rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/api/rtpsender.cc

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2016-06-21 20:13:05 +00:00
/*
* Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/api/rtpsender.h"
#include "webrtc/api/localaudiosource.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/trace_event.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,
size_t 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::AudioSource::Sink* sink) {
rtc::CritScope lock(&lock_);
ASSERT(!sink || !sink_);
sink_ = sink;
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
const std::string& stream_id,
AudioProviderInterface* provider,
StatsCollector* stats)
: id_(track->id()),
stream_id_(stream_id),
provider_(provider),
stats_(stats),
track_(track),
cached_track_enabled_(track->enabled()),
sink_adapter_(new LocalAudioSinkAdapter()) {
RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
AudioProviderInterface* provider,
StatsCollector* stats)
: id_(track->id()),
stream_id_(rtc::CreateRandomUuid()),
provider_(provider),
stats_(stats),
track_(track),
cached_track_enabled_(track->enabled()),
sink_adapter_(new LocalAudioSinkAdapter()) {
RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
}
AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
StatsCollector* stats)
: id_(rtc::CreateRandomUuid()),
stream_id_(rtc::CreateRandomUuid()),
provider_(provider),
stats_(stats),
sink_adapter_(new LocalAudioSinkAdapter()) {}
AudioRtpSender::~AudioRtpSender() {
Stop();
}
void AudioRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "AudioRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
if (can_send_track()) {
SetAudioSend();
}
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetTrack");
if (stopped_) {
LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
}
if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
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.
if (track_) {
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
}
if (can_send_track() && stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
// Attach to new track.
bool prev_can_send_track = can_send_track();
// Keep a reference to the old track to keep it alive until we call
// SetAudioSend.
rtc::scoped_refptr<AudioTrackInterface> old_track = track_;
track_ = audio_track;
if (track_) {
cached_track_enabled_ = track_->enabled();
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
}
// Update audio provider.
if (can_send_track()) {
SetAudioSend();
if (stats_) {
stats_->AddLocalAudioTrack(track_.get(), ssrc_);
}
} else if (prev_can_send_track) {
cricket::AudioOptions options;
provider_->SetAudioSend(ssrc_, false, options, nullptr);
}
return true;
}
RtpParameters AudioRtpSender::GetParameters() const {
return provider_->GetAudioRtpSendParameters(ssrc_);
}
bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
return provider_->SetAudioRtpSendParameters(ssrc_, parameters);
}
void AudioRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "AudioRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
// If we are already sending with a particular SSRC, stop sending.
if (can_send_track()) {
cricket::AudioOptions options;
provider_->SetAudioSend(ssrc_, false, options, nullptr);
if (stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
}
ssrc_ = ssrc;
if (can_send_track()) {
SetAudioSend();
if (stats_) {
stats_->AddLocalAudioTrack(track_.get(), ssrc_);
}
}
}
void AudioRtpSender::Stop() {
TRACE_EVENT0("webrtc", "AudioRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
}
if (track_) {
track_->RemoveSink(sink_adapter_.get());
track_->UnregisterObserver(this);
}
if (can_send_track()) {
cricket::AudioOptions options;
provider_->SetAudioSend(ssrc_, false, options, nullptr);
if (stats_) {
stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
}
}
stopped_ = true;
}
void AudioRtpSender::SetAudioSend() {
RTC_DCHECK(!stopped_ && can_send_track());
cricket::AudioOptions options;
#if !defined(WEBRTC_CHROMIUM_BUILD)
// TODO(tommi): Remove this hack when we move CreateAudioSource out of
// PeerConnection. This is a bit of a strange way to apply local audio
// options since it is also applied to all streams/channels, local or remote.
if (track_->enabled() && track_->GetSource() &&
!track_->GetSource()->remote()) {
// TODO(xians): Remove this static_cast since we should be able to connect
// a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
#endif
cricket::AudioSource* source = sink_adapter_.get();
ASSERT(source != nullptr);
provider_->SetAudioSend(ssrc_, track_->enabled(), options, source);
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
const std::string& stream_id,
VideoProviderInterface* provider)
: id_(track->id()),
stream_id_(stream_id),
provider_(provider),
track_(track),
cached_track_enabled_(track->enabled()) {
RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
VideoProviderInterface* provider)
: id_(track->id()),
stream_id_(rtc::CreateRandomUuid()),
provider_(provider),
track_(track),
cached_track_enabled_(track->enabled()) {
RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
}
VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
: id_(rtc::CreateRandomUuid()),
stream_id_(rtc::CreateRandomUuid()),
provider_(provider) {}
VideoRtpSender::~VideoRtpSender() {
Stop();
}
void VideoRtpSender::OnChanged() {
TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
if (can_send_track()) {
SetVideoSend();
}
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetTrack");
if (stopped_) {
LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
return false;
}
if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
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.
if (track_) {
track_->UnregisterObserver(this);
}
// Attach to new track.
bool prev_can_send_track = can_send_track();
// Keep a reference to the old track to keep it alive until we call
// SetVideoSend.
rtc::scoped_refptr<VideoTrackInterface> old_track = track_;
track_ = video_track;
if (track_) {
cached_track_enabled_ = track_->enabled();
track_->RegisterObserver(this);
}
// Update video provider.
if (can_send_track()) {
SetVideoSend();
} else if (prev_can_send_track) {
ClearVideoSend();
}
return true;
}
RtpParameters VideoRtpSender::GetParameters() const {
return provider_->GetVideoRtpSendParameters(ssrc_);
}
bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
return provider_->SetVideoRtpSendParameters(ssrc_, parameters);
}
void VideoRtpSender::SetSsrc(uint32_t ssrc) {
TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc");
if (stopped_ || ssrc == ssrc_) {
return;
}
// If we are already sending with a particular SSRC, stop sending.
if (can_send_track()) {
ClearVideoSend();
}
ssrc_ = ssrc;
if (can_send_track()) {
SetVideoSend();
}
}
void VideoRtpSender::Stop() {
TRACE_EVENT0("webrtc", "VideoRtpSender::Stop");
// TODO(deadbeef): Need to do more here to fully stop sending packets.
if (stopped_) {
return;
}
if (track_) {
track_->UnregisterObserver(this);
}
if (can_send_track()) {
ClearVideoSend();
}
stopped_ = true;
}
void VideoRtpSender::SetVideoSend() {
RTC_DCHECK(!stopped_ && can_send_track());
cricket::VideoOptions options;
VideoTrackSourceInterface* source = track_->GetSource();
if (source) {
options.is_screencast = rtc::Optional<bool>(source->is_screencast());
options.video_noise_reduction = source->needs_denoising();
}
provider_->SetVideoSend(ssrc_, track_->enabled(), &options, track_);
}
void VideoRtpSender::ClearVideoSend() {
RTC_DCHECK(ssrc_ != 0);
RTC_DCHECK(provider_ != nullptr);
provider_->SetVideoSend(ssrc_, false, nullptr, nullptr);
}
} // namespace webrtc