Files
platform-external-webrtc/webrtc/pc/videotracksource.cc
ossu 7bb87ee4e8 Create //webrtc/api:libjingle_peerconnection_api + refactorings.
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.

Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.

Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.

BUG=webrtc:5883

Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
2017-01-23 12:56:25 +00:00

54 lines
1.4 KiB
C++

/*
* Copyright 2016 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/pc/videotracksource.h"
#include <string>
namespace webrtc {
VideoTrackSource::VideoTrackSource(
rtc::VideoSourceInterface<VideoFrame>* source,
bool remote)
: source_(source), state_(kInitializing), remote_(remote) {
worker_thread_checker_.DetachFromThread();
}
void VideoTrackSource::SetState(SourceState new_state) {
if (state_ != new_state) {
state_ = new_state;
FireOnChanged();
}
}
void VideoTrackSource::OnSourceDestroyed() {
source_ = nullptr;
}
void VideoTrackSource::AddOrUpdateSink(
rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!source_) {
return;
}
source_->AddOrUpdateSink(sink, wants);
}
void VideoTrackSource::RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!source_) {
return;
}
source_->RemoveSink(sink);
}
} // namespace webrtc