Reland 153623004: Remove the unified IO code on the browser

media/audio/mac/audio_synchronized_mac.cc

Index: media/audio/mac/audio_synchronized_mac.cc
diff --git a/media/audio/mac/audio_synchronized_mac.cc b/media/audio/mac/audio_synchronized_mac.cc
deleted file mode 100644
index a9bc88e2bd3be7205dab6062ed63fa44d2db0b7c..0000000000000000000000000000000000000000
--- a/media/audio/mac/audio_synchronized_mac.cc
+++ /dev/null
@@ -1,976 +0,0 @@
-// 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/mac/audio_synchronized_mac.h"
-
-#include <CoreServices/CoreServices.h>
-#include <algorithm>
-
-#include "base/basictypes.h"
-#include "base/debug/trace_event.h"
-#include "base/logging.h"
-#include "base/mac/mac_logging.h"
-#include "media/audio/mac/audio_manager_mac.h"
-#include "media/base/channel_mixer.h"
-
-namespace media {
-
-static const int kHardwareBufferSize = 128;
-static const int kFifoSize = 16384;
-
-// TODO(crogers): handle the non-stereo case.
-static const int kChannels = 2;
-
-// This value was determined empirically for minimum latency while still
-// guarding against FIFO under-runs.
-static const int kBaseTargetFifoFrames = 256 + 64;
-
-// If the input and output sample-rate don't match, then we need to maintain
-// an additional safety margin due to the callback timing jitter and the
-// varispeed buffering.  This value was empirically tuned.
-static const int kAdditionalTargetFifoFrames = 128;
-
-static void ZeroBufferList(AudioBufferList* buffer_list) {
-  for (size_t i = 0; i < buffer_list->mNumberBuffers; ++i)
-    memset(buffer_list->mBuffers[i].mData,
-           0,
-           buffer_list->mBuffers[i].mDataByteSize);
-}
-
-static void WrapBufferList(AudioBufferList* buffer_list,
-                           AudioBus* bus,
-                           int frames) {
-  DCHECK(buffer_list);
-  DCHECK(bus);
-  int channels = bus->channels();
-  int buffer_list_channels = buffer_list->mNumberBuffers;
-
-  // Copy pointers from AudioBufferList.
-  int source_idx = 0;
-  for (int i = 0; i < channels; ++i) {
-    bus->SetChannelData(
-        i, static_cast<float*>(buffer_list->mBuffers[source_idx].mData));
-
-    // It's ok to pass in a |buffer_list| with fewer channels, in which
-    // case we just duplicate the last channel.
-    if (source_idx < buffer_list_channels - 1)
-      ++source_idx;
-  }
-
-  // Finally set the actual length.
-  bus->set_frames(frames);
-}
-
-AudioSynchronizedStream::AudioSynchronizedStream(
-    AudioManagerMac* manager,
-    const AudioParameters& params,
-    AudioDeviceID input_id,
-    AudioDeviceID output_id)
-    : manager_(manager),
-      params_(params),
-      input_sample_rate_(0),
-      output_sample_rate_(0),
-      input_id_(input_id),
-      output_id_(output_id),
-      input_buffer_list_(NULL),
-      fifo_(kChannels, kFifoSize),
-      target_fifo_frames_(kBaseTargetFifoFrames),
-      average_delta_(0.0),
-      fifo_rate_compensation_(1.0),
-      input_unit_(0),
-      varispeed_unit_(0),
-      output_unit_(0),
-      first_input_time_(-1),
-      is_running_(false),
-      hardware_buffer_size_(kHardwareBufferSize),
-      channels_(kChannels) {
-  VLOG(1) << "AudioSynchronizedStream::AudioSynchronizedStream()";
-}
-
-AudioSynchronizedStream::~AudioSynchronizedStream() {
-  DCHECK(!input_unit_);
-  DCHECK(!output_unit_);
-  DCHECK(!varispeed_unit_);
-}
-
-bool AudioSynchronizedStream::Open() {
-  if (params_.channels() != kChannels) {
-    LOG(ERROR) << "Only stereo output is currently supported.";
-    return false;
-  }
-
-  // Create the input, output, and varispeed AudioUnits.
-  OSStatus result = CreateAudioUnits();
-  if (result != noErr) {
-    LOG(ERROR) << "Cannot create AudioUnits.";
-    return false;
-  }
-
-  result = SetupInput(input_id_);
-  if (result != noErr) {
-    LOG(ERROR) << "Error configuring input AudioUnit.";
-    return false;
-  }
-
-  result = SetupOutput(output_id_);
-  if (result != noErr) {
-    LOG(ERROR) << "Error configuring output AudioUnit.";
-    return false;
-  }
-
-  result = SetupCallbacks();
-  if (result != noErr) {
-    LOG(ERROR) << "Error setting up callbacks on AudioUnits.";
-    return false;
-  }
-
-  result = SetupStreamFormats();
-  if (result != noErr) {
-    LOG(ERROR) << "Error configuring stream formats on AudioUnits.";
-    return false;
-  }
-
-  AllocateInputData();
-
-  // Final initialization of the AudioUnits.
-  result = AudioUnitInitialize(input_unit_);
-  if (result != noErr) {
-    LOG(ERROR) << "Error initializing input AudioUnit.";
-    return false;
-  }
-
-  result = AudioUnitInitialize(output_unit_);
-  if (result != noErr) {
-    LOG(ERROR) << "Error initializing output AudioUnit.";
-    return false;
-  }
-
-  result = AudioUnitInitialize(varispeed_unit_);
-  if (result != noErr) {
-    LOG(ERROR) << "Error initializing varispeed AudioUnit.";
-    return false;
-  }
-
-  if (input_sample_rate_ != output_sample_rate_) {
-    // Add extra safety margin.
-    target_fifo_frames_ += kAdditionalTargetFifoFrames;
-  }
-
-  // Buffer initial silence corresponding to target I/O buffering.
-  fifo_.Clear();
-  scoped_ptr<AudioBus> silence =
-      AudioBus::Create(channels_, target_fifo_frames_);
-  silence->Zero();
-  fifo_.Push(silence.get());
-
-  return true;
-}
-
-void AudioSynchronizedStream::Close() {
-  DCHECK(!is_running_);
-
-  if (input_buffer_list_) {
-    free(input_buffer_list_);
-    input_buffer_list_ = 0;
-    input_bus_.reset(NULL);
-    wrapper_bus_.reset(NULL);
-  }
-
-  if (input_unit_) {
-    AudioUnitUninitialize(input_unit_);
-    CloseComponent(input_unit_);
-  }
-
-  if (output_unit_) {
-    AudioUnitUninitialize(output_unit_);
-    CloseComponent(output_unit_);
-  }
-
-  if (varispeed_unit_) {
-    AudioUnitUninitialize(varispeed_unit_);
-    CloseComponent(varispeed_unit_);
-  }
-
-  input_unit_ = NULL;
-  output_unit_ = NULL;
-  varispeed_unit_ = NULL;
-
-  // Inform the audio manager that we have been closed. This can cause our
-  // destruction.
-  manager_->ReleaseOutputStream(this);
-}
-
-void AudioSynchronizedStream::Start(AudioSourceCallback* callback) {
-  DCHECK(callback);
-  DCHECK(input_unit_);
-  DCHECK(output_unit_);
-  DCHECK(varispeed_unit_);
-
-  if (is_running_ || !input_unit_ || !output_unit_ || !varispeed_unit_)
-    return;
-
-  source_ = callback;
-
-  // Reset state variables each time we Start().
-  fifo_rate_compensation_ = 1.0;
-  average_delta_ = 0.0;
-
-  OSStatus result = noErr;
-
-  if (!is_running_) {
-    first_input_time_ = -1;
-
-    result = AudioOutputUnitStart(input_unit_);
-    OSSTATUS_DCHECK(result == noErr, result);
-
-    if (result == noErr) {
-      result = AudioOutputUnitStart(output_unit_);
-      OSSTATUS_DCHECK(result == noErr, result);
-    }
-  }
-
-  is_running_ = true;
-}
-
-void AudioSynchronizedStream::Stop() {
-  OSStatus result = noErr;
-  if (is_running_) {
-    result = AudioOutputUnitStop(input_unit_);
-    OSSTATUS_DCHECK(result == noErr, result);
-
-    if (result == noErr) {
-      result = AudioOutputUnitStop(output_unit_);
-      OSSTATUS_DCHECK(result == noErr, result);
-    }
-  }
-
-  if (result == noErr)
-    is_running_ = false;
-}
-
-bool AudioSynchronizedStream::IsRunning() {
-  return is_running_;
-}
-
-// TODO(crogers): implement - or remove from AudioOutputStream.
-void AudioSynchronizedStream::SetVolume(double volume) {}
-void AudioSynchronizedStream::GetVolume(double* volume) {}
-
-OSStatus AudioSynchronizedStream::SetOutputDeviceAsCurrent(
-    AudioDeviceID output_id) {
-  OSStatus result = noErr;
-
-  // Get the default output device if device is unknown.
-  if (output_id == kAudioDeviceUnknown) {
-    AudioObjectPropertyAddress pa;
-    pa.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
-    pa.mScope = kAudioObjectPropertyScopeGlobal;
-    pa.mElement = kAudioObjectPropertyElementMaster;
-    UInt32 size = sizeof(output_id);
-
-    result = AudioObjectGetPropertyData(
-        kAudioObjectSystemObject,
-        &pa,
-        0,
-        0,
-        &size,
-        &output_id);
-
-    OSSTATUS_DCHECK(result == noErr, result);
-    if (result != noErr)
-      return result;
-  }
-
-  // Set the render frame size.
-  UInt32 frame_size = hardware_buffer_size_;
-  AudioObjectPropertyAddress pa;
-  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
-  pa.mScope = kAudioDevicePropertyScopeInput;
-  pa.mElement = kAudioObjectPropertyElementMaster;
-  result = AudioObjectSetPropertyData(
-      output_id,
-      &pa,
-      0,
-      0,
-      sizeof(frame_size),
-      &frame_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  output_info_.Initialize(output_id, false);
-
-  // Set the Current Device to the Default Output Unit.
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioOutputUnitProperty_CurrentDevice,
-      kAudioUnitScope_Global,
-      0,
-      &output_info_.id_,
-      sizeof(output_info_.id_));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::SetInputDeviceAsCurrent(
-    AudioDeviceID input_id) {
-  OSStatus result = noErr;
-
-  // Get the default input device if device is unknown.
-  if (input_id == kAudioDeviceUnknown) {
-    AudioObjectPropertyAddress pa;
-    pa.mSelector = kAudioHardwarePropertyDefaultInputDevice;
-    pa.mScope = kAudioObjectPropertyScopeGlobal;
-    pa.mElement = kAudioObjectPropertyElementMaster;
-    UInt32 size = sizeof(input_id);
-
-    result = AudioObjectGetPropertyData(
-        kAudioObjectSystemObject,
-        &pa,
-        0,
-        0,
-        &size,
-        &input_id);
-
-    OSSTATUS_DCHECK(result == noErr, result);
-    if (result != noErr)
-      return result;
-  }
-
-  // Set the render frame size.
-  UInt32 frame_size = hardware_buffer_size_;
-  AudioObjectPropertyAddress pa;
-  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
-  pa.mScope = kAudioDevicePropertyScopeInput;
-  pa.mElement = kAudioObjectPropertyElementMaster;
-  result = AudioObjectSetPropertyData(
-      input_id,
-      &pa,
-      0,
-      0,
-      sizeof(frame_size),
-      &frame_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  input_info_.Initialize(input_id, true);
-
-  // Set the Current Device to the AUHAL.
-  // This should be done only after I/O has been enabled on the AUHAL.
-  result = AudioUnitSetProperty(
-      input_unit_,
-      kAudioOutputUnitProperty_CurrentDevice,
-      kAudioUnitScope_Global,
-      0,
-      &input_info_.id_,
-      sizeof(input_info_.id_));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::CreateAudioUnits() {
-  // Q: Why do we need a varispeed unit?
-  // A: If the input device and the output device are running at
-  // different sample rates and/or on different clocks, we will need
-  // to compensate to avoid a pitch change and
-  // to avoid buffer under and over runs.
-  ComponentDescription varispeed_desc;
-  varispeed_desc.componentType = kAudioUnitType_FormatConverter;
-  varispeed_desc.componentSubType = kAudioUnitSubType_Varispeed;
-  varispeed_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  varispeed_desc.componentFlags = 0;
-  varispeed_desc.componentFlagsMask = 0;
-
-  Component varispeed_comp = FindNextComponent(NULL, &varispeed_desc);
-  if (varispeed_comp == NULL)
-    return -1;
-
-  OSStatus result = OpenAComponent(varispeed_comp, &varispeed_unit_);
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Open input AudioUnit.
-  ComponentDescription input_desc;
-  input_desc.componentType = kAudioUnitType_Output;
-  input_desc.componentSubType = kAudioUnitSubType_HALOutput;
-  input_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  input_desc.componentFlags = 0;
-  input_desc.componentFlagsMask = 0;
-
-  Component input_comp = FindNextComponent(NULL, &input_desc);
-  if (input_comp == NULL)
-    return -1;
-
-  result = OpenAComponent(input_comp, &input_unit_);
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Open output AudioUnit.
-  ComponentDescription output_desc;
-  output_desc.componentType = kAudioUnitType_Output;
-  output_desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-  output_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  output_desc.componentFlags = 0;
-  output_desc.componentFlagsMask = 0;
-
-  Component output_comp = FindNextComponent(NULL, &output_desc);
-  if (output_comp == NULL)
-    return -1;
-
-  result = OpenAComponent(output_comp, &output_unit_);
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  return noErr;
-}
-
-OSStatus AudioSynchronizedStream::SetupInput(AudioDeviceID input_id) {
-  // The AUHAL used for input needs to be initialized
-  // before anything is done to it.
-  OSStatus result = AudioUnitInitialize(input_unit_);
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // We must enable the Audio Unit (AUHAL) for input and disable output
-  // BEFORE setting the AUHAL's current device.
-  result = EnableIO();
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  result = SetInputDeviceAsCurrent(input_id);
-  OSSTATUS_DCHECK(result == noErr, result);
-
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::EnableIO() {
-  // Enable input on the AUHAL.
-  UInt32 enable_io = 1;
-  OSStatus result = AudioUnitSetProperty(
-      input_unit_,
-      kAudioOutputUnitProperty_EnableIO,
-      kAudioUnitScope_Input,
-      1,  // input element
-      &enable_io,
-      sizeof(enable_io));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Disable Output on the AUHAL.
-  enable_io = 0;
-  result = AudioUnitSetProperty(
-      input_unit_,
-      kAudioOutputUnitProperty_EnableIO,
-      kAudioUnitScope_Output,
-      0,  // output element
-      &enable_io,
-      sizeof(enable_io));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::SetupOutput(AudioDeviceID output_id) {
-  OSStatus result = noErr;
-
-  result = SetOutputDeviceAsCurrent(output_id);
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Tell the output unit not to reset timestamps.
-  // Otherwise sample rate changes will cause sync loss.
-  UInt32 start_at_zero = 0;
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioOutputUnitProperty_StartTimestampsAtZero,
-      kAudioUnitScope_Global,
-      0,
-      &start_at_zero,
-      sizeof(start_at_zero));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::SetupCallbacks() {
-  // Set the input callback.
-  AURenderCallbackStruct callback;
-  callback.inputProc = InputProc;
-  callback.inputProcRefCon = this;
-  OSStatus result = AudioUnitSetProperty(
-      input_unit_,
-      kAudioOutputUnitProperty_SetInputCallback,
-      kAudioUnitScope_Global,
-      0,
-      &callback,
-      sizeof(callback));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Set the output callback.
-  callback.inputProc = OutputProc;
-  callback.inputProcRefCon = this;
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioUnitProperty_SetRenderCallback,
-      kAudioUnitScope_Input,
-      0,
-      &callback,
-      sizeof(callback));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Set the varispeed callback.
-  callback.inputProc = VarispeedProc;
-  callback.inputProcRefCon = this;
-  result = AudioUnitSetProperty(
-      varispeed_unit_,
-      kAudioUnitProperty_SetRenderCallback,
-      kAudioUnitScope_Input,
-      0,
-      &callback,
-      sizeof(callback));
-
-  OSSTATUS_DCHECK(result == noErr, result);
-
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::SetupStreamFormats() {
-  AudioStreamBasicDescription asbd, asbd_dev1_in, asbd_dev2_out;
-
-  // Get the Stream Format (Output client side).
-  UInt32 property_size = sizeof(asbd_dev1_in);
-  OSStatus result = AudioUnitGetProperty(
-      input_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Input,
-      1,
-      &asbd_dev1_in,
-      &property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Get the Stream Format (client side).
-  property_size = sizeof(asbd);
-  result = AudioUnitGetProperty(
-      input_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Output,
-      1,
-      &asbd,
-      &property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Get the Stream Format (Output client side).
-  property_size = sizeof(asbd_dev2_out);
-  result = AudioUnitGetProperty(
-      output_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Output,
-      0,
-      &asbd_dev2_out,
-      &property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Set the format of all the AUs to the input/output devices channel count.
-  // For a simple case, you want to set this to
-  // the lower of count of the channels in the input device vs output device.
-  asbd.mChannelsPerFrame = std::min(asbd_dev1_in.mChannelsPerFrame,
-                                    asbd_dev2_out.mChannelsPerFrame);
-
-  // We must get the sample rate of the input device and set it to the
-  // stream format of AUHAL.
-  Float64 rate = 0;
-  property_size = sizeof(rate);
-
-  AudioObjectPropertyAddress pa;
-  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
-  pa.mScope = kAudioObjectPropertyScopeWildcard;
-  pa.mElement = kAudioObjectPropertyElementMaster;
-  result = AudioObjectGetPropertyData(
-      input_info_.id_,
-      &pa,
-      0,
-      0,
-      &property_size,
-      &rate);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  input_sample_rate_ = rate;
-
-  asbd.mSampleRate = rate;
-  property_size = sizeof(asbd);
-
-  // Set the new formats to the AUs...
-  result = AudioUnitSetProperty(
-      input_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Output,
-      1,
-      &asbd,
-      property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  result = AudioUnitSetProperty(
-      varispeed_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Input,
-      0,
-      &asbd,
-      property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Set the correct sample rate for the output device,
-  // but keep the channel count the same.
-  property_size = sizeof(rate);
-
-  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
-  pa.mScope = kAudioObjectPropertyScopeWildcard;
-  pa.mElement = kAudioObjectPropertyElementMaster;
-  result = AudioObjectGetPropertyData(
-      output_info_.id_,
-      &pa,
-      0,
-      0,
-      &property_size,
-      &rate);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  output_sample_rate_ = rate;
-
-  // The requested sample-rate must match the hardware sample-rate.
-  if (output_sample_rate_ != params_.sample_rate()) {
-    LOG(ERROR) << "Requested sample-rate: " << params_.sample_rate()
-        <<  " must match the hardware sample-rate: " << output_sample_rate_;
-    return kAudioDeviceUnsupportedFormatError;
-  }
-
-  asbd.mSampleRate = rate;
-  property_size = sizeof(asbd);
-
-  // Set the new audio stream formats for the rest of the AUs...
-  result = AudioUnitSetProperty(
-      varispeed_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Output,
-      0,
-      &asbd,
-      property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Input,
-      0,
-      &asbd,
-      property_size);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  return result;
-}
-
-void AudioSynchronizedStream::AllocateInputData() {
-  // Get the native number of input channels that the hardware supports.
-  int hardware_channels = 0;
-  bool got_hardware_channels = AudioManagerMac::GetDeviceChannels(
-      input_id_, kAudioDevicePropertyScopeInput, &hardware_channels);
-  if (!got_hardware_channels || hardware_channels > 2) {
-    // Only mono and stereo are supported on the input side. When it fails to
-    // get the native channel number or the native channel number is bigger
-    // than 2, we open the device in stereo mode.
-    hardware_channels = 2;
-  }
-
-  // Allocate storage for the AudioBufferList used for the
-  // input data from the input AudioUnit.
-  // We allocate enough space for with one AudioBuffer per channel.
-  size_t malloc_size = offsetof(AudioBufferList, mBuffers[0]) +
-      (sizeof(AudioBuffer) * hardware_channels);
-
-  input_buffer_list_ = static_cast<AudioBufferList*>(malloc(malloc_size));
-  input_buffer_list_->mNumberBuffers = hardware_channels;
-
-  input_bus_ = AudioBus::Create(hardware_channels, hardware_buffer_size_);
-  wrapper_bus_ = AudioBus::CreateWrapper(channels_);
-  if (hardware_channels != params_.input_channels()) {
-    ChannelLayout hardware_channel_layout =
-        GuessChannelLayout(hardware_channels);
-    ChannelLayout requested_channel_layout =
-        GuessChannelLayout(params_.input_channels());
-    channel_mixer_.reset(new ChannelMixer(hardware_channel_layout,
-                                          requested_channel_layout));
-    mixer_bus_ = AudioBus::Create(params_.input_channels(),
-                                  hardware_buffer_size_);
-  }
-
-  // Allocate buffers for AudioBufferList.
-  UInt32 buffer_size_bytes = input_bus_->frames() * sizeof(Float32);
-  for (size_t i = 0; i < input_buffer_list_->mNumberBuffers; ++i) {
-    input_buffer_list_->mBuffers[i].mNumberChannels = 1;
-    input_buffer_list_->mBuffers[i].mDataByteSize = buffer_size_bytes;
-    input_buffer_list_->mBuffers[i].mData = input_bus_->channel(i);
-  }
-}
-
-OSStatus AudioSynchronizedStream::HandleInputCallback(
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  TRACE_EVENT0("audio", "AudioSynchronizedStream::HandleInputCallback");
-
-  if (first_input_time_ < 0.0)
-    first_input_time_ = time_stamp->mSampleTime;
-
-  // Get the new audio input data.
-  OSStatus result = AudioUnitRender(
-      input_unit_,
-      io_action_flags,
-      time_stamp,
-      bus_number,
-      number_of_frames,
-      input_buffer_list_);
-
-  // TODO(xians): Add back the DCHECK after synchronize IO supports all
-  // combination of input and output params. See http://issue/246521.
-  if (result != noErr)
-    return result;
-
-  // Buffer input into FIFO.
-  int available_frames = fifo_.max_frames() - fifo_.frames();
-  if (input_bus_->frames() <= available_frames) {
-    if (channel_mixer_) {
-      channel_mixer_->Transform(input_bus_.get(), mixer_bus_.get());
-      fifo_.Push(mixer_bus_.get());
-    } else {
-      fifo_.Push(input_bus_.get());
-    }
-  }
-
-  return result;
-}
-
-OSStatus AudioSynchronizedStream::HandleVarispeedCallback(
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  // Create a wrapper bus on the AudioBufferList.
-  WrapBufferList(io_data, wrapper_bus_.get(), number_of_frames);
-
-  if (fifo_.frames() < static_cast<int>(number_of_frames)) {
-    // We don't DCHECK here, since this is a possible run-time condition
-    // if the machine is bogged down.
-    wrapper_bus_->Zero();
-    return noErr;
-  }
-
-  // Read from the FIFO to feed the varispeed.
-  fifo_.Consume(wrapper_bus_.get(), 0, number_of_frames);
-
-  return noErr;
-}
-
-OSStatus AudioSynchronizedStream::HandleOutputCallback(
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  // Input callback hasn't run yet or we've suddenly changed sample-rates
-  // -> silence.
-  if (first_input_time_ < 0.0 ||
-      static_cast<int>(number_of_frames) != params_.frames_per_buffer()) {
-    ZeroBufferList(io_data);
-    return noErr;
-  }
-
-  // Use the varispeed playback rate to offset small discrepancies
-  // in hardware clocks, and also any differences in sample-rate
-  // between input and output devices.
-
-  // Calculate a varispeed rate scalar factor to compensate for drift between
-  // input and output.  We use the actual number of frames still in the FIFO
-  // compared with the ideal value of |target_fifo_frames_|.
-  int delta = fifo_.frames() - target_fifo_frames_;
-
-  // Average |delta| because it can jitter back/forth quite frequently
-  // by +/- the hardware buffer-size *if* the input and output callbacks are
-  // happening at almost exactly the same time.  Also, if the input and output
-  // sample-rates are different then |delta| will jitter quite a bit due to
-  // the rate conversion happening in the varispeed, plus the jittering of
-  // the callbacks.  The average value is what's important here.
-  average_delta_ += (delta - average_delta_) * 0.1;
-
-  // Compute a rate compensation which always attracts us back to the
-  // |target_fifo_frames_| over a period of kCorrectionTimeSeconds.
-  const double kCorrectionTimeSeconds = 0.1;
-  double correction_time_frames = kCorrectionTimeSeconds * output_sample_rate_;
-  fifo_rate_compensation_ =
-      (correction_time_frames + average_delta_) / correction_time_frames;
-
-  // Adjust for FIFO drift.
-  OSStatus result = AudioUnitSetParameter(
-      varispeed_unit_,
-      kVarispeedParam_PlaybackRate,
-      kAudioUnitScope_Global,
-      0,
-      fifo_rate_compensation_,
-      0);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Render to the output using the varispeed.
-  result = AudioUnitRender(
-      varispeed_unit_,
-      io_action_flags,
-      time_stamp,
-      0,
-      number_of_frames,
-      io_data);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-  if (result != noErr)
-    return result;
-
-  // Create a wrapper bus on the AudioBufferList.
-  WrapBufferList(io_data, wrapper_bus_.get(), number_of_frames);
-
-  // Process in-place!
-  source_->OnMoreIOData(wrapper_bus_.get(),
-                        wrapper_bus_.get(),
-                        AudioBuffersState(0, 0));
-
-  return noErr;
-}
-
-OSStatus AudioSynchronizedStream::InputProc(
-    void* user_data,
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  AudioSynchronizedStream* stream =
-      static_cast<AudioSynchronizedStream*>(user_data);
-  DCHECK(stream);
-
-  return stream->HandleInputCallback(
-      io_action_flags,
-      time_stamp,
-      bus_number,
-      number_of_frames,
-      io_data);
-}
-
-OSStatus AudioSynchronizedStream::VarispeedProc(
-    void* user_data,
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  AudioSynchronizedStream* stream =
-      static_cast<AudioSynchronizedStream*>(user_data);
-  DCHECK(stream);
-
-  return stream->HandleVarispeedCallback(
-      io_action_flags,
-      time_stamp,
-      bus_number,
-      number_of_frames,
-      io_data);
-}
-
-OSStatus AudioSynchronizedStream::OutputProc(
-    void* user_data,
-    AudioUnitRenderActionFlags* io_action_flags,
-    const AudioTimeStamp* time_stamp,
-    UInt32 bus_number,
-    UInt32 number_of_frames,
-    AudioBufferList* io_data) {
-  AudioSynchronizedStream* stream =
-      static_cast<AudioSynchronizedStream*>(user_data);
-  DCHECK(stream);
-
-  return stream->HandleOutputCallback(
-      io_action_flags,
-      time_stamp,
-      bus_number,
-      number_of_frames,
-      io_data);
-}
-
-void AudioSynchronizedStream::AudioDeviceInfo::Initialize(
-    AudioDeviceID id, bool is_input) {
-  id_ = id;
-  is_input_ = is_input;
-  if (id_ == kAudioDeviceUnknown)
-    return;
-
-  UInt32 property_size = sizeof(buffer_size_frames_);
-
-  AudioObjectPropertyAddress pa;
-  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
-  pa.mScope = kAudioObjectPropertyScopeWildcard;
-  pa.mElement = kAudioObjectPropertyElementMaster;
-  OSStatus result = AudioObjectGetPropertyData(
-      id_,
-      &pa,
-      0,
-      0,
-      &property_size,
-      &buffer_size_frames_);
-
-  OSSTATUS_DCHECK(result == noErr, result);
-}
-
-}  // namespace media