Reland 153623004: Remove the unified IO code on the browser

media/audio/mac/audio_unified_mac.cc

Index: media/audio/mac/audio_unified_mac.cc
diff --git a/media/audio/mac/audio_unified_mac.cc b/media/audio/mac/audio_unified_mac.cc
deleted file mode 100644
index d1dc007e6a847ca86d97d8921c53b676903138e6..0000000000000000000000000000000000000000
--- a/media/audio/mac/audio_unified_mac.cc
+++ /dev/null
@@ -1,397 +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_unified_mac.h"
-
-#include <CoreServices/CoreServices.h>
-
-#include "base/basictypes.h"
-#include "base/logging.h"
-#include "base/mac/mac_logging.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.
-static const int 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: " << format_.mSampleRate
-        <<  " must match the 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_ptr<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_ptr<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;
-
-  if (device_ != kAudioObjectUnknown) {
-    OSStatus result = AudioDeviceStop(device_, io_proc_id_);
-    OSSTATUS_DCHECK(result == noErr, result);
-  }
-
-  is_playing_ = false;
-  source_ = NULL;
-}
-
-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(
-    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 (int channel_index = 0;
-       channel_index < static_cast<int>(format_.mChannelsPerFrame);
-       ++channel_index) {
-    float* dest;
-
-    int dest_channel_index = channel_index;
-
-    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;
-
-  return audio_output->Render(
-      device,
-      now,
-      input_data,
-      input_time,
-      output_data,
-      output_time);
-}
-
-}  // namespace media