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// 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/webaudio_capturer_source.h"
#include "base/logging.h"
#include "base/time/time.h"
#include "content/renderer/media/webrtc_local_audio_track.h"
using media::AudioBus;
using media::AudioFifo;
using media::AudioParameters;
using media::ChannelLayout;
using media::CHANNEL_LAYOUT_MONO;
using media::CHANNEL_LAYOUT_STEREO;
static const int kMaxNumberOfBuffersInFifo = 5;
namespace content {
WebAudioCapturerSource::WebAudioCapturerSource(
const blink::WebMediaStreamSource& blink_source)
: track_(NULL),
audio_format_changed_(false),
blink_source_(blink_source) {
}
WebAudioCapturerSource::~WebAudioCapturerSource() {
DCHECK(thread_checker_.CalledOnValidThread());
removeFromBlinkSource();
}
void WebAudioCapturerSource::setFormat(
size_t number_of_channels, float sample_rate) {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
<< sample_rate << ")";
// If the channel count is greater than 8, use discrete layout. However,
// anything beyond 8 is ignored by the subsequent (WebRTC) audio pipeline.
ChannelLayout channel_layout =
number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
: media::GuessChannelLayout(number_of_channels);
base::AutoLock auto_lock(lock_);
// Set the format used by this WebAudioCapturerSource. We are using 10ms data
// as buffer size since that is the native buffer size of WebRtc packet
// running on.
params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, channel_layout,
sample_rate, 16, sample_rate / 100);
// Take care of the discrete channel layout case.
params_.set_channels_for_discrete(number_of_channels);
audio_format_changed_ = true;
wrapper_bus_ = AudioBus::CreateWrapper(params_.channels());
capture_bus_ = AudioBus::Create(params_);
fifo_.reset(new AudioFifo(
params_.channels(),
kMaxNumberOfBuffersInFifo * params_.frames_per_buffer()));
}
void WebAudioCapturerSource::Start(WebRtcLocalAudioTrack* track) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(track);
base::AutoLock auto_lock(lock_);
track_ = track;
}
void WebAudioCapturerSource::Stop() {
DCHECK(thread_checker_.CalledOnValidThread());
{
base::AutoLock auto_lock(lock_);
track_ = NULL;
}
// removeFromBlinkSource() should not be called while |lock_| is acquired,
// as it could result in a deadlock.
removeFromBlinkSource();
}
void WebAudioCapturerSource::consumeAudio(
const blink::WebVector<const float*>& audio_data,
size_t number_of_frames) {
base::AutoLock auto_lock(lock_);
if (!track_)
return;
// Update the downstream client if the audio format has been changed.
if (audio_format_changed_) {
track_->OnSetFormat(params_);
audio_format_changed_ = false;
}
wrapper_bus_->set_frames(number_of_frames);
// Make sure WebKit is honoring what it told us up front
// about the channels.
DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size()));
for (size_t i = 0; i < audio_data.size(); ++i)
wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i]));
// Handle mismatch between WebAudio buffer-size and WebRTC.
int available = fifo_->max_frames() - fifo_->frames();
if (available < static_cast<int>(number_of_frames)) {
NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun.";
return;
}
// Compute the estimated capture time of the first sample frame of audio that
// will be consumed from the FIFO in the loop below.
base::TimeTicks estimated_capture_time = base::TimeTicks::Now() -
fifo_->frames() * base::TimeDelta::FromSeconds(1) / params_.sample_rate();
fifo_->Push(wrapper_bus_.get());
while (fifo_->frames() >= capture_bus_->frames()) {
fifo_->Consume(capture_bus_.get(), 0, capture_bus_->frames());
track_->Capture(*capture_bus_, estimated_capture_time, false);
// Advance the estimated capture time for the next FIFO consume operation.
estimated_capture_time +=
capture_bus_->frames() * base::TimeDelta::FromSeconds(1) /
params_.sample_rate();
}
}
// If registered as audio consumer in |blink_source_|, deregister from
// |blink_source_| and stop keeping a reference to |blink_source_|.
// Failure to call this method when stopping the track might leave an invalid
// WebAudioCapturerSource reference still registered as an audio consumer on
// the blink side.
void WebAudioCapturerSource::removeFromBlinkSource() {
if (!blink_source_.isNull()) {
blink_source_.removeAudioConsumer(this);
blink_source_.reset();
}
}
} // namespace content
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