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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
84 lines
2.8 KiB
C++
84 lines
2.8 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/mediastreamobserver.h"
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#include <algorithm>
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namespace webrtc {
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MediaStreamObserver::MediaStreamObserver(MediaStreamInterface* stream)
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: stream_(stream),
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cached_audio_tracks_(stream->GetAudioTracks()),
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cached_video_tracks_(stream->GetVideoTracks()) {
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stream_->RegisterObserver(this);
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}
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MediaStreamObserver::~MediaStreamObserver() {
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stream_->UnregisterObserver(this);
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}
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void MediaStreamObserver::OnChanged() {
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AudioTrackVector new_audio_tracks = stream_->GetAudioTracks();
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VideoTrackVector new_video_tracks = stream_->GetVideoTracks();
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// Find removed audio tracks.
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for (const auto& cached_track : cached_audio_tracks_) {
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auto it = std::find_if(
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new_audio_tracks.begin(), new_audio_tracks.end(),
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[cached_track](const AudioTrackVector::value_type& new_track) {
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return new_track->id().compare(cached_track->id()) == 0;
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});
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if (it == new_audio_tracks.end()) {
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SignalAudioTrackRemoved(cached_track.get(), stream_);
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}
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}
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// Find added audio tracks.
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for (const auto& new_track : new_audio_tracks) {
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auto it = std::find_if(
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cached_audio_tracks_.begin(), cached_audio_tracks_.end(),
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[new_track](const AudioTrackVector::value_type& cached_track) {
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return new_track->id().compare(cached_track->id()) == 0;
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});
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if (it == cached_audio_tracks_.end()) {
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SignalAudioTrackAdded(new_track.get(), stream_);
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}
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}
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// Find removed video tracks.
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for (const auto& cached_track : cached_video_tracks_) {
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auto it = std::find_if(
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new_video_tracks.begin(), new_video_tracks.end(),
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[cached_track](const VideoTrackVector::value_type& new_track) {
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return new_track->id().compare(cached_track->id()) == 0;
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});
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if (it == new_video_tracks.end()) {
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SignalVideoTrackRemoved(cached_track.get(), stream_);
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}
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}
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// Find added video tracks.
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for (const auto& new_track : new_video_tracks) {
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auto it = std::find_if(
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cached_video_tracks_.begin(), cached_video_tracks_.end(),
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[new_track](const VideoTrackVector::value_type& cached_track) {
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return new_track->id().compare(cached_track->id()) == 0;
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});
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if (it == cached_video_tracks_.end()) {
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SignalVideoTrackAdded(new_track.get(), stream_);
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}
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}
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cached_audio_tracks_ = new_audio_tracks;
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cached_video_tracks_ = new_video_tracks;
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}
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} // namespace webrtc
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