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// 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 "base/singleton.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_;
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_;
};
}
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 = new float[buffer_size];
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.
filter_ = AudioMessageFilterCreator::SharedFilter();
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 = static_cast<int>(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_);
}
}
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