media/audio/audio_output_resampler.cc

   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/metrics/histogram.h"
  11 #include "base/single_thread_task_runner.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/sample_rates.h"
  18 #include "media/base/audio_converter.h"
  19 #include "media/base/limits.h"
  20
  21 namespace media {
  22
  23 class OnMoreDataConverter
  24     : public AudioOutputStream::AudioSourceCallback,
  25       public AudioConverter::InputCallback {
  26  public:
  27   OnMoreDataConverter(const AudioParameters& input_params,
  28                       const AudioParameters& output_params);
  29   virtual ~OnMoreDataConverter();
  30
  31   // AudioSourceCallback interface.
  32   virtual int OnMoreData(AudioBus* dest,
  33                          AudioBuffersState buffers_state) OVERRIDE;
  34   virtual int OnMoreIOData(AudioBus* source,
  35                            AudioBus* dest,
  36                            AudioBuffersState buffers_state) OVERRIDE;
  37   virtual void OnError(AudioOutputStream* stream) OVERRIDE;
  38
  39   // Sets |source_callback_|.  If this is not a new object, then Stop() must be
  40   // called before Start().
  41   void Start(AudioOutputStream::AudioSourceCallback* callback);
  42
  43   // Clears |source_callback_| and flushes the resampler.
  44   void Stop();
  45
  46   bool started() { return source_callback_ != NULL; }
  47
  48  private:
  49   // AudioConverter::InputCallback implementation.
  50   virtual double ProvideInput(AudioBus* audio_bus,
  51                               base::TimeDelta buffer_delay) OVERRIDE;
  52
  53   // Ratio of input bytes to output bytes used to correct playback delay with
  54   // regard to buffering and resampling.
  55   const double io_ratio_;
  56
  57   // Source callback.
  58   AudioOutputStream::AudioSourceCallback* source_callback_;
  59
  60   // Last AudioBuffersState object received via OnMoreData(), used to correct
  61   // playback delay by ProvideInput() and passed on to |source_callback_|.
  62   AudioBuffersState current_buffers_state_;
  63
  64   const int input_bytes_per_second_;
  65
  66   // Handles resampling, buffering, and channel mixing between input and output
  67   // parameters.
  68   AudioConverter audio_converter_;
  69
  70   DISALLOW_COPY_AND_ASSIGN(OnMoreDataConverter);
  71 };
  72
  73 // Record UMA statistics for hardware output configuration.
  74 static void RecordStats(const AudioParameters& output_params) {
  75   // Note the 'PRESUBMIT_IGNORE_UMA_MAX's below, these silence the PRESUBMIT.py
  76   // check for uma enum max usage, since we're abusing UMA_HISTOGRAM_ENUMERATION
  77   // to report a discrete value.
  78   UMA_HISTOGRAM_ENUMERATION(
  79       "Media.HardwareAudioBitsPerChannel",
  80       output_params.bits_per_sample(),
  81       limits::kMaxBitsPerSample);  // PRESUBMIT_IGNORE_UMA_MAX
  82   UMA_HISTOGRAM_ENUMERATION(
  83       "Media.HardwareAudioChannelLayout", output_params.channel_layout(),
  84       CHANNEL_LAYOUT_MAX + 1);
  85   UMA_HISTOGRAM_ENUMERATION(
  86       "Media.HardwareAudioChannelCount", output_params.channels(),
  87       limits::kMaxChannels);  // PRESUBMIT_IGNORE_UMA_MAX
  88
  89   AudioSampleRate asr;
  90   if (ToAudioSampleRate(output_params.sample_rate(), &asr)) {
  91     UMA_HISTOGRAM_ENUMERATION(
  92         "Media.HardwareAudioSamplesPerSecond", asr, kAudioSampleRateMax + 1);
  93   } else {
  94     UMA_HISTOGRAM_COUNTS(
  95         "Media.HardwareAudioSamplesPerSecondUnexpected",
  96         output_params.sample_rate());
  97   }
  98 }
  99
 100 // Record UMA statistics for hardware output configuration after fallback.
 101 static void RecordFallbackStats(const AudioParameters& output_params) {
 102   UMA_HISTOGRAM_BOOLEAN("Media.FallbackToHighLatencyAudioPath", true);
 103   // Note the 'PRESUBMIT_IGNORE_UMA_MAX's below, these silence the PRESUBMIT.py
 104   // check for uma enum max usage, since we're abusing UMA_HISTOGRAM_ENUMERATION
 105   // to report a discrete value.
 106   UMA_HISTOGRAM_ENUMERATION(
 107       "Media.FallbackHardwareAudioBitsPerChannel",
 108       output_params.bits_per_sample(),
 109       limits::kMaxBitsPerSample);  // PRESUBMIT_IGNORE_UMA_MAX
 110   UMA_HISTOGRAM_ENUMERATION(
 111       "Media.FallbackHardwareAudioChannelLayout",
 112       output_params.channel_layout(), CHANNEL_LAYOUT_MAX + 1);
 113   UMA_HISTOGRAM_ENUMERATION(
 114       "Media.FallbackHardwareAudioChannelCount", output_params.channels(),
 115       limits::kMaxChannels);  // PRESUBMIT_IGNORE_UMA_MAX
 116
 117   AudioSampleRate asr;
 118   if (ToAudioSampleRate(output_params.sample_rate(), &asr)) {
 119     UMA_HISTOGRAM_ENUMERATION(
 120         "Media.FallbackHardwareAudioSamplesPerSecond",
 121         asr, kAudioSampleRateMax + 1);
 122   } else {
 123     UMA_HISTOGRAM_COUNTS(
 124         "Media.FallbackHardwareAudioSamplesPerSecondUnexpected",
 125         output_params.sample_rate());
 126   }
 127 }
 128
 129 // Converts low latency based |output_params| into high latency appropriate
 130 // output parameters in error situations.
 131 void AudioOutputResampler::SetupFallbackParams() {
 132 // Only Windows has a high latency output driver that is not the same as the low
 133 // latency path.
 134 #if defined(OS_WIN)
 135   // Choose AudioParameters appropriate for opening the device in high latency
 136   // mode.  |kMinLowLatencyFrameSize| is arbitrarily based on Pepper Flash's
 137   // MAXIMUM frame size for low latency.
 138   static const int kMinLowLatencyFrameSize = 2048;
 139   const int frames_per_buffer =
 140       std::max(params_.frames_per_buffer(), kMinLowLatencyFrameSize);
 141
 142   output_params_ = AudioParameters(
 143       AudioParameters::AUDIO_PCM_LINEAR, params_.channel_layout(),
 144       params_.sample_rate(), params_.bits_per_sample(),
 145       frames_per_buffer);
 146   device_id_ = "";
 147   Initialize();
 148 #endif
 149 }
 150
 151 AudioOutputResampler::AudioOutputResampler(AudioManager* audio_manager,
 152                                            const AudioParameters& input_params,
 153                                            const AudioParameters& output_params,
 154                                            const std::string& output_device_id,
 155                                            const base::TimeDelta& close_delay)
 156     : AudioOutputDispatcher(audio_manager, input_params, output_device_id),
 157       close_delay_(close_delay),
 158       output_params_(output_params),
 159       streams_opened_(false) {
 160   DCHECK(input_params.IsValid());
 161   DCHECK(output_params.IsValid());
 162   DCHECK_EQ(output_params_.format(), AudioParameters::AUDIO_PCM_LOW_LATENCY);
 163
 164   // Record UMA statistics for the hardware configuration.
 165   RecordStats(output_params);
 166
 167   Initialize();
 168 }
 169
 170 AudioOutputResampler::~AudioOutputResampler() {
 171   DCHECK(callbacks_.empty());
 172 }
 173
 174 void AudioOutputResampler::Initialize() {
 175   DCHECK(!streams_opened_);
 176   DCHECK(callbacks_.empty());
 177   dispatcher_ = new AudioOutputDispatcherImpl(
 178       audio_manager_, output_params_, device_id_, close_delay_);
 179 }
 180
 181 bool AudioOutputResampler::OpenStream() {
 182   DCHECK(task_runner_->BelongsToCurrentThread());
 183
 184   if (dispatcher_->OpenStream()) {
 185     // Only record the UMA statistic if we didn't fallback during construction
 186     // and only for the first stream we open.
 187     if (!streams_opened_ &&
 188         output_params_.format() == AudioParameters::AUDIO_PCM_LOW_LATENCY) {
 189       UMA_HISTOGRAM_BOOLEAN("Media.FallbackToHighLatencyAudioPath", false);
 190     }
 191     streams_opened_ = true;
 192     return true;
 193   }
 194
 195   // If we've already tried to open the stream in high latency mode or we've
 196   // successfully opened a stream previously, there's nothing more to be done.
 197   if (output_params_.format() != AudioParameters::AUDIO_PCM_LOW_LATENCY ||
 198       streams_opened_ || !callbacks_.empty()) {
 199     return false;
 200   }
 201
 202   DCHECK_EQ(output_params_.format(), AudioParameters::AUDIO_PCM_LOW_LATENCY);
 203
 204   // Record UMA statistics about the hardware which triggered the failure so
 205   // we can debug and triage later.
 206   RecordFallbackStats(output_params_);
 207
 208   // Only Windows has a high latency output driver that is not the same as the
 209   // low latency path.
 210 #if defined(OS_WIN)
 211   DLOG(ERROR) << "Unable to open audio device in low latency mode.  Falling "
 212               << "back to high latency audio output.";
 213
 214   SetupFallbackParams();
 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
 238 bool AudioOutputResampler::StartStream(
 239     AudioOutputStream::AudioSourceCallback* callback,
 240     AudioOutputProxy* stream_proxy) {
 241   DCHECK(task_runner_->BelongsToCurrentThread());
 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
 259 void AudioOutputResampler::StreamVolumeSet(AudioOutputProxy* stream_proxy,
 260                                            double volume) {
 261   DCHECK(task_runner_->BelongsToCurrentThread());
 262   dispatcher_->StreamVolumeSet(stream_proxy, volume);
 263 }
 264
 265 void AudioOutputResampler::StopStream(AudioOutputProxy* stream_proxy) {
 266   DCHECK(task_runner_->BelongsToCurrentThread());
 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
 277 void AudioOutputResampler::CloseStream(AudioOutputProxy* stream_proxy) {
 278   DCHECK(task_runner_->BelongsToCurrentThread());
 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
 290 void AudioOutputResampler::Shutdown() {
 291   DCHECK(task_runner_->BelongsToCurrentThread());
 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
 301 OnMoreDataConverter::OnMoreDataConverter(const AudioParameters& input_params,
 302                                          const AudioParameters& output_params)
 303     : io_ratio_(static_cast<double>(input_params.GetBytesPerSecond()) /
 304                 output_params.GetBytesPerSecond()),
 305       source_callback_(NULL),
 306       input_bytes_per_second_(input_params.GetBytesPerSecond()),
 307       audio_converter_(input_params, output_params, false) {}
 308
 309 OnMoreDataConverter::~OnMoreDataConverter() {
 310   // Ensure Stop() has been called so we don't end up with an AudioOutputStream
 311   // calling back into OnMoreData() after destruction.
 312   CHECK(!source_callback_);
 313 }
 314
 315 void OnMoreDataConverter::Start(
 316     AudioOutputStream::AudioSourceCallback* callback) {
 317   CHECK(!source_callback_);
 318   source_callback_ = callback;
 319
 320   // While AudioConverter can handle multiple inputs, we're using it only with
 321   // a single input currently.  Eventually this may be the basis for a browser
 322   // side mixer.
 323   audio_converter_.AddInput(this);
 324 }
 325
 326 void OnMoreDataConverter::Stop() {
 327   CHECK(source_callback_);
 328   source_callback_ = NULL;
 329   audio_converter_.RemoveInput(this);
 330 }
 331
 332 int OnMoreDataConverter::OnMoreData(AudioBus* dest,
 333                                     AudioBuffersState buffers_state) {
 334   return OnMoreIOData(NULL, dest, buffers_state);
 335 }
 336
 337 int OnMoreDataConverter::OnMoreIOData(AudioBus* source,
 338                                       AudioBus* dest,
 339                                       AudioBuffersState buffers_state) {
 340   // Note: The input portion of OnMoreIOData() is not supported when a converter
 341   // has been injected.  Downstream clients prefer silence to potentially split
 342   // apart input data.
 343
 344   current_buffers_state_ = buffers_state;
 345   audio_converter_.Convert(dest);
 346
 347   // Always return the full number of frames requested, ProvideInput()
 348   // will pad with silence if it wasn't able to acquire enough data.
 349   return dest->frames();
 350 }
 351
 352 double OnMoreDataConverter::ProvideInput(AudioBus* dest,
 353                                          base::TimeDelta buffer_delay) {
 354   // Adjust playback delay to include |buffer_delay|.
 355   // TODO(dalecurtis): Stop passing bytes around, it doesn't make sense since
 356   // AudioBus is just float data.  Use TimeDelta instead.
 357   AudioBuffersState new_buffers_state;
 358   new_buffers_state.pending_bytes =
 359       io_ratio_ * (current_buffers_state_.total_bytes() +
 360                    buffer_delay.InSecondsF() * input_bytes_per_second_);
 361
 362   // Retrieve data from the original callback.
 363   const int frames = source_callback_->OnMoreIOData(
 364       NULL, dest, new_buffers_state);
 365
 366   // Zero any unfilled frames if anything was filled, otherwise we'll just
 367   // return a volume of zero and let AudioConverter drop the output.
 368   if (frames > 0 && frames < dest->frames())
 369     dest->ZeroFramesPartial(frames, dest->frames() - frames);
 370   return frames > 0 ? 1 : 0;
 371 }
 372
 373 void OnMoreDataConverter::OnError(AudioOutputStream* stream) {
 374   source_callback_->OnError(stream);
 375 }
 376
 377 }  // namespace media