Moves audio files from content\renderer\ to content\renderer\media.

content/renderer/audio_device.cc

Index: content/renderer/audio_device.cc
===================================================================
--- content/renderer/audio_device.cc	(revision 88131)
+++ content/renderer/audio_device.cc	(working copy)
@@ -1,254 +0,0 @@
-// Copyright (c) 2011 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 "content/renderer/audio_device.h"
-
-#include "base/memory/singleton.h"
-#include "base/message_loop.h"
-#include "content/common/audio_messages.h"
-#include "content/common/child_process.h"
-#include "content/common/view_messages.h"
-#include "content/renderer/render_thread.h"
-#include "media/audio/audio_util.h"
-
-scoped_refptr<AudioMessageFilter> AudioDevice::filter_;
-
-namespace {
-
-// AudioMessageFilterCreator is intended to be used as a singleton so we can
-// get access to a shared AudioMessageFilter.
-// Example usage:
-//   AudioMessageFilter* filter = AudioMessageFilterCreator::SharedFilter();
-
-class AudioMessageFilterCreator {
- public:
-  AudioMessageFilterCreator() {
-    int routing_id;
-    RenderThread::current()->Send(
-        new ViewHostMsg_GenerateRoutingID(&routing_id));
-    filter_ = new AudioMessageFilter(routing_id);
-    RenderThread::current()->AddFilter(filter_);
-  }
-
-  static AudioMessageFilter* SharedFilter() {
-    return GetInstance()->filter_.get();
-  }
-
-  static AudioMessageFilterCreator* GetInstance() {
-    return Singleton<AudioMessageFilterCreator>::get();
-  }
-
- private:
-  scoped_refptr<AudioMessageFilter> filter_;
-};
-
-}  // namespace
-
-AudioDevice::AudioDevice(size_t buffer_size,
-                         int channels,
-                         double sample_rate,
-                         RenderCallback* callback)
-    : buffer_size_(buffer_size),
-      channels_(channels),
-      bits_per_sample_(16),
-      sample_rate_(sample_rate),
-      callback_(callback),
-      audio_delay_milliseconds_(0),
-      volume_(1.0),
-      stream_id_(0) {
-  audio_data_.reserve(channels);
-  for (int i = 0; i < channels; ++i) {
-    float* channel_data = new float[buffer_size];
-    audio_data_.push_back(channel_data);
-  }
-  // Lazily create the message filter and share across AudioDevice instances.
-  filter_ = AudioMessageFilterCreator::SharedFilter();
-}
-
-AudioDevice::~AudioDevice() {
-  // Make sure we have been shut down.
-  DCHECK_EQ(0, stream_id_);
-  Stop();
-  for (int i = 0; i < channels_; ++i)
-    delete [] audio_data_[i];
-}
-
-bool AudioDevice::Start() {
-  // Make sure we don't call Start() more than once.
-  DCHECK_EQ(0, stream_id_);
-  if (stream_id_)
-    return false;
-
-  AudioParameters params;
-  params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
-  params.channels = channels_;
-  params.sample_rate = static_cast<int>(sample_rate_);
-  params.bits_per_sample = bits_per_sample_;
-  params.samples_per_packet = buffer_size_;
-
-  // Ensure that the initialization task is posted on the I/O thread by
-  // accessing the I/O message loop directly. This approach avoids a race
-  // condition which could exist if the message loop of the filter was
-  // used instead.
-  DCHECK(ChildProcess::current()) << "Must be in the renderer";
-  MessageLoop* message_loop = ChildProcess::current()->io_message_loop();
-  if (!message_loop)
-    return false;
-
-  message_loop->PostTask(FROM_HERE,
-      NewRunnableMethod(this, &AudioDevice::InitializeOnIOThread, params));
-
-  return true;
-}
-
-bool AudioDevice::Stop() {
-  if (!stream_id_)
-    return false;
-
-  filter_->message_loop()->PostTask(FROM_HERE,
-      NewRunnableMethod(this, &AudioDevice::ShutDownOnIOThread));
-
-  if (audio_thread_.get()) {
-    socket_->Close();
-    audio_thread_->Join();
-  }
-
-  return true;
-}
-
-bool AudioDevice::SetVolume(double volume) {
-  if (!stream_id_)
-    return false;
-
-  if (volume < 0 || volume > 1.0)
-    return false;
-
-  filter_->message_loop()->PostTask(FROM_HERE,
-      NewRunnableMethod(this, &AudioDevice::SetVolumeOnIOThread, volume));
-
-  volume_ = volume;
-
-  return true;
-}
-
-bool AudioDevice::GetVolume(double* volume) {
-  if (!stream_id_)
-    return false;
-
-  // Return a locally cached version of the current scaling factor.
-  *volume = volume_;
-
-  return true;
-}
-
-void AudioDevice::InitializeOnIOThread(const AudioParameters& params) {
-  stream_id_ = filter_->AddDelegate(this);
-  filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true));
-}
-
-void AudioDevice::StartOnIOThread() {
-  if (stream_id_)
-    filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_));
-}
-
-void AudioDevice::ShutDownOnIOThread() {
-  // Make sure we don't call shutdown more than once.
-  if (!stream_id_)
-    return;
-
-  filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_));
-  filter_->RemoveDelegate(stream_id_);
-  stream_id_ = 0;
-}
-
-void AudioDevice::SetVolumeOnIOThread(double volume) {
-  if (stream_id_)
-    filter_->Send(new AudioHostMsg_SetVolume(0, stream_id_, volume));
-}
-
-void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
-  // This method does not apply to the low-latency system.
-  NOTIMPLEMENTED();
-}
-
-void AudioDevice::OnStateChanged(AudioStreamState state) {
-  if (state == kAudioStreamError) {
-    DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
-  }
-  NOTIMPLEMENTED();
-}
-
-void AudioDevice::OnCreated(
-    base::SharedMemoryHandle handle, uint32 length) {
-  // Not needed in this simple implementation.
-  NOTIMPLEMENTED();
-}
-
-void AudioDevice::OnLowLatencyCreated(
-    base::SharedMemoryHandle handle,
-    base::SyncSocket::Handle socket_handle,
-    uint32 length) {
-#if defined(OS_WIN)
-  DCHECK(handle);
-  DCHECK(socket_handle);
-#else
-  DCHECK_GE(handle.fd, 0);
-  DCHECK_GE(socket_handle, 0);
-#endif
-  DCHECK(length);
-
-  // TODO(crogers) : check that length is big enough for buffer_size_
-
-  shared_memory_.reset(new base::SharedMemory(handle, false));
-  shared_memory_->Map(length);
-
-  socket_.reset(new base::SyncSocket(socket_handle));
-  // Allow the client to pre-populate the buffer.
-  FireRenderCallback();
-
-  audio_thread_.reset(
-      new base::DelegateSimpleThread(this, "renderer_audio_thread"));
-  audio_thread_->Start();
-
-  if (filter_) {
-    filter_->message_loop()->PostTask(FROM_HERE,
-        NewRunnableMethod(this, &AudioDevice::StartOnIOThread));
-  }
-}
-
-void AudioDevice::OnVolume(double volume) {
-  NOTIMPLEMENTED();
-}
-
-// Our audio thread runs here.
-void AudioDevice::Run() {
-    audio_thread_->SetThreadPriority(base::kThreadPriority_RealtimeAudio);
-
-  int pending_data;
-  const int samples_per_ms = static_cast<int>(sample_rate_) / 1000;
-  const int bytes_per_ms = channels_ * (bits_per_sample_ / 8) * samples_per_ms;
-
-  while (sizeof(pending_data) == socket_->Receive(&pending_data,
-                                                  sizeof(pending_data)) &&
-                                                  pending_data >= 0) {
-    {
-      // Convert the number of pending bytes in the render buffer
-      // into milliseconds.
-      audio_delay_milliseconds_ = pending_data / bytes_per_ms;
-    }
-
-    FireRenderCallback();
-  }
-}
-
-void AudioDevice::FireRenderCallback() {
-  if (callback_) {
-    // Update the audio-delay measurement then ask client to render audio.
-    callback_->Render(audio_data_, buffer_size_, audio_delay_milliseconds_);
-
-    // Interleave, scale, and clip to int16.
-    int16* output_buffer16 = static_cast<int16*>(shared_memory_data());
-    media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_);
-  }
-}