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Unified Diff: talk/app/webrtc/mediastream.cc

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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Index: talk/app/webrtc/mediastream.cc
diff --git a/talk/app/webrtc/mediastream.cc b/talk/app/webrtc/mediastream.cc
deleted file mode 100644
index 0d206300dfff0f8eb1a8c07086f25563b62ccade..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/mediastream.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * libjingle
- * Copyright 2011 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/mediastream.h"
-#include "webrtc/base/logging.h"
-
-namespace webrtc {
-
-template <class V>
-static typename V::iterator FindTrack(V* vector,
- const std::string& track_id) {
- typename V::iterator it = vector->begin();
- for (; it != vector->end(); ++it) {
- if ((*it)->id() == track_id) {
- break;
- }
- }
- return it;
-};
-
-rtc::scoped_refptr<MediaStream> MediaStream::Create(
- const std::string& label) {
- rtc::RefCountedObject<MediaStream>* stream =
- new rtc::RefCountedObject<MediaStream>(label);
- return stream;
-}
-
-MediaStream::MediaStream(const std::string& label)
- : label_(label) {
-}
-
-bool MediaStream::AddTrack(AudioTrackInterface* track) {
- return AddTrack<AudioTrackVector, AudioTrackInterface>(&audio_tracks_, track);
-}
-
-bool MediaStream::AddTrack(VideoTrackInterface* track) {
- return AddTrack<VideoTrackVector, VideoTrackInterface>(&video_tracks_, track);
-}
-
-bool MediaStream::RemoveTrack(AudioTrackInterface* track) {
- return RemoveTrack<AudioTrackVector>(&audio_tracks_, track);
-}
-
-bool MediaStream::RemoveTrack(VideoTrackInterface* track) {
- return RemoveTrack<VideoTrackVector>(&video_tracks_, track);
-}
-
-rtc::scoped_refptr<AudioTrackInterface>
-MediaStream::FindAudioTrack(const std::string& track_id) {
- AudioTrackVector::iterator it = FindTrack(&audio_tracks_, track_id);
- if (it == audio_tracks_.end())
- return NULL;
- return *it;
-}
-
-rtc::scoped_refptr<VideoTrackInterface>
-MediaStream::FindVideoTrack(const std::string& track_id) {
- VideoTrackVector::iterator it = FindTrack(&video_tracks_, track_id);
- if (it == video_tracks_.end())
- return NULL;
- return *it;
-}
-
-template <typename TrackVector, typename Track>
-bool MediaStream::AddTrack(TrackVector* tracks, Track* track) {
- typename TrackVector::iterator it = FindTrack(tracks, track->id());
- if (it != tracks->end())
- return false;
- tracks->push_back(track);
- FireOnChanged();
- return true;
-}
-
-template <typename TrackVector>
-bool MediaStream::RemoveTrack(TrackVector* tracks,
- MediaStreamTrackInterface* track) {
- ASSERT(tracks != NULL);
- if (!track)
- return false;
- typename TrackVector::iterator it = FindTrack(tracks, track->id());
- if (it == tracks->end())
- return false;
- tracks->erase(it);
- FireOnChanged();
- return true;
-}
-
-} // namespace webrtc
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