audio_output_resampler.cc revision 58537e28ecd584eab876aee8be7156509866d23a
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 "media/audio/audio_output_resampler.h" 6 7#include "base/bind.h" 8#include "base/bind_helpers.h" 9#include "base/compiler_specific.h" 10#include "base/message_loop/message_loop.h" 11#include "base/metrics/histogram.h" 12#include "base/time/time.h" 13#include "build/build_config.h" 14#include "media/audio/audio_io.h" 15#include "media/audio/audio_output_dispatcher_impl.h" 16#include "media/audio/audio_output_proxy.h" 17#include "media/audio/audio_util.h" 18#include "media/audio/sample_rates.h" 19#include "media/base/audio_converter.h" 20#include "media/base/limits.h" 21 22namespace media { 23 24class OnMoreDataConverter 25 : public AudioOutputStream::AudioSourceCallback, 26 public AudioConverter::InputCallback { 27 public: 28 OnMoreDataConverter(const AudioParameters& input_params, 29 const AudioParameters& output_params); 30 virtual ~OnMoreDataConverter(); 31 32 // AudioSourceCallback interface. 33 virtual int OnMoreData(AudioBus* dest, 34 AudioBuffersState buffers_state) OVERRIDE; 35 virtual int OnMoreIOData(AudioBus* source, 36 AudioBus* dest, 37 AudioBuffersState buffers_state) OVERRIDE; 38 virtual void OnError(AudioOutputStream* stream) OVERRIDE; 39 40 // Sets |source_callback_|. If this is not a new object, then Stop() must be 41 // called before Start(). 42 void Start(AudioOutputStream::AudioSourceCallback* callback); 43 44 // Clears |source_callback_| and flushes the resampler. 45 void Stop(); 46 47 private: 48 // AudioConverter::InputCallback implementation. 49 virtual double ProvideInput(AudioBus* audio_bus, 50 base::TimeDelta buffer_delay) OVERRIDE; 51 52 // Ratio of input bytes to output bytes used to correct playback delay with 53 // regard to buffering and resampling. 54 double io_ratio_; 55 56 // Source callback and associated lock. 57 base::Lock source_lock_; 58 AudioOutputStream::AudioSourceCallback* source_callback_; 59 60 // |source| passed to OnMoreIOData() which should be passed downstream. 61 AudioBus* source_bus_; 62 63 // Last AudioBuffersState object received via OnMoreData(), used to correct 64 // playback delay by ProvideInput() and passed on to |source_callback_|. 65 AudioBuffersState current_buffers_state_; 66 67 const int input_bytes_per_second_; 68 69 // Handles resampling, buffering, and channel mixing between input and output 70 // parameters. 71 AudioConverter audio_converter_; 72 73 DISALLOW_COPY_AND_ASSIGN(OnMoreDataConverter); 74}; 75 76// Record UMA statistics for hardware output configuration. 77static void RecordStats(const AudioParameters& output_params) { 78 UMA_HISTOGRAM_ENUMERATION( 79 "Media.HardwareAudioBitsPerChannel", output_params.bits_per_sample(), 80 limits::kMaxBitsPerSample); 81 UMA_HISTOGRAM_ENUMERATION( 82 "Media.HardwareAudioChannelLayout", output_params.channel_layout(), 83 CHANNEL_LAYOUT_MAX); 84 UMA_HISTOGRAM_ENUMERATION( 85 "Media.HardwareAudioChannelCount", output_params.channels(), 86 limits::kMaxChannels); 87 88 AudioSampleRate asr = media::AsAudioSampleRate(output_params.sample_rate()); 89 if (asr != kUnexpectedAudioSampleRate) { 90 UMA_HISTOGRAM_ENUMERATION( 91 "Media.HardwareAudioSamplesPerSecond", asr, kUnexpectedAudioSampleRate); 92 } else { 93 UMA_HISTOGRAM_COUNTS( 94 "Media.HardwareAudioSamplesPerSecondUnexpected", 95 output_params.sample_rate()); 96 } 97} 98 99// Record UMA statistics for hardware output configuration after fallback. 100static void RecordFallbackStats(const AudioParameters& output_params) { 101 UMA_HISTOGRAM_BOOLEAN("Media.FallbackToHighLatencyAudioPath", true); 102 UMA_HISTOGRAM_ENUMERATION( 103 "Media.FallbackHardwareAudioBitsPerChannel", 104 output_params.bits_per_sample(), limits::kMaxBitsPerSample); 105 UMA_HISTOGRAM_ENUMERATION( 106 "Media.FallbackHardwareAudioChannelLayout", 107 output_params.channel_layout(), CHANNEL_LAYOUT_MAX); 108 UMA_HISTOGRAM_ENUMERATION( 109 "Media.FallbackHardwareAudioChannelCount", 110 output_params.channels(), limits::kMaxChannels); 111 112 AudioSampleRate asr = media::AsAudioSampleRate(output_params.sample_rate()); 113 if (asr != kUnexpectedAudioSampleRate) { 114 UMA_HISTOGRAM_ENUMERATION( 115 "Media.FallbackHardwareAudioSamplesPerSecond", 116 asr, kUnexpectedAudioSampleRate); 117 } else { 118 UMA_HISTOGRAM_COUNTS( 119 "Media.FallbackHardwareAudioSamplesPerSecondUnexpected", 120 output_params.sample_rate()); 121 } 122} 123 124// Only Windows has a high latency output driver that is not the same as the low 125// latency path. 126#if defined(OS_WIN) 127// Converts low latency based |output_params| into high latency appropriate 128// output parameters in error situations. 129static AudioParameters SetupFallbackParams( 130 const AudioParameters& input_params, const AudioParameters& output_params) { 131 // Choose AudioParameters appropriate for opening the device in high latency 132 // mode. |kMinLowLatencyFrameSize| is arbitrarily based on Pepper Flash's 133 // MAXIMUM frame size for low latency. 134 static const int kMinLowLatencyFrameSize = 2048; 135 int frames_per_buffer = std::min( 136 std::max(input_params.frames_per_buffer(), kMinLowLatencyFrameSize), 137 static_cast<int>( 138 GetHighLatencyOutputBufferSize(input_params.sample_rate()))); 139 140 return AudioParameters( 141 AudioParameters::AUDIO_PCM_LINEAR, input_params.channel_layout(), 142 input_params.sample_rate(), input_params.bits_per_sample(), 143 frames_per_buffer); 144} 145#endif 146 147AudioOutputResampler::AudioOutputResampler(AudioManager* audio_manager, 148 const AudioParameters& input_params, 149 const AudioParameters& output_params, 150 const std::string& output_device_id, 151 const std::string& input_device_id, 152 const base::TimeDelta& close_delay) 153 : AudioOutputDispatcher(audio_manager, input_params, output_device_id, 154 input_device_id), 155 close_delay_(close_delay), 156 output_params_(output_params), 157 streams_opened_(false) { 158 DCHECK(input_params.IsValid()); 159 DCHECK(output_params.IsValid()); 160 DCHECK_EQ(output_params_.format(), AudioParameters::AUDIO_PCM_LOW_LATENCY); 161 162 // Record UMA statistics for the hardware configuration. 163 RecordStats(output_params); 164 165 Initialize(); 166} 167 168AudioOutputResampler::~AudioOutputResampler() { 169 DCHECK(callbacks_.empty()); 170} 171 172void AudioOutputResampler::Initialize() { 173 DCHECK(!streams_opened_); 174 DCHECK(callbacks_.empty()); 175 dispatcher_ = new AudioOutputDispatcherImpl( 176 audio_manager_, output_params_, output_device_id_, input_device_id_, 177 close_delay_); 178} 179 180bool AudioOutputResampler::OpenStream() { 181 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 182 183 if (dispatcher_->OpenStream()) { 184 // Only record the UMA statistic if we didn't fallback during construction 185 // and only for the first stream we open. 186 if (!streams_opened_ && 187 output_params_.format() == AudioParameters::AUDIO_PCM_LOW_LATENCY) { 188 UMA_HISTOGRAM_BOOLEAN("Media.FallbackToHighLatencyAudioPath", false); 189 } 190 streams_opened_ = true; 191 return true; 192 } 193 194 // If we've already tried to open the stream in high latency mode or we've 195 // successfully opened a stream previously, there's nothing more to be done. 196 if (output_params_.format() != AudioParameters::AUDIO_PCM_LOW_LATENCY || 197 streams_opened_ || !callbacks_.empty()) { 198 return false; 199 } 200 201 DCHECK_EQ(output_params_.format(), AudioParameters::AUDIO_PCM_LOW_LATENCY); 202 203 // Record UMA statistics about the hardware which triggered the failure so 204 // we can debug and triage later. 205 RecordFallbackStats(output_params_); 206 207 // Only Windows has a high latency output driver that is not the same as the 208 // low latency path. 209#if defined(OS_WIN) 210 DLOG(ERROR) << "Unable to open audio device in low latency mode. Falling " 211 << "back to high latency audio output."; 212 213 output_params_ = SetupFallbackParams(params_, output_params_); 214 Initialize(); 215 if (dispatcher_->OpenStream()) { 216 streams_opened_ = true; 217 return true; 218 } 219#endif 220 221 DLOG(ERROR) << "Unable to open audio device in high latency mode. Falling " 222 << "back to fake audio output."; 223 224 // Finally fall back to a fake audio output device. 225 output_params_.Reset( 226 AudioParameters::AUDIO_FAKE, params_.channel_layout(), 227 params_.channels(), params_.input_channels(), params_.sample_rate(), 228 params_.bits_per_sample(), params_.frames_per_buffer()); 229 Initialize(); 230 if (dispatcher_->OpenStream()) { 231 streams_opened_ = true; 232 return true; 233 } 234 235 return false; 236} 237 238bool AudioOutputResampler::StartStream( 239 AudioOutputStream::AudioSourceCallback* callback, 240 AudioOutputProxy* stream_proxy) { 241 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 242 243 OnMoreDataConverter* resampler_callback = NULL; 244 CallbackMap::iterator it = callbacks_.find(stream_proxy); 245 if (it == callbacks_.end()) { 246 resampler_callback = new OnMoreDataConverter(params_, output_params_); 247 callbacks_[stream_proxy] = resampler_callback; 248 } else { 249 resampler_callback = it->second; 250 } 251 252 resampler_callback->Start(callback); 253 bool result = dispatcher_->StartStream(resampler_callback, stream_proxy); 254 if (!result) 255 resampler_callback->Stop(); 256 return result; 257} 258 259void AudioOutputResampler::StreamVolumeSet(AudioOutputProxy* stream_proxy, 260 double volume) { 261 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 262 dispatcher_->StreamVolumeSet(stream_proxy, volume); 263} 264 265void AudioOutputResampler::StopStream(AudioOutputProxy* stream_proxy) { 266 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 267 dispatcher_->StopStream(stream_proxy); 268 269 // Now that StopStream() has completed the underlying physical stream should 270 // be stopped and no longer calling OnMoreData(), making it safe to Stop() the 271 // OnMoreDataConverter. 272 CallbackMap::iterator it = callbacks_.find(stream_proxy); 273 if (it != callbacks_.end()) 274 it->second->Stop(); 275} 276 277void AudioOutputResampler::CloseStream(AudioOutputProxy* stream_proxy) { 278 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 279 dispatcher_->CloseStream(stream_proxy); 280 281 // We assume that StopStream() is always called prior to CloseStream(), so 282 // that it is safe to delete the OnMoreDataConverter here. 283 CallbackMap::iterator it = callbacks_.find(stream_proxy); 284 if (it != callbacks_.end()) { 285 delete it->second; 286 callbacks_.erase(it); 287 } 288} 289 290void AudioOutputResampler::Shutdown() { 291 DCHECK_EQ(base::MessageLoop::current(), message_loop_); 292 293 // No AudioOutputProxy objects should hold a reference to us when we get 294 // to this stage. 295 DCHECK(HasOneRef()) << "Only the AudioManager should hold a reference"; 296 297 dispatcher_->Shutdown(); 298 DCHECK(callbacks_.empty()); 299} 300 301OnMoreDataConverter::OnMoreDataConverter(const AudioParameters& input_params, 302 const AudioParameters& output_params) 303 : source_callback_(NULL), 304 source_bus_(NULL), 305 input_bytes_per_second_(input_params.GetBytesPerSecond()), 306 audio_converter_(input_params, output_params, false) { 307 io_ratio_ = 308 static_cast<double>(input_params.GetBytesPerSecond()) / 309 output_params.GetBytesPerSecond(); 310} 311 312OnMoreDataConverter::~OnMoreDataConverter() { 313 // Ensure Stop() has been called so we don't end up with an AudioOutputStream 314 // calling back into OnMoreData() after destruction. 315 CHECK(!source_callback_); 316} 317 318void OnMoreDataConverter::Start( 319 AudioOutputStream::AudioSourceCallback* callback) { 320 base::AutoLock auto_lock(source_lock_); 321 CHECK(!source_callback_); 322 source_callback_ = callback; 323 324 // While AudioConverter can handle multiple inputs, we're using it only with 325 // a single input currently. Eventually this may be the basis for a browser 326 // side mixer. 327 audio_converter_.AddInput(this); 328} 329 330void OnMoreDataConverter::Stop() { 331 base::AutoLock auto_lock(source_lock_); 332 CHECK(source_callback_); 333 source_callback_ = NULL; 334 audio_converter_.RemoveInput(this); 335} 336 337int OnMoreDataConverter::OnMoreData(AudioBus* dest, 338 AudioBuffersState buffers_state) { 339 return OnMoreIOData(NULL, dest, buffers_state); 340} 341 342int OnMoreDataConverter::OnMoreIOData(AudioBus* source, 343 AudioBus* dest, 344 AudioBuffersState buffers_state) { 345 base::AutoLock auto_lock(source_lock_); 346 // While we waited for |source_lock_| the callback might have been cleared. 347 if (!source_callback_) { 348 dest->Zero(); 349 return dest->frames(); 350 } 351 352 source_bus_ = source; 353 current_buffers_state_ = buffers_state; 354 audio_converter_.Convert(dest); 355 356 // Always return the full number of frames requested, ProvideInput_Locked() 357 // will pad with silence if it wasn't able to acquire enough data. 358 return dest->frames(); 359} 360 361double OnMoreDataConverter::ProvideInput(AudioBus* dest, 362 base::TimeDelta buffer_delay) { 363 source_lock_.AssertAcquired(); 364 365 // Adjust playback delay to include |buffer_delay|. 366 // TODO(dalecurtis): Stop passing bytes around, it doesn't make sense since 367 // AudioBus is just float data. Use TimeDelta instead. 368 AudioBuffersState new_buffers_state; 369 new_buffers_state.pending_bytes = 370 io_ratio_ * (current_buffers_state_.total_bytes() + 371 buffer_delay.InSecondsF() * input_bytes_per_second_); 372 373 // Retrieve data from the original callback. 374 int frames = source_callback_->OnMoreIOData( 375 source_bus_, dest, new_buffers_state); 376 377 // |source_bus_| should only be provided once. 378 // TODO(dalecurtis, crogers): This is not a complete fix. If ProvideInput() 379 // is called multiple times, we need to do something more clever here. 380 source_bus_ = NULL; 381 382 // Zero any unfilled frames if anything was filled, otherwise we'll just 383 // return a volume of zero and let AudioConverter drop the output. 384 if (frames > 0 && frames < dest->frames()) 385 dest->ZeroFramesPartial(frames, dest->frames() - frames); 386 387 // TODO(dalecurtis): Return the correct volume here. 388 return frames > 0 ? 1 : 0; 389} 390 391void OnMoreDataConverter::OnError(AudioOutputStream* stream) { 392 base::AutoLock auto_lock(source_lock_); 393 if (source_callback_) 394 source_callback_->OnError(stream); 395} 396 397} // namespace media 398