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/*
* libjingle
* Copyright 2015 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/rtpsender.h"
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/videosourceinterface.h"
namespace webrtc {
LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
rtc::CritScope lock(&lock_);
if (sink_)
sink_->OnClose();
}
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
int number_of_channels,
size_t number_of_frames) {
rtc::CritScope lock(&lock_);
if (sink_) {
sink_->OnData(audio_data, bits_per_sample, sample_rate, number_of_channels,
number_of_frames);
}
}
void LocalAudioSinkAdapter::SetSink(cricket::AudioRenderer::Sink* sink) {
rtc::CritScope lock(&lock_);
ASSERT(!sink || !sink_);
sink_ = sink;
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
uint32_t ssrc,
AudioProviderInterface* provider)
: id_(track->id()),
track_(track),
ssrc_(ssrc),
provider_(provider),
cached_track_enabled_(track->enabled()),
sink_adapter_(new LocalAudioSinkAdapter()) {
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
Reconfigure();
}
AudioRtpSender::~AudioRtpSender() {
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
Stop();
}
void AudioRtpSender::OnChanged() {
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
if (track->kind() != "audio") {
LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
<< " track.";
return false;
}
AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
// Detach from old track.
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
// Attach to new track.
track_ = audio_track;
cached_track_enabled_ = track_->enabled();
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
Reconfigure();
return true;
}
void AudioRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (!provider_) {
return;
}
cricket::AudioOptions options;
provider_->SetAudioSend(ssrc_, false, options, nullptr);
provider_ = nullptr;
}
void AudioRtpSender::Reconfigure() {
if (!provider_) {
return;
}
cricket::AudioOptions options;
if (track_->enabled() && track_->GetSource()) {
// TODO(xians): Remove this static_cast since we should be able to connect
// a remote audio track to peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
// Use the renderer if the audio track has one, otherwise use the sink
// adapter owned by this class.
cricket::AudioRenderer* renderer =
track_->GetRenderer() ? track_->GetRenderer() : sink_adapter_.get();
ASSERT(renderer != nullptr);
provider_->SetAudioSend(ssrc_, track_->enabled(), options, renderer);
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
uint32_t ssrc,
VideoProviderInterface* provider)
: id_(track->id()),
track_(track),
ssrc_(ssrc),
provider_(provider),
cached_track_enabled_(track->enabled()) {
track_->RegisterObserver(this);
VideoSourceInterface* source = track_->GetSource();
if (source) {
provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
}
Reconfigure();
}
VideoRtpSender::~VideoRtpSender() {
track_->UnregisterObserver(this);
Stop();
}
void VideoRtpSender::OnChanged() {
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
Reconfigure();
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
if (track->kind() != "video") {
LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
<< " track.";
return false;
}
VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
// Detach from old track.
track_->UnregisterObserver(this);
// Attach to new track.
track_ = video_track;
cached_track_enabled_ = track_->enabled();
track_->RegisterObserver(this);
Reconfigure();
return true;
}
void VideoRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (!provider_) {
return;
}
provider_->SetCaptureDevice(ssrc_, nullptr);
provider_->SetVideoSend(ssrc_, false, nullptr);
provider_ = nullptr;
}
void VideoRtpSender::Reconfigure() {
if (!provider_) {
return;
}
const cricket::VideoOptions* options = nullptr;
VideoSourceInterface* source = track_->GetSource();
if (track_->enabled() && source) {
options = source->options();
}
provider_->SetVideoSend(ssrc_, track_->enabled(), options);
}
} // namespace webrtc
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