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

media/audio/mac/audio_unified_mac.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/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"
+
+namespace media {
+
+// TODO(crogers): support more than hard-coded stereo input.
+// Ideally we would like to receive this value as a constructor argument.
+int AudioHardwareUnifiedStream::kDefaultInputChannels = 2;
+
+AudioHardwareUnifiedStream::AudioHardwareUnifiedStream(
+    AudioManagerMac* manager, const AudioParameters& params)
+    : manager_(manager),
+      source_(NULL),
+      client_input_channels_(kDefaultInputChannels),
+      volume_(1.0f),
+      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) {
+  DCHECK(manager_);
+
+  // A frame is one sample across all channels. In interleaved audio the per
+  // 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;
+
+  input_bus_ = AudioBus::Create(client_input_channels_,
+                                params.frames_per_buffer());
+  output_bus_ = AudioBus::Create(params);
+}
+
+AudioHardwareUnifiedStream::~AudioHardwareUnifiedStream() {
+  DCHECK_EQ(device_, kAudioObjectUnknown);
+}
+
+bool AudioHardwareUnifiedStream::Open() {
+  // 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.";

[no longer working on chromium, 2012/09/11 15:40:28]

nit, we can log only one time:
<< "Requested sample-rate: " << format_.mSampleRate
<<  " must match the hardware sample-rate: " << sample_rate;

[Chris Rogers, 2012/09/14 19:17:59]

Done.

+    LOG(ERROR) << "Requested sample-rate: " << format_.mSampleRate;
+    LOG(ERROR) << "Hardware sample-rate: " << sample_rate;
+    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,
+      sizeof(frame_size),
+      &frame_size);
+
+  if (result != noErr) {
+    LOG(ERROR) << "Unable to set input buffer frame size: "  << frame_size;
+    return false;
+  }
+
+  pa.mScope = kAudioDevicePropertyScopeOutput;
+  result = AudioObjectSetPropertyData(
+      device_,
+      &pa,
+      0,
+      0,
+      sizeof(frame_size),
+      &frame_size);
+
+  if (result != noErr) {
+    LOG(ERROR) << "Unable to set output buffer frame size: "  << frame_size;
+    return false;
+  }
+
+  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;
+
+  result = AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr && size > 0) {
+    // Allocate storage.
+    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);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      // 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) {
+        // 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 (result != noErr || 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;
+
+  result = AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr && size > 0) {
+    // Allocate storage.
+    scoped_array<uint8> output_list_storage(new uint8[size]);
+    AudioBufferList& output_list =
+        *reinterpret_cast<AudioBufferList*>(output_list_storage.get());
+
+    result = AudioObjectGetPropertyData(
+        device_,
+        &pa,
+        0,
+        0,
+        &size,
+        &output_list);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      // Determine number of output channels.
+      output_channels_per_frame_ = output_list.mBuffers[0].mNumberChannels;
+      if (output_channels_per_frame_ == 1 && output_list.mNumberBuffers > 1) {
+        // 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 (result != noErr ||
+      output_channels_ < static_cast<int>(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() {
+  DCHECK(!is_playing_);
+
+  OSStatus result = AudioDeviceDestroyIOProcID(device_, io_proc_id_);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  io_proc_id_ = 0;
+  device_ = kAudioObjectUnknown;
+
+  // Inform the audio manager that we have been closed. This can cause our
+  // destruction.
+  manager_->ReleaseOutputStream(this);
+}
+
+void AudioHardwareUnifiedStream::Start(AudioSourceCallback* callback) {
+  DCHECK(callback);
+  DCHECK_NE(device_, kAudioObjectUnknown);
+  DCHECK(!is_playing_);
+  if (device_ == kAudioObjectUnknown || is_playing_)
+    return;
+
+  source_ = callback;
+
+  OSStatus result = AudioDeviceStart(device_, io_proc_id_);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr)
+    is_playing_ = true;
+}
+
+void AudioHardwareUnifiedStream::Stop() {
+  if (!is_playing_)
+    return;
+
+  source_ = NULL;
+
+  if (device_ != kAudioObjectUnknown) {
+    OSStatus result = AudioDeviceStop(device_, io_proc_id_);

[no longer working on chromium, 2012/09/11 15:40:28]

I think you will have a compiling warning about "Unused variable result" in
release build here.

+    OSSTATUS_DCHECK(result == noErr, result);
+  }
+
+  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(

[no longer working on chromium, 2012/09/11 15:40:28]

This function will only return noErr, how do you thinking making it void, and
RenderProc return noErr after calling Render().
I am actually fine with the current solution, so ignore this comment if you
prefer the current approach.

[Chris Rogers, 2012/09/14 19:17:59]

It's probably fine either way, but leaving it because we may modify the code to
return an actual error and to be consistent with similar code in
AUAudioOutputStream and AudioSynchronizedStream.

+    AudioDeviceID device,
+    const AudioTimeStamp* now,
+    const AudioBufferList* input_data,
+    const AudioTimeStamp* input_time,
+    AudioBufferList* output_data,
+    const AudioTimeStamp* output_time) {
+  // Convert the input data accounting for possible interleaving.
+  // TODO(crogers): it's better to simply memcpy() if source is already planar.
+  if (input_channels_ >= client_input_channels_) {
+    for (int channel_index = 0; channel_index < client_input_channels_;
+         ++channel_index) {
+      float* source;
+
+      int source_channel_index = channel_index;
+
+      if (input_channels_per_frame_ > 1) {
+        // Interleaved.
+        source = static_cast<float*>(input_data->mBuffers[0].mData) +
+            source_channel_index;
+      } else {
+        // Non-interleaved.
+        source = static_cast<float*>(
+            input_data->mBuffers[source_channel_index].mData);
+      }
+
+      float* p = input_bus_->channel(channel_index);
+      for (int i = 0; i < number_of_frames_; ++i) {
+        p[i] = *source;
+        source += 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): it's better to simply memcpy() if dest is already planar.
+
+  for (unsigned channel_index = 0; channel_index < format_.mChannelsPerFrame;

[scherkus (not reviewing), 2012/09/11 12:22:25]

s/unsigned/int/

[Chris Rogers, 2012/09/14 19:17:59]

Done.

+       ++channel_index) {
+    float* dest;
+
+    unsigned dest_channel_index = channel_index;

[scherkus (not reviewing), 2012/09/11 12:22:25]

s/unsigned/int/

[Chris Rogers, 2012/09/14 19:17:59]

Done.

+
+    if (output_channels_per_frame_ > 1) {
+      // Interleaved.
+      dest = static_cast<float*>(output_data->mBuffers[0].mData) +
+          dest_channel_index;
+    } else {
+      // Non-interleaved.
+      dest = static_cast<float*>(
+          output_data->mBuffers[dest_channel_index].mData);
+    }
+
+    float* p = output_bus_->channel(channel_index);
+    for (int i = 0; i < number_of_frames_; ++i) {
+      *dest = p[i];
+      dest += output_channels_per_frame_;
+    }
+  }
+
+  return noErr;
+}
+
+OSStatus AudioHardwareUnifiedStream::RenderProc(
+    AudioDeviceID device,
+    const AudioTimeStamp* now,
+    const AudioBufferList* input_data,
+    const AudioTimeStamp* input_time,
+    AudioBufferList* output_data,
+    const AudioTimeStamp* output_time,
+    void* user_data) {
+  AudioHardwareUnifiedStream* audio_output =
+      static_cast<AudioHardwareUnifiedStream*>(user_data);
+  DCHECK(audio_output);
+  if (!audio_output)
+    return -1;

[scherkus (not reviewing), 2012/09/11 12:22:25]

q: is there a better return code to return here or does it not matter?

[Chris Rogers, 2012/09/14 19:17:59]

This is really an anomaly which should never happen in any real run-time
situation.  I don't think there's any reasonable CoreAudio error code here, and
this value will just be ignored.

+
+  return audio_output->Render(
+      device,
+      now,
+      input_data,
+      input_time,
+      output_data,
+      output_time);
+}
+
+}  // namespace media