the audio thread, which is the capture audio thread for | |
| 71 // |MediaStreamAudioProcessor::capture_converter_|, and render audio thread | |
| 72 // for |MediaStreamAudioProcessor::render_converter_|. | |
| 73 // Return false if there is no 10ms data in the FIFO. | |
| 74 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 75 if (fifo_->frames() < (source_params_.sample_rate() / 100)) | |
| 76 return false; | |
| 77 | |
| 78 // Convert 10ms data to the output format, this will trigger ProvideInput(). | |
| 79 audio_converter_.Convert(audio_wrapper_.get()); | |
| 80 | |
| 81 // TODO(xians): Figure out a better way to handle the interleaved and | |
| 82 // deinterleaved format switching. | |
| 83 audio_wrapper_->ToInterleaved(audio_wrapper_->frames(), | |
| 84 sink_params_.bits_per_sample() / 8, | |
| 85 out->data_); | |
| 86 | |
| 87 out->samples_per_channel_ = sink_params_.frames_per_buffer(); | |
| 88 out->sample_rate_hz_ = sink_params_.sample_rate(); | |
| 89 out->speech_type_ = webrtc::AudioFrame::kNormalSpeech; | |
| 90 out->vad_activity_ = webrtc::AudioFrame::kVadUnknown; | |
| 91 out->num_channels_ = sink_params_.channels(); | |
| 92 | |
| 93 return true; | |
| 94 } | |
| 95 | |
| 96 const media::AudioParameters& source_parameters() const { | |
| 97 return source_params_; | |
| 98 } | |
| 99 const media::AudioParameters& sink_parameters() const { | |
| 100 return sink_params_; | |
| 101 } | |
| 102 | |
| 103 private: | |
| 104 // AudioConverter::InputCallback implementation. | |
| 105 virtual double ProvideInput(media::AudioBus* audio_bus, | |
| 106 base::TimeDelta buffer_delay) OVERRIDE { | |
| 107 // Called on realtime audio thread. | |
| 108 // TODO(xians): Figure out why the first Convert() triggers ProvideInput | |
| 109 // two times. | |
| 110 if (fifo_->frames() < audio_bus->frames()) | |
| 111 return 0; | |
| 112 | |
| 113 fifo_->Consume(audio_bus, 0, audio_bus->frames()); | |
| 114 | |
| 115 // Return 1.0 to indicate no volume scaling on the data. | |
| 116 return 1.0; | |
| 117 } | |
| 118 | |
| 119 base::ThreadChecker thread_checker_; | |
| 120 const media::AudioParameters source_params_; | |
| 121 const media::AudioParameters sink_params_; | |
| 122 | |
| 123 // TODO(xians): consider using SincResampler to save some memcpy. | |
| 124 // Handles mixing and resampling between input and output parameters. | |
| 125 media::AudioConverter audio_converter_; | |
| 126 scoped_ptr<media::AudioBus> audio_wrapper_; | |
| 127 scoped_ptr<media::AudioFifo> fifo_; | |
| 128 }; | |
| 129 | |
| 130 MediaStreamAudioProcessor::MediaStreamAudioProcessor( | |
| 131 const webrtc::MediaConstraintsInterface* constraints) | |
| 132 : render_delay_ms_(0) { | |
| 133 capture_thread_checker_.DetachFromThread(); | |
| 134 render_thread_checker_.DetachFromThread(); | |
| 135 InitializeAudioProcessingModule(constraints); | |
| 136 } | |
| 137 | |
| 138 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() { | |
| 139 DCHECK(main_thread_checker_.CalledOnValidThread()); | |
| 140 StopAudioProcessing(); | |
| 141 } | |
| 142 | |
| 143 void MediaStreamAudioProcessor::PushCaptureData(media::AudioBus* audio_source) { | |
| 144 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 145 capture_converter_->Push(audio_source); | |
| 146 } | |
| 147 | |
| 148 void MediaStreamAudioProcessor::PushRenderData( | |
| 149 const int16* render_audio, int sample_rate, int number_of_channels, | |
| 150 int number_of_frames, base::TimeDelta render_delay) { | |
| 151 DCHECK(render_thread_checker_.CalledOnValidThread()); | |
| 152 | |
| 153 // Return immediately if the echo cancellation is off. | |
| 154 if (!audio_processing_ || | |
| 155 !audio_processing_->echo_cancellation()->is_enabled()) { | |
| 156 return; | |
| 157 } | |
| 158 | |
| 159 TRACE_EVENT0("audio", | |
| 160 "MediaStreamAudioProcessor::FeedRenderDataToAudioProcessing"); | |
| 161 int64 new_render_delay_ms = render_delay.InMilliseconds(); | |
| 162 DCHECK_LT(new_render_delay_ms, | |
| 163 std::numeric_limits<base::subtle::Atomic32>::max()); | |
| 164 base::subtle::Release_Store(&render_delay_ms_, new_render_delay_ms); | |
| 165 | |
| 166 InitializeRenderConverterIfNeeded(sample_rate, number_of_channels, | |
| 167 number_of_frames); | |
| 168 | |
| 169 // TODO(xians): Avoid this extra interleave/deinterleave. | |
| 170 render_data_bus_->FromInterleaved(render_audio, | |
| 171 render_data_bus_->frames(), | |
| 172 sizeof(render_audio[0])); | |
| 173 render_converter_->Push(render_data_bus_.get()); | |
| 174 while (render_converter_->Convert(&render_frame_)) | |
| 175 audio_processing_->AnalyzeReverseStream(&render_frame_); | |
| 176 } | |
| 177 | |
| 178 bool MediaStreamAudioProcessor::ProcessAndConsumeData( | |
| 179 base::TimeDelta capture_delay, int volume, bool key_pressed, | |
| 180 int16** out) { | |
| 181 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 182 TRACE_EVENT0("audio", | |
| 183 "MediaStreamAudioProcessor::ProcessAndConsumeData"); | |
| 184 | |
| 185 if (!capture_converter_->Convert(&capture_frame_)) | |
| 186 return false; | |
| 187 | |
| 188 ProcessData(&capture_frame_, capture_delay, volume, key_pressed); | |
| 189 *out = capture_frame_.data_; | |
| 190 | |
| 191 return true; | |
| 192 } | |
| 193 | |
| 194 void MediaStreamAudioProcessor::SetCaptureFormat( | |
| 195 const media::AudioParameters& source_params) { | |
| 196 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 197 DCHECK(source_params.IsValid()); | |
| 198 | |
| 199 // Create and initialize audio converter for the source data. | |
| 200 // When the webrtc AudioProcessing is enabled, the sink format of the | |
| 201 // converter will be the same as the post-processed data format, which is | |
| 202 // 32k mono for desktops and 16k mono for Android. When the AudioProcessing | |
| 203 // is disabled, the sink format will be the same as the source format. | |
| 204 const int sink_sample_rate = audio_processing_ ? | |
| 205 kAudioProcessingSampleRate : source_params.sample_rate(); | |
| 206 const media::ChannelLayout sink_channel_layout = audio_processing_ ? | |
| 207 media::CHANNEL_LAYOUT_MONO : source_params.channel_layout(); | |
| 208 | |
| 209 // WebRtc is using 10ms data as its native packet size. | |
| 210 media::AudioParameters sink_params( | |
| 211 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, sink_channel_layout, | |
| 212 sink_sample_rate, 16, sink_sample_rate / 100); | |
| 213 capture_converter_.reset( | |
| 214 new MediaStreamAudioConverter(source_params, sink_params)); | |
| 215 } | |
| 216 | |
| 217 const media::AudioParameters& MediaStreamAudioProcessor::OutputFormat() const { | |
| 218 return capture_converter_->sink_parameters(); | |
| 219 } | |
| 220 | |
| 221 void MediaStreamAudioProcessor::InitializeAudioProcessingModule( | |
| 222 const webrtc::MediaConstraintsInterface* constraints) { | |
| 223 DCHECK(!audio_processing_); | |
| 224 DCHECK(constraints); | |
| 225 if (!CommandLine::ForCurrentProcess()->HasSwitch( | |
| 226 switches::kEnableAudioTrackProcessing)) { | |
| 227 return; | |
| 228 } | |
| 229 | |
| 230 const bool enable_aec = GetPropertyFromConstraints( | |
| 231 constraints, MediaConstraintsInterface::kEchoCancellation); | |
| 232 const bool enable_ns = GetPropertyFromConstraints( | |
| 233 constraints, MediaConstraintsInterface::kNoiseSuppression); | |
| 234 const bool enable_high_pass_filter = GetPropertyFromConstraints( | |
| 235 constraints, MediaConstraintsInterface::kHighpassFilter); | |
| 236 const bool start_aec_dump = GetPropertyFromConstraints( | |
| 237 constraints, MediaConstraintsInterface::kInternalAecDump); | |
| 238 #if defined(IOS) || defined(ANDROID) | |
| 239 const bool enable_experimental_aec = false; | |
| 240 const bool enable_typing_detection = false; | |
| 241 #else | |
| 242 const bool enable_experimental_aec = GetPropertyFromConstraints( | |
| 243 constraints, MediaConstraintsInterface::kExperimentalEchoCancellation); | |
| 244 const bool enable_typing_detection = GetPropertyFromConstraints( | |
| 245 constraints, MediaConstraintsInterface::kTypingNoiseDetection); | |
| 246 #endif | |
| 247 | |
| 248 // Return immediately if no audio processing component is enabled. | |
| 249 if (!enable_aec && !enable_experimental_aec && !enable_ns && | |
| 250 !enable_high_pass_filter && !enable_typing_detection) { | |
| 251 return; | |
| 252 } | |
| 253 | |
| 254 // Create and configure the webrtc::AudioProcessing. | |
| 255 audio_processing_.reset(webrtc::AudioProcessing::Create(0)); | |
| 256 | |
| 257 // Enable the audio processing components. | |
| 258 if (enable_aec) { | |
| 259 EnableEchoCancellation(audio_processing_.get()); | |
| 260 if (enable_experimental_aec) | |
| 261 EnableExperimentalEchoCancellation(audio_processing_.get()); | |
| 262 } | |
| 263 | |
| 264 if (enable_ns) | |
| 265 EnableNoiseSuppression(audio_processing_.get()); | |
| 266 | |
| 267 if (enable_high_pass_filter) | |
| 268 EnableHighPassFilter(audio_processing_.get()); | |
| 269 | |
| 270 if (enable_typing_detection) | |
| 271 EnableTypingDetection(audio_processing_.get()); | |
| 272 | |
| 273 if (enable_aec && start_aec_dump) | |
| 274 StartAecDump(audio_processing_.get()); | |
| 275 | |
| 276 // Configure the audio format the audio processing is running on. This | |
| 277 // has to be done after all the needed components are enabled. | |
| 278 CHECK_EQ(audio_processing_->set_sample_rate_hz(kAudioProcessingSampleRate), | |
| 279 0); | |
| 280 CHECK_EQ(audio_processing_->set_num_channels(kAudioProcessingNumberOfChannel, | |
| 281 kAudioProcessingNumberOfChannel), | |
| 282 0); | |
| 283 } | |
| 284 | |
| 285 void MediaStreamAudioProcessor::InitializeRenderConverterIfNeeded( | |
| 286 int sample_rate, int number_of_channels, int frames_per_buffer) { | |
| 287 DCHECK(render_thread_checker_.CalledOnValidThread()); | |
| 288 // TODO(xians): Figure out if we need to handle the buffer size change. | |
| 289 if (render_converter_.get() && | |
| 290 render_converter_->source_parameters().sample_rate() == sample_rate && | |
| 291 render_converter_->source_parameters().channels() == number_of_channels) { | |
| 292 // Do nothing if the |render_converter_| has been setup properly. | |
| 293 return; | |
| 294 } | |
| 295 | |
| 296 // Create and initialize audio converter for the render data. | |
| 297 // webrtc::AudioProcessing accepts the same format as what it uses to process | |
| 298 // capture data, which is 32k mono for desktops and 16k mono for Android. | |
| 299 media::AudioParameters source_params( | |
| 300 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 301 media::GuessChannelLayout(number_of_channels), sample_rate, 16, | |
| 302 frames_per_buffer); | |
| 303 media::AudioParameters sink_params( | |
| 304 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 305 media::CHANNEL_LAYOUT_MONO, kAudioProcessingSampleRate, 16, | |
| 306 kAudioProcessingSampleRate / 100); | |
| 307 render_converter_.reset( | |
| 308 new MediaStreamAudioConverter(source_params, sink_params)); | |
| 309 render_data_bus_ = media::AudioBus::Create(number_of_channels, | |
| 310 frames_per_buffer); | |
| 311 } | |
| 312 | |
| 313 void MediaStreamAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame, | |
| 314 base::TimeDelta capture_delay, | |
| 315 int volume, | |
| 316 bool key_pressed) { | |
| 317 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 318 if (!audio_processing_) | |
| 319 return; | |
| 320 | |
| 321 TRACE_EVENT0("audio", "MediaStreamAudioProcessor::Process10MsData"); | |
| 322 DCHECK_EQ(audio_processing_->sample_rate_hz(), | |
| 323 capture_converter_->sink_parameters().sample_rate()); | |
| 324 DCHECK_EQ(audio_processing_->num_input_channels(), | |
| 325 capture_converter_->sink_parameters().channels()); | |
| 326 DCHECK_EQ(audio_processing_->num_output_channels(), | |
| 327 capture_converter_->sink_parameters().channels()); | |
| 328 | |
| 329 base::subtle::Atomic32 render_delay_ms = | |
| 330 base::subtle::Acquire_Load(&render_delay_ms_); | |
| 331 int64 capture_delay_ms = capture_delay.InMilliseconds(); | |
| 332 DCHECK_LT(capture_delay_ms, | |
| 333 std::numeric_limits<base::subtle::Atomic32>::max()); | |
| 334 int total_delay_ms = capture_delay_ms + render_delay_ms; | |
| 335 if (total_delay_ms > 1000) { | |
| 336 LOG(WARNING) << "Large audio delay, capture delay: " << capture_delay_ms | |
| 337 << "ms; render delay: " << render_delay_ms << "ms"; | |
| 338 } | |
| 339 | |
| 340 audio_processing_->set_stream_delay_ms(total_delay_ms); | |
| 341 webrtc::GainControl* agc = audio_processing_->gain_control(); | |
| 342 int err = agc->set_stream_analog_level(volume); | |
| 343 DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err; | |
| 344 err = audio_processing_->ProcessStream(audio_frame); | |
| 345 DCHECK_EQ(err, 0) << "ProcessStream() error: " << err; | |
| 346 | |
| 347 // TODO(xians): Add support for AGC, typing detection, audio level | |
| 348 // calculation, stereo swapping. | |
| 349 } | |
| 350 | |
| 351 void MediaStreamAudioProcessor::StopAudioProcessing() { | |
| 352 if (!audio_processing_.get()) | |
| 353 return; | |
| 354 | |
| 355 // It is safe to stop the AEC dump even it is not started. | |
| 356 StopAecDump(audio_processing_.get()); | |
| 357 | |
| 358 audio_processing_.reset(); | |
| 359 } | |
| 360 | |
| 361 } // namespace content | |
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Chromium Code Reviews
Issue 88283003:
Revert 237311 "Added an "enable-audio-processor" flag and WebRtc..." (Closed)
Base URL: svn://svn.chromium.org/chrome/