blob: a1051e0e5d2b3df8ede8693c993fda0d7ca9e38e [file] [log] [blame]
// Copyright (c) 2012 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/media/render_audiosourceprovider.h"
#include "base/basictypes.h"
#include "base/command_line.h"
#include "base/logging.h"
#include "content/public/common/content_switches.h"
#include "content/renderer/media/audio_device_factory.h"
#include "content/renderer/media/audio_renderer_mixer_manager.h"
#include "content/renderer/render_thread_impl.h"
#include "media/base/audio_renderer_mixer_input.h"
#include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProviderClient.h"
using content::AudioDeviceFactory;
using content::AudioRendererMixerManager;
using std::vector;
using WebKit::WebVector;
RenderAudioSourceProvider::RenderAudioSourceProvider()
: is_initialized_(false),
channels_(0),
sample_rate_(0),
is_running_(false),
renderer_(NULL),
client_(NULL) {
// We create an AudioRendererSink here, 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 AudioRendererSink::Initialize()
// to fully initialize it.
const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
if (cmd_line->HasSwitch(switches::kEnableRendererSideMixing)) {
default_sink_ = RenderThreadImpl::current()->
GetAudioRendererMixerManager()->CreateInput();
} else {
default_sink_ = AudioDeviceFactory::NewOutputDevice();
}
}
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_->Stop();
// 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(audio_data[0]) * number_of_frames);
}
}
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);
return true;
}
void RenderAudioSourceProvider::Initialize(
const media::AudioParameters& params,
RenderCallback* renderer) {
base::AutoLock auto_lock(sink_lock_);
CHECK(!is_initialized_);
renderer_ = renderer;
default_sink_->Initialize(params, renderer);
// Keep track of the format in case the client hasn't yet been set.
channels_ = params.channels();
sample_rate_ = params.sample_rate();
if (client_) {
// Inform WebKit about the audio stream format.
client_->setFormat(channels_, sample_rate_);
}
is_initialized_ = true;
}
RenderAudioSourceProvider::~RenderAudioSourceProvider() {}