| Index: content/renderer/render_audiosourceprovider.cc
|
| ===================================================================
|
| --- content/renderer/render_audiosourceprovider.cc (revision 0)
|
| +++ content/renderer/render_audiosourceprovider.cc (revision 0)
|
| @@ -0,0 +1,148 @@
|
| +// 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/render_audiosourceprovider.h"
|
| +
|
| +#include "base/basictypes.h"
|
| +#include "base/logging.h"
|
| +#include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProviderClient.h"
|
| +
|
| +using std::vector;
|
| +using WebKit::WebVector;
|
| +
|
| +RenderAudioSourceProvider::RenderAudioSourceProvider()
|
| + : is_initialized_(false),
|
| + channels_(0),
|
| + sample_rate_(0.0),
|
| + is_running_(false),
|
| + volume_(1.0),
|
| + renderer_(NULL),
|
| + client_(NULL) {
|
| + // We create the AudioDevice here because it must be created in the
|
| + // main thread. But we don't yet know the audio format (sample-rate, etc.)
|
| + // at this point. Later, when Initialize() is called, we have
|
| + // the audio format information and call the AudioDevice::Initialize()
|
| + // method to fully initialize it.
|
| + default_sink_ = new AudioDevice();
|
| +}
|
| +
|
| +RenderAudioSourceProvider::~RenderAudioSourceProvider() {}
|
| +
|
| +void RenderAudioSourceProvider::Start() {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + if (!client_)
|
| + default_sink_->Start();
|
| + is_running_ = true;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::Stop() {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + if (!client_)
|
| + default_sink_->Stop();
|
| + is_running_ = false;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::Play() {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + if (!client_)
|
| + default_sink_->Play();
|
| + is_running_ = true;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::Pause(bool flush) {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + if (!client_)
|
| + default_sink_->Pause(flush);
|
| + is_running_ = false;
|
| +}
|
| +
|
| +bool RenderAudioSourceProvider::SetVolume(double volume) {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + if (!client_)
|
| + default_sink_->SetVolume(volume);
|
| + volume_ = volume;
|
| + return true;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::GetVolume(double* volume) {
|
| + if (!client_)
|
| + default_sink_->GetVolume(volume);
|
| + else if (volume)
|
| + *volume = volume_;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::Initialize(
|
| + size_t buffer_size,
|
| + int channels,
|
| + double sample_rate,
|
| + AudioParameters::Format latency_format,
|
| + RenderCallback* renderer) {
|
| + base::AutoLock auto_lock(sink_lock_);
|
| + CHECK(!is_initialized_);
|
| + renderer_ = renderer;
|
| +
|
| + default_sink_->Initialize(buffer_size,
|
| + channels,
|
| + sample_rate,
|
| + latency_format,
|
| + renderer);
|
| +
|
| + if (client_) {
|
| + // Inform WebKit about the audio stream format.
|
| + client_->setFormat(channels, sample_rate);
|
| + }
|
| +
|
| + // Keep track of the format in case the client hasn't yet been set.
|
| + channels_ = channels;
|
| + sample_rate_ = sample_rate;
|
| + is_initialized_ = true;
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::setClient(
|
| + WebKit::WebAudioSourceProviderClient* client) {
|
| + // Synchronize with other uses of client_ and default_sink_.
|
| + base::AutoLock auto_lock(sink_lock_);
|
| +
|
| + if (client && client != client_) {
|
| + // Detach the audio renderer from normal playback.
|
| + default_sink_->Pause(true);
|
| +
|
| + // The client will now take control by calling provideInput() periodically.
|
| + client_ = client;
|
| +
|
| + if (is_initialized_) {
|
| + // The client needs to be notified of the audio format, if available.
|
| + // If the format is not yet available, we'll be notified later
|
| + // when Initialize() is called.
|
| +
|
| + // Inform WebKit about the audio stream format.
|
| + client->setFormat(channels_, sample_rate_);
|
| + }
|
| + } else if (!client && client_) {
|
| + // Restore normal playback.
|
| + client_ = NULL;
|
| + // TODO(crogers): We should call default_sink_->Play() if we're
|
| + // in the playing state.
|
| + }
|
| +}
|
| +
|
| +void RenderAudioSourceProvider::provideInput(
|
| + const WebVector<float*>& audio_data, size_t number_of_frames) {
|
| + DCHECK(client_);
|
| +
|
| + if (renderer_ && is_initialized_ && is_running_) {
|
| + // Wrap WebVector as std::vector.
|
| + vector<float*> v(audio_data.size());
|
| + for (size_t i = 0; i < audio_data.size(); ++i)
|
| + v[i] = audio_data[i];
|
| +
|
| + // TODO(crogers): figure out if we should volume scale here or in common
|
| + // WebAudio code. In any case we need to take care of volume.
|
| + renderer_->Render(v, number_of_frames, 0);
|
| + } else {
|
| + // Provide silence if the source is not running.
|
| + for (size_t i = 0; i < audio_data.size(); ++i)
|
| + memset(audio_data[i], 0, sizeof(float) * number_of_frames);
|
| + }
|
| +}
|
|
|