blob: b722a33af4c18e3073fcea36576f1775234a440e [file] [log] [blame]
// Copyright 2013 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/webaudiosourceprovider_impl.h"
#include <vector>
#include "base/logging.h"
#include "third_party/WebKit/public/web/WebAudioSourceProviderClient.h"
using WebKit::WebVector;
namespace content {
namespace {
// Simple helper class for Try() locks. Lock is Try()'d on construction and
// must be checked via the locked() attribute. If acquisition was successful
// the lock will be released upon destruction.
// TODO(dalecurtis): This should probably move to base/ if others start using
// this pattern.
class AutoTryLock {
public:
explicit AutoTryLock(base::Lock& lock)
: lock_(lock),
acquired_(lock_.Try()) {}
bool locked() const { return acquired_; }
~AutoTryLock() {
if (acquired_) {
lock_.AssertAcquired();
lock_.Release();
}
}
private:
base::Lock& lock_;
const bool acquired_;
DISALLOW_COPY_AND_ASSIGN(AutoTryLock);
};
} // namespace
WebAudioSourceProviderImpl::WebAudioSourceProviderImpl(
const scoped_refptr<media::AudioRendererSink>& sink)
: channels_(0),
sample_rate_(0),
volume_(1.0),
state_(kStopped),
renderer_(NULL),
client_(NULL),
sink_(sink) {
}
WebAudioSourceProviderImpl::~WebAudioSourceProviderImpl() {}
void WebAudioSourceProviderImpl::setClient(
WebKit::WebAudioSourceProviderClient* client) {
base::AutoLock auto_lock(sink_lock_);
if (client && client != client_) {
// Detach the audio renderer from normal playback.
sink_->Stop();
// The client will now take control by calling provideInput() periodically.
client_ = client;
if (renderer_) {
// 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;
sink_->SetVolume(volume_);
if (state_ >= kStarted)
sink_->Start();
if (state_ >= kPlaying)
sink_->Play();
}
}
void WebAudioSourceProviderImpl::provideInput(
const WebVector<float*>& audio_data, size_t number_of_frames) {
if (!bus_wrapper_ ||
static_cast<size_t>(bus_wrapper_->channels()) != audio_data.size()) {
bus_wrapper_ = media::AudioBus::CreateWrapper(audio_data.size());
}
bus_wrapper_->set_frames(number_of_frames);
for (size_t i = 0; i < audio_data.size(); ++i)
bus_wrapper_->SetChannelData(i, audio_data[i]);
// Use a try lock to avoid contention in the real-time audio thread.
AutoTryLock auto_try_lock(sink_lock_);
if (!auto_try_lock.locked() || state_ != kPlaying) {
// Provide silence if we failed to acquire the lock or the source is not
// running.
bus_wrapper_->Zero();
return;
}
DCHECK(renderer_);
DCHECK(client_);
DCHECK_EQ(channels_, bus_wrapper_->channels());
renderer_->Render(bus_wrapper_.get(), 0);
bus_wrapper_->Scale(volume_);
}
void WebAudioSourceProviderImpl::Start() {
base::AutoLock auto_lock(sink_lock_);
DCHECK_EQ(state_, kStopped);
state_ = kStarted;
if (!client_)
sink_->Start();
}
void WebAudioSourceProviderImpl::Stop() {
base::AutoLock auto_lock(sink_lock_);
state_ = kStopped;
if (!client_)
sink_->Stop();
}
void WebAudioSourceProviderImpl::Play() {
base::AutoLock auto_lock(sink_lock_);
DCHECK_EQ(state_, kStarted);
state_ = kPlaying;
if (!client_)
sink_->Play();
}
void WebAudioSourceProviderImpl::Pause() {
base::AutoLock auto_lock(sink_lock_);
DCHECK(state_ == kPlaying || state_ == kStarted);
state_ = kStarted;
if (!client_)
sink_->Pause();
}
bool WebAudioSourceProviderImpl::SetVolume(double volume) {
base::AutoLock auto_lock(sink_lock_);
volume_ = volume;
if (!client_)
sink_->SetVolume(volume);
return true;
}
void WebAudioSourceProviderImpl::Initialize(
const media::AudioParameters& params,
RenderCallback* renderer) {
base::AutoLock auto_lock(sink_lock_);
CHECK(!renderer_);
renderer_ = renderer;
DCHECK_EQ(state_, kStopped);
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_);
}
}
} // namespace content