Add Mac OS X synchronized audio I/O back-end

media/audio/mac/audio_synchronized_mac.cc

Index: media/audio/mac/audio_synchronized_mac.cc
===================================================================
--- media/audio/mac/audio_synchronized_mac.cc	(revision 0)
+++ media/audio/mac/audio_synchronized_mac.cc	(revision 0)
@@ -0,0 +1,927 @@
+// 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/logging.h"
+#include "base/mac/mac_logging.h"
+#include "media/audio/audio_util.h"
+#include "media/audio/mac/audio_manager_mac.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.
+// TODO(crogers): refine this, taking into account different input/output
+// sample-rate combinations.
+static const int kTargetDelayFrames = 256;
+
+static void ZeroBufferList(AudioBufferList* io_data) {
+  for (UInt32 i = 0; i < io_data->mNumberBuffers; ++i)
+    memset(io_data->mBuffers[i].mData, 0, io_data->mBuffers[i].mDataByteSize);
+}
+
+AudioSynchronizedStream::AudioSynchronizedStream(
+    AudioManagerMac* manager,
+    const AudioParameters&,
+    AudioDeviceID input_id,
+    AudioDeviceID output_id)
+    : manager_(manager),
+      input_id_(input_id),
+      output_id_(output_id),
+      input_data_(NULL),
+      fifo_(kChannels, kFifoSize),
+      is_fifo_initialized_(false),
+      fifo_rate_compensation_(1.0),
+      output_sample_rate_(0),
+      input_unit_(0),
+      varispeed_unit_(0),
+      output_unit_(0),
+      first_input_time_(-1),
+      first_output_time_(-1),
+      in_to_out_sample_offset_(0),
+      is_running_(false),
+      hardware_buffer_size_(kHardwareBufferSize),
+      channels_(kChannels) {
+  // TODO(crogers): actually do something with |params|.
+  // We at least need to verify the sample-rate matches the hardware
+  // sample-rate of the output device, and we should take into account
+  // the |channels|.  For now we're limited to stereo output.
+}
+
+AudioSynchronizedStream::~AudioSynchronizedStream() {
+  DCHECK(!input_unit_);
+  DCHECK(!output_unit_);
+  DCHECK(!varispeed_unit_);
+}
+
+bool AudioSynchronizedStream::Open() {
+  // 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;
+  }
+
+  ComputeThruOffset();
+
+  return true;
+}
+
+void AudioSynchronizedStream::Close() {
+  DCHECK(!is_running_);
+
+  if (input_data_) {
+    for (UInt32 i = 0; i < input_data_->mNumberBuffers; ++i)
+      free(input_data_->mBuffers[i].mData);
+    free(input_data_);
+    input_data_ = 0;
+  }
+
+  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;
+
+  OSStatus result = noErr;
+
+  if (!is_running_) {
+    first_input_time_ = -1;
+    first_output_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 SetVolume()/GetVolume() from AudioOutputStream.

[scherkus (not reviewing), 2012/09/13 13:06:03]

nit: some of this comment can go on previous line

[Chris Rogers, 2012/09/15 00:06:08]

Done.

+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;
+
+  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;
+
+  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() {
+  // Allocate storage for the AudioBufferList used for the
+  // input data from the input AudioUnit.
+  // We allocate enough space for with one AudioBuffer per channel.
+  UInt32 malloc_size = offsetof(AudioBufferList, mBuffers[0]) +

[scherkus (not reviewing), 2012/09/13 13:06:03]

for the sake of other chromium developers can you attempt to use stddint types
(int, size_t, uint32, et al) as much as possible?

[Chris Rogers, 2012/09/15 00:06:08]

Done.

+      (sizeof(AudioBuffer) * channels_);
+
+  input_data_ = static_cast<AudioBufferList*>(malloc(malloc_size));

[scherkus (not reviewing), 2012/09/13 13:06:03]

any reason to prefer malloc/free over new/delete here + below?

[Chris Rogers, 2012/09/15 00:06:08]

For the AudioBufferList allocation, I'm using malloc since otherwise (I think)
I'd have to create two member variables (input_data_ and "input_data_storage_"
of type uint8*)

+  input_data_->mNumberBuffers = channels_;
+
+  // Allocate buffers for AudioBufferList.
+  UInt32 buffer_size_bytes = hardware_buffer_size_ * sizeof(Float32);
+  for (UInt32 i = 0; i < input_data_->mNumberBuffers; ++i) {
+    input_data_->mBuffers[i].mNumberChannels = 1;
+    input_data_->mBuffers[i].mDataByteSize = buffer_size_bytes;
+    input_data_->mBuffers[i].mData = malloc(buffer_size_bytes);

[Chris Rogers, 2012/09/15 00:06:08]

I've removed the malloc() here and am now using an AudioBus allocation
|input_bus_|

+  }
+}
+
+void AudioSynchronizedStream::ComputeThruOffset() {
+  // The initial latency will at least be the safety offsets
+  // of the devices + the buffer sizes.
+  in_to_out_sample_offset_ = SInt32(input_info_.buffer_size_frames_ +
+      output_info_.buffer_size_frames_);
+}
+
+OSStatus AudioSynchronizedStream::HandleInputCallback(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  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_data_);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Buffer input into FIFO.
+  if (is_fifo_initialized_) {
+    // TODO(crogers): remove this locking once AudioFifo becomes thread-safe.
+    if (fifo_lock_.Try()) {
+      AudioBus bus(channels_, number_of_frames, input_data_);
+      fifo_.Push(&bus);
+      fifo_lock_.Release();
+    }
+  }
+
+  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.
+  AudioBus bus(channels_, number_of_frames, io_data);
+
+  if (fifo_.frames() < static_cast<int>(number_of_frames)) {

[scherkus (not reviewing), 2012/09/13 13:06:03]

FYI: fifo_ not locked

See my comment on line 817 regarding which threads these various procs are
running on 

[Chris Rogers, 2012/09/15 00:06:08]

I've completely removed this fifo_lock_ and have actually made AudioFifo
lock-free / thread-safe.  As I mentioned, having to lock in a real-time thread
can and does cause glitching.  I was hitting it quite a bit in my tuning on
Thursday, so realized that we really need to do this properly.  In other words,
no need to lock anymore :)

+    // We don't DCHECK here, since this is a possible run-time condition
+    // if the machine is bogged down.
+    bus.Zero();
+    return noErr;
+  }
+
+  // TODO(crogers): remove this locking once AudioFifo becomes thread-safe.
+  if (fifo_lock_.Try()) {
+    // Read from the FIFO to feed the varispeed.
+    fifo_.Consume(&bus, 0, number_of_frames);
+    fifo_lock_.Release();
+  }
+
+  // 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 kTargetDelayFrames.
+  size_t n = fifo_.frames();

[scherkus (not reviewing), 2012/09/13 13:06:03]

not locked

[Chris Rogers, 2012/09/15 00:06:08]

Fixed in AudioFifo thread-safety

+  int delta = n - kTargetDelayFrames;
+  double sample_rate = output_sample_rate_;
+  double x = (sample_rate + delta) / sample_rate;
+
+  fifo_rate_compensation_ = x;
+
+  return noErr;
+}
+
+OSStatus AudioSynchronizedStream::HandleOutputCallback(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  if (first_input_time_ < 0.0) {
+    // Input callback hasn't run yet -> silence.
+    ZeroBufferList(io_data);
+    return noErr;
+  }
+
+  AudioTimeStamp input_ts;
+  OSStatus result = AudioDeviceGetCurrentTime(input_info_.id_, &input_ts);
+
+  if (result != noErr) {
+    ZeroBufferList(io_data);
+    return noErr;
+  }
+
+  AudioTimeStamp output_ts;
+  result = AudioDeviceGetCurrentTime(output_info_.id_, &output_ts);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Use the varispeed playback rate to offset small discrepancies
+  // in hardware clocks, and also any differences in sample-rate
+  // between input and output devices.
+
+  // Adjust for rate scalars of the input and output devices.
+  double rate = input_ts.mRateScalar / output_ts.mRateScalar;
+
+  // Adjust for FIFO drift.
+  rate *= fifo_rate_compensation_;
+
+  result = AudioUnitSetParameter(
+      varispeed_unit_,
+      kVarispeedParam_PlaybackRate,
+      kAudioUnitScope_Global,
+      0,
+      rate,
+      0);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Get the delta between the devices and add it to the offset.
+  if (first_output_time_ < 0.) {

[scherkus (not reviewing), 2012/09/13 13:06:03]

0.0

[Chris Rogers, 2012/09/15 00:06:08]

This code has now been removed.

+    first_output_time_ = time_stamp->mSampleTime;
+    ComputeThruOffset();
+
+    // Buffer initial silence corresponding to I/O delay.
+    unsigned n = static_cast<unsigned>(in_to_out_sample_offset_);

[scherkus (not reviewing), 2012/09/13 13:06:03]

s/unsigned/int

[Chris Rogers, 2012/09/15 00:06:08]

Removed/simplified

+    AudioBus silence(channels_, n);
+    silence.Zero();
+    fifo_.Push(&silence);

[scherkus (not reviewing), 2012/09/13 13:06:03]

FYI: not locked

to confirm.. the output+varispeed proc run together on one thread and the input
proc run on a separate thread?

for example it appears you set is_fifo_initialized_ in the output proc but read
it from the input proc -- should it also be protected?

do we want to be protecting every access to fifo_ or only certain calls?

[Chris Rogers, 2012/09/15 00:06:08]

The AudioFifo now has changes to make it lock-free thread-safe in a limited
sense where one thread calls Push() and a second thread calls Consume()

The AudioFifo::frames() method will always return an amount which is or was very
recently valid.  The "very recently" valid may sound worrisome, but it turns out
that it will either be a value larger than it really is for the Push() thread,
or a value smaller than it really is for the Consume() thread, so is always
safe.

+    is_fifo_initialized_ = true;
+
+    ZeroBufferList(io_data);
+    return noErr;
+  }
+
+  // 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.
+  AudioBus bus(channels_, number_of_frames, io_data);
+
+  // Process in-place!
+  source_->OnMoreIOData(&bus, &bus, 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