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-rw-r--r--content/renderer/audio_device.cc123
1 files changed, 95 insertions, 28 deletions
diff --git a/content/renderer/audio_device.cc b/content/renderer/audio_device.cc
index 0038467..1f38f29 100644
--- a/content/renderer/audio_device.cc
+++ b/content/renderer/audio_device.cc
@@ -1,12 +1,14 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
+// 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/message_loop.h"
#include "base/singleton.h"
#include "chrome/renderer/render_thread.h"
#include "content/common/audio_messages.h"
+#include "content/common/child_process.h"
#include "content/common/view_messages.h"
#include "media/audio/audio_util.h"
@@ -41,7 +43,7 @@ class AudioMessageFilterCreator {
scoped_refptr<AudioMessageFilter> filter_;
};
-}
+} // namespace
AudioDevice::AudioDevice(size_t buffer_size,
int channels,
@@ -49,17 +51,24 @@ AudioDevice::AudioDevice(size_t buffer_size,
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];
@@ -71,44 +80,91 @@ bool AudioDevice::Start() {
if (stream_id_)
return false;
- // Lazily create the message filter and share across AudioDevice instances.
- filter_ = AudioMessageFilterCreator::SharedFilter();
-
- stream_id_ = filter_->AddDelegate(this);
-
AudioParameters params;
params.format = AudioParameters::AUDIO_PCM_LOW_LATENCY;
params.channels = channels_;
params.sample_rate = static_cast<int>(sample_rate_);
- params.bits_per_sample = 16;
+ params.bits_per_sample = bits_per_sample_;
params.samples_per_packet = buffer_size_;
- filter_->Send(new AudioHostMsg_CreateStream(0, stream_id_, params, true));
+ // 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_) {
- OnDestroy();
- return true;
+ 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 false;
+
+ 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;
}
-void AudioDevice::OnDestroy() {
- // Make sure we don't call destroy more than once.
- DCHECK_NE(0, stream_id_);
+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_->RemoveDelegate(stream_id_);
filter_->Send(new AudioHostMsg_CloseStream(0, stream_id_));
+ filter_->RemoveDelegate(stream_id_);
stream_id_ = 0;
- if (audio_thread_.get()) {
- socket_->Close();
- audio_thread_->Join();
- }
+}
+
+void AudioDevice::SetVolumeOnIOThread(double volume) {
+ if (stream_id_)
+ filter_->Send(new AudioHostMsg_SetVolume(0, stream_id_, volume));
}
void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
@@ -117,7 +173,9 @@ void AudioDevice::OnRequestPacket(AudioBuffersState buffers_state) {
}
void AudioDevice::OnStateChanged(AudioStreamState state) {
- // Not needed in this simple implementation.
+ if (state == kAudioStreamError) {
+ DLOG(WARNING) << "AudioDevice::OnStateChanged(kError)";
+ }
NOTIMPLEMENTED();
}
@@ -131,7 +189,6 @@ void AudioDevice::OnLowLatencyCreated(
base::SharedMemoryHandle handle,
base::SyncSocket::Handle socket_handle,
uint32 length) {
-
#if defined(OS_WIN)
DCHECK(handle);
DCHECK(socket_handle);
@@ -140,7 +197,6 @@ void AudioDevice::OnLowLatencyCreated(
DCHECK_GE(socket_handle, 0);
#endif
DCHECK(length);
- DCHECK(!audio_thread_.get());
// TODO(crogers) : check that length is big enough for buffer_size_
@@ -156,28 +212,39 @@ void AudioDevice::OnLowLatencyCreated(
new base::DelegateSimpleThread(this, "renderer_audio_thread"));
audio_thread_->Start();
- filter_->Send(new AudioHostMsg_PlayStream(0, stream_id_));
+ if (filter_) {
+ filter_->message_loop()->PostTask(FROM_HERE,
+ NewRunnableMethod(this, &AudioDevice::StartOnIOThread));
+ }
}
void AudioDevice::OnVolume(double volume) {
- // Not needed in this simple implementation.
NOTIMPLEMENTED();
}
// Our audio thread runs here.
void AudioDevice::Run() {
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_) {
- // Ask client to render audio.
- callback_->Render(audio_data_, buffer_size_);
+ // 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());