Remove deprecated non-AUHAL audio path on OSX.

media/audio/mac/audio_low_latency_output_mac.cc

Index: media/audio/mac/audio_low_latency_output_mac.cc
diff --git a/media/audio/mac/audio_low_latency_output_mac.cc b/media/audio/mac/audio_low_latency_output_mac.cc
deleted file mode 100644
index afa480aefb956e44b219e40ba83efe3d2db5a1de..0000000000000000000000000000000000000000
--- a/media/audio/mac/audio_low_latency_output_mac.cc
+++ /dev/null
@@ -1,416 +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_low_latency_output_mac.h"
-
-#include <CoreServices/CoreServices.h>
-
-#include "base/basictypes.h"
-#include "base/command_line.h"
-#include "base/logging.h"
-#include "base/mac/mac_logging.h"
-#include "media/audio/mac/audio_manager_mac.h"
-#include "media/base/media_switches.h"
-
-namespace media {
-
-static std::ostream& operator<<(std::ostream& os,
-                                const AudioStreamBasicDescription& format) {
-  os << "sample rate       : " << format.mSampleRate << std::endl
-     << "format ID         : " << format.mFormatID << std::endl
-     << "format flags      : " << format.mFormatFlags << std::endl
-     << "bytes per packet  : " << format.mBytesPerPacket << std::endl
-     << "frames per packet : " << format.mFramesPerPacket << std::endl
-     << "bytes per frame   : " << format.mBytesPerFrame << std::endl
-     << "channels per frame: " << format.mChannelsPerFrame << std::endl
-     << "bits per channel  : " << format.mBitsPerChannel;
-  return os;
-}
-
-static AudioObjectPropertyAddress kDefaultOutputDeviceAddress = {
-  kAudioHardwarePropertyDefaultOutputDevice,
-  kAudioObjectPropertyScopeGlobal,
-  kAudioObjectPropertyElementMaster
-};
-
-// Overview of operation:
-// 1) An object of AUAudioOutputStream is created by the AudioManager
-// factory: audio_man->MakeAudioStream().
-// 2) Next some thread will call Open(), at that point the underlying
-// default output Audio Unit is created and configured.
-// 3) Then some thread will call Start(source).
-// Then the Audio Unit is started which creates its own thread which
-// periodically will call the source for more data as buffers are being
-// consumed.
-// 4) At some point some thread will call Stop(), which we handle by directly
-// stopping the default output Audio Unit.
-// 6) The same thread that called stop will call Close() where we cleanup
-// and notify the audio manager, which likely will destroy this object.
-
-AUAudioOutputStream::AUAudioOutputStream(
-    AudioManagerMac* manager, const AudioParameters& params)
-    : manager_(manager),
-      source_(NULL),
-      output_unit_(0),
-      output_device_id_(kAudioObjectUnknown),
-      volume_(1),
-      hardware_latency_frames_(0),
-      stopped_(false),
-      audio_bus_(AudioBus::Create(params)) {
-  // We must have a manager.
-  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;
-
-  DVLOG(1) << "Desired ouput format: " << format_;
-
-  // Calculate the number of sample frames per callback.
-  number_of_frames_ = params.frames_per_buffer();
-  DVLOG(1) << "Number of frames per callback: " << number_of_frames_;
-}
-
-AUAudioOutputStream::~AUAudioOutputStream() {
-}
-
-bool AUAudioOutputStream::Open() {
-  // Obtain the current input device selected by the user.
-  UInt32 size = sizeof(output_device_id_);
-  OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                                               &kDefaultOutputDeviceAddress,
-                                               0,
-                                               0,
-                                               &size,
-                                               &output_device_id_);
-  if (result != noErr || output_device_id_ == kAudioObjectUnknown) {
-    OSSTATUS_DLOG(ERROR, result)
-        << "Could not get default audio output device.";
-    return false;
-  }
-
-  // Open and initialize the DefaultOutputUnit.
-  AudioComponent comp;
-  AudioComponentDescription desc;
-
-  desc.componentType = kAudioUnitType_Output;
-  desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-  desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  desc.componentFlags = 0;
-  desc.componentFlagsMask = 0;
-  comp = AudioComponentFindNext(0, &desc);
-  if (!comp)
-    return false;
-
-  result = AudioComponentInstanceNew(comp, &output_unit_);
-  if (result != noErr) {
-    OSSTATUS_DLOG(ERROR, result) << "AudioComponentInstanceNew() failed.";
-    return false;
-  }
-
-  result = AudioUnitInitialize(output_unit_);
-  if (result != noErr) {
-    OSSTATUS_DLOG(ERROR, result) << "AudioUnitInitialize() failed.";
-    return false;
-  }
-
-  hardware_latency_frames_ = GetHardwareLatency();
-
-  return Configure();
-}
-
-bool AUAudioOutputStream::Configure() {
-  // Set the render callback.
-  AURenderCallbackStruct input;
-  input.inputProc = InputProc;
-  input.inputProcRefCon = this;
-  OSStatus result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioUnitProperty_SetRenderCallback,
-      kAudioUnitScope_Global,
-      0,
-      &input,
-      sizeof(input));
-  if (result != noErr) {
-    OSSTATUS_DLOG(ERROR, result)
-      << "AudioUnitSetProperty(kAudioUnitProperty_SetRenderCallback) failed.";
-    return false;
-  }
-
-  // Set the stream format.
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioUnitProperty_StreamFormat,
-      kAudioUnitScope_Input,
-      0,
-      &format_,
-      sizeof(format_));
-  if (result != noErr) {
-    OSSTATUS_DLOG(ERROR, result)
-        << "AudioUnitSetProperty(kAudioUnitProperty_StreamFormat) failed.";
-    return false;
-  }
-
-  // Set the buffer frame size.
-  // WARNING: Setting this value changes the frame size for all audio units in
-  // the current process.  It's imperative that the input and output frame sizes
-  // be the same as the frames_per_buffer() returned by
-  // GetDefaultOutputStreamParameters.
-  // See http://crbug.com/154352 for details.
-  const AudioParameters hw_params =
-      manager_->GetDefaultOutputStreamParameters();
-  if (number_of_frames_ != static_cast<size_t>(hw_params.frames_per_buffer())) {
-    DLOG(ERROR) << "Audio buffer size does not match hardware buffer size.";
-    return false;
-  }
-
-  UInt32 buffer_size = number_of_frames_;
-  result = AudioUnitSetProperty(
-      output_unit_,
-      kAudioDevicePropertyBufferFrameSize,
-      kAudioUnitScope_Output,
-      0,
-      &buffer_size,
-      sizeof(buffer_size));
-  if (result != noErr) {
-    OSSTATUS_DLOG(ERROR, result)
-        << "AudioUnitSetProperty(kAudioDevicePropertyBufferFrameSize) failed.";
-    return false;
-  }
-
-  return true;
-}
-
-void AUAudioOutputStream::Close() {
-  if (output_unit_)
-    AudioComponentInstanceDispose(output_unit_);
-
-  // Inform the audio manager that we have been closed. This can cause our
-  // destruction.
-  manager_->ReleaseOutputStream(this);
-}
-
-void AUAudioOutputStream::Start(AudioSourceCallback* callback) {
-  DCHECK(callback);
-  if (!output_unit_) {
-    DLOG(ERROR) << "Open() has not been called successfully";
-    return;
-  }
-
-  stopped_ = false;
-  {
-    base::AutoLock auto_lock(source_lock_);
-    source_ = callback;
-  }
-
-  AudioOutputUnitStart(output_unit_);
-}
-
-void AUAudioOutputStream::Stop() {
-  if (stopped_)
-    return;
-
-  AudioOutputUnitStop(output_unit_);
-
-  base::AutoLock auto_lock(source_lock_);
-  source_ = NULL;
-  stopped_ = true;
-}
-
-void AUAudioOutputStream::SetVolume(double volume) {
-  if (!output_unit_)
-    return;
-  volume_ = static_cast<float>(volume);
-
-  // TODO(crogers): set volume property
-}
-
-void AUAudioOutputStream::GetVolume(double* volume) {
-  if (!output_unit_)
-    return;
-  *volume = volume_;
-}
-
-// Pulls on our provider to get rendered audio stream.
-// 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 AUAudioOutputStream::Render(UInt32 number_of_frames,
-                                     AudioBufferList* io_data,
-                                     const AudioTimeStamp* output_time_stamp) {
-  // Update the playout latency.
-  double playout_latency_frames = GetPlayoutLatency(output_time_stamp);
-
-  AudioBuffer& buffer = io_data->mBuffers[0];
-  uint8* audio_data = reinterpret_cast<uint8*>(buffer.mData);
-  uint32 hardware_pending_bytes = static_cast<uint32>
-      ((playout_latency_frames + 0.5) * format_.mBytesPerFrame);
-
-  // Unfortunately AUAudioInputStream and AUAudioOutputStream share the frame
-  // size set by kAudioDevicePropertyBufferFrameSize above on a per process
-  // basis.  What this means is that the |number_of_frames| value may be larger
-  // or smaller than the value set during Configure().  In this case either
-  // audio input or audio output will be broken, so just output silence.
-  // TODO(crogers): Figure out what can trigger a change in |number_of_frames|.
-  // See http://crbug.com/154352 for details.
-  if (number_of_frames != static_cast<UInt32>(audio_bus_->frames())) {
-    memset(audio_data, 0, number_of_frames * format_.mBytesPerFrame);
-    return noErr;
-  }
-
-  int frames_filled = 0;
-  {
-    // Render() shouldn't be called except between AudioOutputUnitStart() and
-    // AudioOutputUnitStop() calls, but crash reports have shown otherwise:
-    // http://crbug.com/178765.  We use |source_lock_| to prevent races and
-    // crashes in Render() when |source_| is cleared.
-    base::AutoLock auto_lock(source_lock_);
-    if (!source_) {
-      memset(audio_data, 0, number_of_frames * format_.mBytesPerFrame);
-      return noErr;
-    }
-
-    frames_filled = source_->OnMoreData(
-        audio_bus_.get(), AudioBuffersState(0, hardware_pending_bytes));
-  }
-
-  // Note: If this ever changes to output raw float the data must be clipped and
-  // sanitized since it may come from an untrusted source such as NaCl.
-  audio_bus_->Scale(volume_);
-  audio_bus_->ToInterleaved(
-      frames_filled, format_.mBitsPerChannel / 8, audio_data);
-
-  return noErr;
-}
-
-// DefaultOutputUnit callback
-OSStatus AUAudioOutputStream::InputProc(void* user_data,
-                                        AudioUnitRenderActionFlags*,
-                                        const AudioTimeStamp* output_time_stamp,
-                                        UInt32,
-                                        UInt32 number_of_frames,
-                                        AudioBufferList* io_data) {
-  AUAudioOutputStream* audio_output =
-      static_cast<AUAudioOutputStream*>(user_data);
-  if (!audio_output)
-    return -1;
-
-  return audio_output->Render(number_of_frames, io_data, output_time_stamp);
-}
-
-int AUAudioOutputStream::HardwareSampleRate() {
-  // Determine the default output device's sample-rate.
-  AudioDeviceID device_id = kAudioObjectUnknown;
-  UInt32 info_size = sizeof(device_id);
-  OSStatus result = AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                                               &kDefaultOutputDeviceAddress,
-                                               0,
-                                               0,
-                                               &info_size,
-                                               &device_id);
-  if (result != noErr || device_id == kAudioObjectUnknown) {
-    OSSTATUS_DLOG(WARNING, result)
-        << "Could not get default audio output device.";
-    return 0;
-  }
-
-  Float64 nominal_sample_rate;
-  info_size = sizeof(nominal_sample_rate);
-
-  AudioObjectPropertyAddress nominal_sample_rate_address = {
-      kAudioDevicePropertyNominalSampleRate,
-      kAudioObjectPropertyScopeGlobal,
-      kAudioObjectPropertyElementMaster
-  };
-  result = AudioObjectGetPropertyData(device_id,
-                                      &nominal_sample_rate_address,
-                                      0,
-                                      0,
-                                      &info_size,
-                                      &nominal_sample_rate);
-  if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result)
-        << "Could not get default sample rate for device: " << device_id;
-    return 0;
-  }
-
-  return static_cast<int>(nominal_sample_rate);
-}
-
-double AUAudioOutputStream::GetHardwareLatency() {
-  if (!output_unit_ || output_device_id_ == kAudioObjectUnknown) {
-    DLOG(WARNING) << "Audio unit object is NULL or device ID is unknown";
-    return 0.0;
-  }
-
-  // Get audio unit latency.
-  Float64 audio_unit_latency_sec = 0.0;
-  UInt32 size = sizeof(audio_unit_latency_sec);
-  OSStatus result = AudioUnitGetProperty(output_unit_,
-                                         kAudioUnitProperty_Latency,
-                                         kAudioUnitScope_Global,
-                                         0,
-                                         &audio_unit_latency_sec,
-                                         &size);
-  if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result) << "Could not get audio unit latency";
-    return 0.0;
-  }
-
-  // Get output audio device latency.
-  AudioObjectPropertyAddress property_address = {
-    kAudioDevicePropertyLatency,
-    kAudioDevicePropertyScopeOutput,
-    kAudioObjectPropertyElementMaster
-  };
-  UInt32 device_latency_frames = 0;
-  size = sizeof(device_latency_frames);
-  result = AudioObjectGetPropertyData(output_device_id_,
-                                      &property_address,
-                                      0,
-                                      NULL,
-                                      &size,
-                                      &device_latency_frames);
-  if (result != noErr) {
-    OSSTATUS_DLOG(WARNING, result) << "Could not get audio unit latency";
-    return 0.0;
-  }
-
-  return static_cast<double>((audio_unit_latency_sec *
-      format_.mSampleRate) + device_latency_frames);
-}
-
-double AUAudioOutputStream::GetPlayoutLatency(
-    const AudioTimeStamp* output_time_stamp) {
-  // Ensure mHostTime is valid.
-  if ((output_time_stamp->mFlags & kAudioTimeStampHostTimeValid) == 0)
-    return 0;
-
-  // Get the delay between the moment getting the callback and the scheduled
-  // time stamp that tells when the data is going to be played out.
-  UInt64 output_time_ns = AudioConvertHostTimeToNanos(
-      output_time_stamp->mHostTime);
-  UInt64 now_ns = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-  // Prevent overflow leading to huge delay information; occurs regularly on
-  // the bots, probably less so in the wild.
-  if (now_ns > output_time_ns)
-    return 0;
-
-  double delay_frames = static_cast<double>
-      (1e-9 * (output_time_ns - now_ns) * format_.mSampleRate);
-
-  return (delay_frames + hardware_latency_frames_);
-}
-
-}  // namespace media