diff options
Diffstat (limited to 'content/renderer/audio_device.cc')
-rw-r--r-- | content/renderer/audio_device.cc | 123 |
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()); |