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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 "third_party/blink/renderer/platform/mediastream/webaudio_media_stream_source.h"
#include <utility>
#include "base/bind.h"
#include "base/bind_helpers.h"
#include "base/logging.h"
namespace blink {
WebAudioMediaStreamSource::WebAudioMediaStreamSource(
WebMediaStreamSource* blink_source,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: MediaStreamAudioSource(std::move(task_runner), false /* is_remote */),
is_registered_consumer_(false),
fifo_(base::Bind(&WebAudioMediaStreamSource::DeliverRebufferedAudio,
base::Unretained(this))),
blink_source_(*blink_source) {
DVLOG(1) << "WebAudioMediaStreamSource::WebAudioMediaStreamSource()";
}
WebAudioMediaStreamSource::~WebAudioMediaStreamSource() {
DVLOG(1) << "WebAudioMediaStreamSource::~WebAudioMediaStreamSource()";
EnsureSourceIsStopped();
}
void WebAudioMediaStreamSource::SetFormat(size_t number_of_channels,
float sample_rate) {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
VLOG(1) << "WebAudio media stream source changed format to: channels="
<< number_of_channels << ", sample_rate=" << sample_rate;
// If the channel count is greater than 8, use discrete layout. However,
// anything beyond 8 is ignored by some audio tracks/sinks.
media::ChannelLayout channel_layout =
number_of_channels > 8 ? media::CHANNEL_LAYOUT_DISCRETE
: media::GuessChannelLayout(number_of_channels);
// Set the format used by this WebAudioMediaStreamSource. We are using 10ms
// data as a buffer size since that is the native buffer size of WebRtc packet
// running on.
//
// TODO(miu): Re-evaluate whether this is needed. For now (this refactoring),
// I did not want to change behavior. https://crbug.com/577874
fifo_.Reset(sample_rate / 100);
media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
channel_layout, sample_rate,
fifo_.frames_per_buffer());
// Take care of the discrete channel layout case.
params.set_channels_for_discrete(number_of_channels);
MediaStreamAudioSource::SetFormat(params);
if (!wrapper_bus_ || wrapper_bus_->channels() != params.channels())
wrapper_bus_ = media::AudioBus::CreateWrapper(params.channels());
}
bool WebAudioMediaStreamSource::EnsureSourceIsStarted() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (is_registered_consumer_)
return true;
if (blink_source_.IsNull() || !blink_source_.RequiresAudioConsumer())
return false;
VLOG(1) << "Starting WebAudio media stream source.";
blink_source_.AddAudioConsumer(this);
is_registered_consumer_ = true;
return true;
}
void WebAudioMediaStreamSource::EnsureSourceIsStopped() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (!is_registered_consumer_)
return;
is_registered_consumer_ = false;
DCHECK(!blink_source_.IsNull());
blink_source_.RemoveAudioConsumer(this);
blink_source_.Reset();
VLOG(1) << "Stopped WebAudio media stream source. Final audio parameters={"
<< GetAudioParameters().AsHumanReadableString() << "}.";
}
void WebAudioMediaStreamSource::ConsumeAudio(
const WebVector<const float*>& audio_data,
size_t number_of_frames) {
// TODO(miu): Plumbing is needed to determine the actual capture timestamp
// of the audio, instead of just snapshotting base::TimeTicks::Now(), for
// proper audio/video sync. https://crbug.com/335335
current_reference_time_ = base::TimeTicks::Now();
wrapper_bus_->set_frames(number_of_frames);
DCHECK_EQ(wrapper_bus_->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]));
// The following will result in zero, one, or multiple synchronous calls to
// DeliverRebufferedAudio().
fifo_.Push(*wrapper_bus_);
}
void WebAudioMediaStreamSource::DeliverRebufferedAudio(
const media::AudioBus& audio_bus,
int frame_delay) {
const base::TimeTicks reference_time =
current_reference_time_ +
base::TimeDelta::FromMicroseconds(
frame_delay * base::Time::kMicrosecondsPerSecond /
MediaStreamAudioSource::GetAudioParameters().sample_rate());
MediaStreamAudioSource::DeliverDataToTracks(audio_bus, reference_time);
}
} // namespace blink
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