Implement renderer AudioDevice API for low-latency audio output...

chrome/renderer/audio_device.cc

Index: chrome/renderer/audio_device.cc
===================================================================
--- chrome/renderer/audio_device.cc	(revision 0)
+++ chrome/renderer/audio_device.cc	(revision 0)
@@ -0,0 +1,165 @@
+// Copyright (c) 2010 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 "chrome/renderer/audio_device.h"
+
+#include "chrome/common/render_messages_params.h"
+#include "chrome/renderer/render_thread.h"
+#include "media/audio/audio_util.h"
+
+scoped_refptr<AudioMessageFilter> AudioDevice::filter_;
+Lock AudioDevice::message_filter_lock_;

[scherkus (not reviewing), 2011/01/12 03:11:43]

hrmm.. global/static non-basic-types are no good I'm afraid :\

have you looked into how other MessageFilter classes are used?  perhaps
base/singleton.h might help out here

+
+AudioDevice::AudioDevice(size_t buffer_size,
+                         int channels,
+                         double sample_rate,
+                         RenderCallback* callback)
+    : buffer_size_(buffer_size),
+      channels_(channels),
+      sample_rate_(sample_rate),
+      callback_(callback),
+      stream_id_(0) {
+  audio_data_.reserve(channels);
+  for (int i = 0; i < channels; ++i) {
+    float* channel_data = static_cast<float*>(new float[buffer_size]);

[scherkus (not reviewing), 2011/01/12 03:11:43]

don't think we need static_cast<> anymore :)

[Chris Rogers, 2011/01/12 21:26:10]

Done.

+    audio_data_.push_back(channel_data);
+  }
+}
+
+AudioDevice::~AudioDevice() {
+  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;
+
+  // Lazily create the message filter and share across AudioDevice instances.
+  {
+    AutoLock auto_lock(message_filter_lock_);
+    if (!filter_.get()) {
+      int routing_id;
+      RenderThread::current()->Send(
+          new ViewHostMsg_GenerateRoutingID(&routing_id));
+      filter_ = new AudioMessageFilter(routing_id);
+      RenderThread::current()->AddFilter(filter_);
+    }
+  }
+
+  stream_id_ = filter_->AddDelegate(this);
+
+  ViewHostMsg_Audio_CreateStream_Params params;
+  params.params.format = AudioParameters::AUDIO_PCM_LINEAR;
+  params.params.channels = channels_;
+  params.params.sample_rate = sample_rate_;
+  params.params.bits_per_sample = 16;
+  params.params.samples_per_packet = buffer_size_;
+
+  filter_->Send(
+      new ViewHostMsg_CreateAudioStream(0, stream_id_, params, true));
+
+  return true;
+}
+
+bool AudioDevice::Stop() {
+  if (stream_id_) {
+    OnDestroy();
+    return true;
+  }
+  return false;
+}
+
+void AudioDevice::OnDestroy() {
+  // Make sure we don't call destroy more than once.
+  DCHECK_NE(0, stream_id_);
+  if (!stream_id_)
+    return;
+
+  filter_->RemoveDelegate(stream_id_);
+  filter_->Send(new ViewHostMsg_CloseAudioStream(0, stream_id_));
+  stream_id_ = 0;
+  if (audio_thread_.get()) {
+    socket_->Close();
+    audio_thread_->Join();
+  }
+}
+
+void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
+  // This method does not apply to the low-latency system.
+  NOTIMPLEMENTED();
+}
+
+void AudioDevice::OnStateChanged(
+    const ViewMsg_AudioStreamState_Params& state) {
+  // Not needed in this simple implementation.
+  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);
+  DCHECK(!audio_thread_.get());
+
+  // 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();
+
+  // TODO(crogers): we could optionally set the thread to high-priority
+  audio_thread_.reset(
+      new base::DelegateSimpleThread(this, "renderer_audio_thread"));
+  audio_thread_->Start();
+
+  filter_->Send(new ViewHostMsg_PlayAudioStream(0, stream_id_));
+}
+
+void AudioDevice::OnVolume(double volume) {
+  // Not needed in this simple implementation.
+  NOTIMPLEMENTED();
+}
+
+// Our audio thread runs here.
+void AudioDevice::Run() {
+  int pending_data;
+  while (sizeof(pending_data) == socket_->Receive(&pending_data,
+                                                  sizeof(pending_data)) &&
+                                                  pending_data >= 0) {
+    FireRenderCallback();
+  }
+}
+
+void AudioDevice::FireRenderCallback() {
+  if (callback_) {
+    // Ask client to render audio.
+    callback_->Render(audio_data_, buffer_size_);
+
+    // Interleave, scale, and clip to int16.
+    int16* output_buffer16 = static_cast<int16*>(shared_memory_data());
+    media::InterleaveFloatToInt16(audio_data_, output_buffer16, buffer_size_);
+  }
+}
Property changes on: chrome/renderer/audio_device.cc
___________________________________________________________________
Added: svn:eol-style
   + LF