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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 "content/renderer/media/webrtc_local_audio_source_provider.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()
: track_(NULL),
source_provider_(NULL) {
}
WebAudioCapturerSource::~WebAudioCapturerSource() {
}
void WebAudioCapturerSource::setFormat(
size_t number_of_channels, float sample_rate) {
DCHECK(thread_checker_.CalledOnValidThread());
DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate="
<< sample_rate << ")";
if (number_of_channels > 2) {
// TODO(xians): Handle more than just the mono and stereo cases.
LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format.";
return;
}
ChannelLayout channel_layout =
number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO;
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, number_of_channels, 0, sample_rate, 16,
sample_rate / 100);
// Update the downstream client to use the same format as what WebKit
// is using.
if (track_)
track_->SetCaptureFormat(params_);
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,
WebRtcLocalAudioSourceProvider* source_provider) {
DCHECK(thread_checker_.CalledOnValidThread());
DCHECK(track);
// |source_provider| may be NULL if no getUserMedia has been called before
// calling CreateMediaStreamDestination.
// The downstream client should be configured the same as what WebKit
// is feeding it.
track->SetCaptureFormat(params_);
base::AutoLock auto_lock(lock_);
track_ = track;
source_provider_ = source_provider;
}
void WebAudioCapturerSource::Stop() {
DCHECK(thread_checker_.CalledOnValidThread());
base::AutoLock auto_lock(lock_);
track_ = NULL;
source_provider_ = NULL;
}
void WebAudioCapturerSource::consumeAudio(
const WebKit::WebVector<const float*>& audio_data,
size_t number_of_frames) {
base::AutoLock auto_lock(lock_);
if (!track_)
return;
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;
}
fifo_->Push(wrapper_bus_.get());
int capture_frames = params_.frames_per_buffer();
int delay_ms = 0;
int volume = 0;
bool key_pressed = false;
while (fifo_->frames() >= capture_frames) {
if (source_provider_) {
source_provider_->GetAudioProcessingParams(
&delay_ms, &volume, &key_pressed);
}
fifo_->Consume(capture_bus_.get(), 0, capture_frames);
track_->Capture(capture_bus_.get(), delay_ms, volume, key_pressed);
}
}
} // namespace content
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