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

media/audio/mac/audio_unified_mac.cc

Index: media/audio/mac/audio_unified_mac.cc
===================================================================
--- media/audio/mac/audio_unified_mac.cc	(revision 0)
+++ media/audio/mac/audio_unified_mac.cc	(revision 0)
@@ -0,0 +1,356 @@
+// 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_unified_mac.h"
+
+#include <CoreServices/CoreServices.h>
+
+#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"
+
+// !!!! For testing only.
+// Set to 4 to test headphone output on Stanton FinalScratch
+#define DEST_CHANNEL_OFFSET 0

[scherkus (not reviewing), 2012/09/10 11:26:19]

can we remove this or is there some other way to accomplish the same testing w/o
checking in test code like this?

[Chris Rogers, 2012/09/10 22:19:06]

Yes, this was only for testing -- removed!

+
+// TODO(crogers): support more than hard-coded stereo input.
+// Ideally we would like to receive this value as a constructor argument.
+const int kDefaultInputChannels = 2;

[scherkus (not reviewing), 2012/09/10 11:26:19]

static

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+namespace media {
+
+AudioHardwareUnifiedStream::AudioHardwareUnifiedStream(
+    AudioManagerMac* manager, const AudioParameters& params)
+    : manager_(manager),
+      source_(NULL),
+      volume_(1),

[no longer working on chromium, 2012/09/10 12:14:59]

nit, 1.0f?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+      input_channels_(0),
+      output_channels_(0),
+      input_channels_per_frame_(0),
+      output_channels_per_frame_(0),
+      io_proc_id_(0),
+      device_(kAudioObjectUnknown),
+      is_playing_(false) {
+  // We must have a manager.

[scherkus (not reviewing), 2012/09/10 11:26:19]

nit: remove comment (not adding any value)

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  DCHECK(manager_);
+  // A frame is one sample across all channels. In interleaved audio the per

[scherkus (not reviewing), 2012/09/10 11:26:19]

nit: blank lines before new // comment blocks

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  // frame fields identify the set of n |channels|. In uncompressed audio, a
+  // packet is always one frame.
+  format_.mSampleRate = params.sample_rate();
+  format_.mFormatID = kAudioFormatLinearPCM;
+  format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
+                         kLinearPCMFormatFlagIsSignedInteger;
+  format_.mBitsPerChannel = params.bits_per_sample();
+  format_.mChannelsPerFrame = params.channels();
+  format_.mFramesPerPacket = 1;
+  format_.mBytesPerPacket = (format_.mBitsPerChannel * params.channels()) / 8;
+  format_.mBytesPerFrame = format_.mBytesPerPacket;
+  format_.mReserved = 0;
+
+  // Calculate the number of sample frames per callback.
+  number_of_frames_ = params.GetBytesPerBuffer() / format_.mBytesPerPacket;
+
+  client_input_channels_ = kDefaultInputChannels;

[scherkus (not reviewing), 2012/09/10 11:26:19]

looks like we can move this into ctor initializer list

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  input_bus_ = AudioBus::Create(client_input_channels_,

[scherkus (not reviewing), 2012/09/10 11:26:19]

move client_input_channels_ to next line w/ 4 space indent OR align next line w/
client_input_channels_

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+      params.frames_per_buffer());
+  output_bus_ = AudioBus::Create(params);
+}
+
+AudioHardwareUnifiedStream::~AudioHardwareUnifiedStream() {
+    Stop();

[scherkus (not reviewing), 2012/09/10 11:26:19]

fix indent (should be 2 spaces)

[no longer working on chromium, 2012/09/10 12:14:59]

It looks like a programming error if the clients do not stop/close the stream
before deleting it. Remove the Stop() and add a DCHECK on device_.

[Chris Rogers, 2012/09/10 22:19:06]

Removed, and added DCHECK

+}
+
+bool AudioHardwareUnifiedStream::Open() {
+  device_ = kAudioDeviceUnknown;

[no longer working on chromium, 2012/09/10 12:14:59]

nit, device_ has been initialized in the constructor, use a DCHECK_EQ(device_,
kAudioDeviceUnknown) instead

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  // Obtain the current output device selected by the user.
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  pa.mScope = kAudioObjectPropertyScopeGlobal;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  UInt32 size = sizeof(device_);
+
+  OSStatus result = AudioObjectGetPropertyData(
+      kAudioObjectSystemObject,
+      &pa,
+      0,
+      0,
+      &size,
+      &device_);
+
+  if ((result != kAudioHardwareNoError) || (device_ == kAudioDeviceUnknown)) {
+    LOG(ERROR) << "Cannot open unified AudioDevice.";
+    return false;
+  }
+
+  // The requested sample-rate must match the hardware sample-rate.
+  Float64 sample_rate = 0.0;
+  size = sizeof(sample_rate);
+
+  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
+  pa.mScope = kAudioObjectPropertyScopeWildcard;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  result = AudioObjectGetPropertyData(
+      device_,
+      &pa,
+      0,
+      0,
+      &size,
+      &sample_rate);
+
+  if (result != noErr || sample_rate != format_.mSampleRate) {
+    LOG(ERROR) << "Requested sample-rate must match the hardware sample-rate.";

[scherkus (not reviewing), 2012/09/10 11:26:19]

nit: can you emit sample the two sample rates?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+    return false;
+  }
+
+  // Configure buffer frame size.
+  UInt32 frame_size = number_of_frames_;
+
+  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectSetPropertyData(device_, &pa, 0, 0,

[scherkus (not reviewing), 2012/09/10 11:26:19]

nit: fix indent style -- should either look like the other
AudioObjectGetProperyData() calls here OR align next line to match (

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+      sizeof(frame_size), &frame_size);
+

[no longer working on chromium, 2012/09/10 12:14:59]

add a DCHECK here to verify the result?

[Chris Rogers, 2012/09/10 22:19:06]

Fixed: I'm now logging an error here and returning early.

+  pa.mScope = kAudioDevicePropertyScopeOutput;
+  result = AudioObjectSetPropertyData(device_, &pa, 0, 0,

[scherkus (not reviewing), 2012/09/10 11:26:19]

ditto

[no longer working on chromium, 2012/09/10 12:14:59]

ditto

[Chris Rogers, 2012/09/10 22:19:06]

Done.

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+      sizeof(frame_size), &frame_size);
+
+  DVLOG(1) << "Sample rate: " << sample_rate;
+  DVLOG(1) << "Frame size: " << frame_size;
+
+  // Determine the number of input and output channels.
+  // We handle both the interleaved and non-interleaved cases.
+
+  // Get input stream configuration.
+  pa.mSelector = kAudioDevicePropertyStreamConfiguration;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  // Allocate storage.
+  result = AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);

[no longer working on chromium, 2012/09/10 12:14:59]

We are not sure if the machine has input device or not, check the result and
size before allocating memory,

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  scoped_array<uint8> input_list_storage(new uint8[size]);
+  AudioBufferList& input_list =
+      *reinterpret_cast<AudioBufferList*>(input_list_storage.get());
+
+  result = AudioObjectGetPropertyData(device_, &pa, 0, 0, &size, &input_list);

[no longer working on chromium, 2012/09/10 12:14:59]

DCHECK(result) ?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  // Determine number of input channels.
+  input_channels_per_frame_ = input_list.mNumberBuffers > 0 ?
+      input_list.mBuffers[0].mNumberChannels : 0;
+  if (input_channels_per_frame_ == 1 && input_list.mNumberBuffers > 1) {

[no longer working on chromium, 2012/09/10 12:14:59]

We don't need to check input_list.mNumberBuffers > 1 here.

[Chris Rogers, 2012/09/10 22:19:06]

I think it's better to be more explicit since this condition must be true

+    // Non-interleaved.
+    input_channels_ = input_list.mNumberBuffers;
+  } else {
+    // Interleaved.
+    input_channels_ = input_channels_per_frame_;
+  }
+
+  DVLOG(1) << "Input channels: " << input_channels_;
+  DVLOG(1) << "Input channels per frame: " << input_channels_per_frame_;
+
+  // The hardware must have at least the requested input channels.
+  if (client_input_channels_ > input_channels_) {
+    LOG(ERROR) << "AudioDevice does not support requested input channels.";
+    return false;
+  }
+
+  // Get output stream configuration.
+  pa.mSelector = kAudioDevicePropertyStreamConfiguration;
+  pa.mScope = kAudioDevicePropertyScopeOutput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  // Allocate storage.
+  AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);

[scherkus (not reviewing), 2012/09/10 11:26:19]

do we need to check the return value of these calls + ones below?

[no longer working on chromium, 2012/09/10 12:14:59]

DCHECK the size bigger than 0

[Chris Rogers, 2012/09/10 22:19:06]

Done.

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  scoped_array<uint8> output_list_storage(new uint8[size]);
+  AudioBufferList& output_list =
+      *reinterpret_cast<AudioBufferList*>(output_list_storage.get());
+
+  AudioObjectGetPropertyData(device_, &pa, 0, 0, &size, &output_list);
+
+  // Determine number of output channels.
+  output_channels_per_frame_ = output_list.mBuffers[0].mNumberChannels;
+  if (output_channels_per_frame_ == 1 && output_list.mNumberBuffers > 1) {

[no longer working on chromium, 2012/09/10 12:14:59]

no need to check output_list.mNumberBuffers > 1

[Chris Rogers, 2012/09/10 22:19:06]

I just want to be more explicit here.

+    // Non-interleaved.
+    output_channels_ = output_list.mNumberBuffers;
+  } else {
+    // Interleaved.
+    output_channels_ = output_channels_per_frame_;
+  }
+
+  DVLOG(1) << "Output channels: " << output_channels_;
+  DVLOG(1) << "Output channels per frame: " << output_channels_per_frame_;
+
+  // The hardware must have at least the requested output channels.
+  if (output_channels_ < format_.mChannelsPerFrame) {
+    LOG(ERROR) << "AudioDevice does not support requested output channels.";
+    return false;
+  }
+
+  // Setup the I/O proc.
+  result = AudioDeviceCreateIOProcID(device_, RenderProc, this, &io_proc_id_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error creating IOProc.";
+    return false;
+  }
+
+  return true;
+}
+
+void AudioHardwareUnifiedStream::Close() {
+  Stop();  // make sure to stop if the client forgot.

[scherkus (not reviewing), 2012/09/10 11:26:19]

Can we treat this as a programmer error / DCHECK()?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  AudioDeviceDestroyIOProcID(device_, io_proc_id_);

[no longer working on chromium, 2012/09/10 12:14:59]

set device_ to kAudioObjectUnknown

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  io_proc_id_ = 0;
+
+  // Inform the audio manager that we have been closed. This can cause our
+  // destruction.
+  manager_->ReleaseOutputStream(this);
+}
+
+void AudioHardwareUnifiedStream::Start(AudioSourceCallback* callback) {
+  DCHECK(callback);
+  DLOG_IF(ERROR, (device_ == kAudioObjectUnknown))
+      << "Open() has not been called successfully";

[scherkus (not reviewing), 2012/09/10 11:26:19]

DCHECK() instead?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  if (device_ == kAudioObjectUnknown || is_playing_)
+    return;
+
+  source_ = callback;
+
+  // // Allocate storage for our audio source.
+  // if (input_channels_)
+  //   input_data_.reset(new int16[number_of_frames_ * input_channels_]);

[scherkus (not reviewing), 2012/09/10 11:26:19]

???

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  // output_data_.reset(new int16[number_of_frames_ * output_channels_]);

[no longer working on chromium, 2012/09/10 12:14:59]

looks like debugging code, remove?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+  OSStatus result = AudioDeviceStart(device_, io_proc_id_);
+  if (result == noErr)
+    is_playing_ = true;
+}
+
+void AudioHardwareUnifiedStream::Stop() {
+  if (!is_playing_)
+    return;
+
+  source_ = NULL;
+
+  if (device_ != kAudioObjectUnknown)
+    AudioDeviceStop(device_, io_proc_id_);
+
+  is_playing_ = false;
+}
+
+void AudioHardwareUnifiedStream::SetVolume(double volume) {
+  volume_ = static_cast<float>(volume);
+  // TODO(crogers): set volume property
+}
+
+void AudioHardwareUnifiedStream::GetVolume(double* volume) {
+  *volume = volume_;
+}
+
+// Pulls on our provider with optional input, asking it to render output.
+// Note to future hackers of this function: Do not add locks here because this
+// is running on a real-time thread (for low-latency).
+OSStatus AudioHardwareUnifiedStream::Render(

[scherkus (not reviewing), 2012/09/10 11:26:19]

Should we have two Render() methods for interleaved vs non-interleaved?

There seems to be a lot of internal branching + TODOs re: memcpy() if source is
planar. I feel the minor code duplication will be worth the clarity for handling
the interleaved case vs non-interleaved case.

[Chris Rogers, 2012/09/10 22:19:06]

The problem is that the interleave vs. non-interleaved can happen on both the
input *and* output side, so there would need to be 2x2 == 4 separate methods.

I'll have another look to see if I can clean up more...

+    AudioDeviceID inDevice,

[scherkus (not reviewing), 2012/09/10 11:26:19]

unix_hacker

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+    const AudioTimeStamp* inNow,
+    const AudioBufferList* inInputData,
+    const AudioTimeStamp* inInputTime,
+    AudioBufferList* outOutputData,
+    const AudioTimeStamp* inOutputTime) {
+  // Convert the input data accounting for possible interleaving.
+  // TODO(crogers): its better to simply memcpy() if source is already planar.

[no longer working on chromium, 2012/09/10 12:14:59]

nit, it's or remove its

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+  if (input_channels_ >= client_input_channels_) {
+    for (unsigned channel_index = 0; channel_index < client_input_channels_;

[scherkus (not reviewing), 2012/09/10 11:26:19]

s/unsigned/int/

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+        ++channel_index) {

[scherkus (not reviewing), 2012/09/10 11:26:19]

indent one more space

[no longer working on chromium, 2012/09/10 12:14:59]

nit, fix the indentation.

[Chris Rogers, 2012/09/10 22:19:06]

Done.

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+      float* sourceP;

[no longer working on chromium, 2012/09/10 12:14:59]

source, or source_ptr?

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+      unsigned source_channel_index = channel_index;
+
+      if (input_channels_per_frame_ > 1) {
+        // Interleaved.
+        sourceP = static_cast<float*>(inInputData->mBuffers[0].mData) +

[scherkus (not reviewing), 2012/09/10 11:26:19]

unix_hacker

perhaps _ptr instead of P?

[Chris Rogers, 2012/09/10 22:19:06]

just simplified to |source|

+            source_channel_index;
+      } else {
+        // Non-interleaved.
+        sourceP = static_cast<float*>(
+            inInputData->mBuffers[source_channel_index].mData);
+      }
+
+      float* p = input_bus_->channel(channel_index);
+      for (unsigned i = 0; i < number_of_frames_; ++i) {
+        p[i] = *sourceP;
+        sourceP += input_channels_per_frame_;
+      }
+    }
+  } else if (input_channels_) {
+    input_bus_->Zero();
+  }
+
+  // Give the client optional input data and have it render the output data.
+  source_->OnMoreIOData(input_bus_.get(),
+                        output_bus_.get(),
+                        AudioBuffersState(0, 0));
+
+  // TODO(crogers): handle final Core Audio 5.1 layout for 5.1 audio.
+
+  // Handle interleaving as necessary.
+  // TODO(crogers): its better to simply memcpy() if dest is already planar.
+
+  for (unsigned channel_index = 0; channel_index < format_.mChannelsPerFrame;
+      ++channel_index) {

[scherkus (not reviewing), 2012/09/10 11:26:19]

indent one more space

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+    float* destP;

[no longer working on chromium, 2012/09/10 12:14:59]

dest or dest_ptr

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+
+    unsigned dest_channel_index = channel_index + DEST_CHANNEL_OFFSET;
+
+    if (output_channels_per_frame_ > 1) {
+      // Interleaved.
+      destP = static_cast<float*>(outOutputData->mBuffers[0].mData) +
+          dest_channel_index;
+    } else {
+      // Non-interleaved.
+      destP = static_cast<float*>(
+          outOutputData->mBuffers[dest_channel_index].mData);
+    }
+
+    float* p = output_bus_->channel(channel_index);
+    for (unsigned i = 0; i < number_of_frames_; ++i) {
+      *destP = p[i];
+      destP += output_channels_per_frame_;
+    }
+  }
+
+  return noErr;
+}
+
+OSStatus AudioHardwareUnifiedStream::RenderProc(
+    AudioDeviceID inDevice,

[scherkus (not reviewing), 2012/09/10 11:26:19]

unix_hacker

[Chris Rogers, 2012/09/10 22:19:06]

Done.

+    const AudioTimeStamp* inNow,
+    const AudioBufferList* inInputData,
+    const AudioTimeStamp* inInputTime,
+    AudioBufferList* outOutputData,
+    const AudioTimeStamp* inOutputTime,
+    void* user_data) {
+  AudioHardwareUnifiedStream* audio_output =
+      static_cast<AudioHardwareUnifiedStream*>(user_data);
+  DCHECK(audio_output);
+  if (!audio_output)
+    return -1;
+
+  return audio_output->Render(
+      inDevice,
+      inNow,
+      inInputData,
+      inInputTime,
+      outOutputData,
+      inOutputTime);
+}
+
+}  // namespace media