Browser-wide audio mirroring for TabCapture API

media/audio/audio_output_resampler.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "media/audio/audio_output_resampler.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/command_line.h"
 #include "base/compiler_specific.h"
 #include "base/message_loop.h"
 #include "base/metrics/histogram.h"
 #include "base/time.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_output_dispatcher_impl.h"
 #include "media/audio/audio_output_proxy.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
 #include "media/base/audio_pull_fifo.h"
 #include "media/base/channel_mixer.h"
 #include "media/base/limits.h"
 #include "media/base/media_switches.h"
 #include "media/base/multi_channel_resampler.h"
 
 namespace media {
 
-class OnMoreDataResampler : public AudioOutputStream::AudioSourceCallback {
- public:
-  OnMoreDataResampler(double io_ratio,
-                      const AudioParameters& input_params,
-                      const AudioParameters& output_params);
-  virtual ~OnMoreDataResampler();
-
-  // AudioSourceCallback interface.
-  virtual int OnMoreData(AudioBus* dest,
-                         AudioBuffersState buffers_state) OVERRIDE;
-  virtual int OnMoreIOData(AudioBus* source,
-                           AudioBus* dest,
-                           AudioBuffersState buffers_state) OVERRIDE;
-  virtual void OnError(AudioOutputStream* stream, int code) OVERRIDE;
-  virtual void WaitTillDataReady() OVERRIDE;
-
-  // Sets |source_callback_|.  If this is not a new object, then Stop() must be
-  // called before Start().
-  void Start(AudioOutputStream::AudioSourceCallback* callback);
-
-  // Clears |source_callback_| and flushes the resampler.
-  void Stop();
-
- private:
-  // Called by MultiChannelResampler when more data is necessary.
-  void ProvideInput(AudioBus* audio_bus);
-
-  // Called by AudioPullFifo when more data is necessary.  Requires
-  // |source_lock_| to have been acquired.
-  void SourceCallback_Locked(AudioBus* audio_bus);
-
-  // Passes through |source| to the |source_callback_| OnMoreIOData() call.
-  void SourceIOCallback_Locked(AudioBus* source, AudioBus* dest);
-
-  // Ratio of input bytes to output bytes used to correct playback delay with
-  // regard to buffering and resampling.
-  double io_ratio_;
-
-  // Source callback and associated lock.
-  base::Lock source_lock_;
-  AudioOutputStream::AudioSourceCallback* source_callback_;
-
-  // Last AudioBuffersState object received via OnMoreData(), used to correct
-  // playback delay by ProvideInput() and passed on to |source_callback_|.
-  AudioBuffersState current_buffers_state_;
-
-  // Total number of bytes (in terms of output parameters) stored in resampler
-  // or FIFO buffers which have not been sent to the audio device.
-  int outstanding_audio_bytes_;
-
-  // Used to buffer data between the client and the output device in cases where
-  // the client buffer size is not the same as the output device buffer size.
-  // Bound to SourceCallback_Locked() so must only be used when |source_lock_|
-  // has already been acquired.
-  scoped_ptr<AudioPullFifo> audio_fifo_;
-
-  // Handles resampling.
-  scoped_ptr<MultiChannelResampler> resampler_;
-
-  // Handles channel transforms.  |unmixed_audio_| is a temporary destination
-  // for audio data before it goes into the channel mixer.
-  scoped_ptr<ChannelMixer> channel_mixer_;
-  scoped_ptr<AudioBus> unmixed_audio_;
-
-  int output_bytes_per_frame_;
-  int input_bytes_per_frame_;
-
-  // Since resampling is expensive, figure out if we should downmix channels
-  // before resampling.
-  bool downmix_early_;
-
-  DISALLOW_COPY_AND_ASSIGN(OnMoreDataResampler);
-};
-
 // Record UMA statistics for hardware output configuration.
 static void RecordStats(const AudioParameters& output_params) {
   UMA_HISTOGRAM_ENUMERATION(
       "Media.HardwareAudioBitsPerChannel", output_params.bits_per_sample(),
       limits::kMaxBitsPerSample);
   UMA_HISTOGRAM_ENUMERATION(
       "Media.HardwareAudioChannelLayout", output_params.channel_layout(),
       CHANNEL_LAYOUT_MAX);
   UMA_HISTOGRAM_ENUMERATION(
       "Media.HardwareAudioChannelCount", output_params.channels(),
@@ skipping 398 matching lines @@
     source_callback_->OnError(stream, code);
 }
 
 void OnMoreDataResampler::WaitTillDataReady() {
   base::AutoLock auto_lock(source_lock_);
   if (source_callback_ && !outstanding_audio_bytes_)
     source_callback_->WaitTillDataReady();
 }
 
 }  // namespace media