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Unified Diff: talk/app/webrtc/streamcollection.h

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/streamcollection.h
diff --git a/talk/app/webrtc/streamcollection.h b/talk/app/webrtc/streamcollection.h
deleted file mode 100644
index 07a30a68c8e85dfd5ba368acd0bb9627f3bb5e86..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/streamcollection.h
+++ /dev/null
@@ -1,125 +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.
- */
-
-#ifndef TALK_APP_WEBRTC_STREAMCOLLECTION_H_
-#define TALK_APP_WEBRTC_STREAMCOLLECTION_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/app/webrtc/peerconnectioninterface.h"
-
-namespace webrtc {
-
-// Implementation of StreamCollection.
-class StreamCollection : public StreamCollectionInterface {
- public:
- static rtc::scoped_refptr<StreamCollection> Create() {
- rtc::RefCountedObject<StreamCollection>* implementation =
- new rtc::RefCountedObject<StreamCollection>();
- return implementation;
- }
-
- static rtc::scoped_refptr<StreamCollection> Create(
- StreamCollection* streams) {
- rtc::RefCountedObject<StreamCollection>* implementation =
- new rtc::RefCountedObject<StreamCollection>(streams);
- return implementation;
- }
-
- virtual size_t count() {
- return media_streams_.size();
- }
-
- virtual MediaStreamInterface* at(size_t index) {
- return media_streams_.at(index);
- }
-
- virtual MediaStreamInterface* find(const std::string& label) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(label) == 0) {
- return (*it);
- }
- }
- return NULL;
- }
-
- virtual MediaStreamTrackInterface* FindAudioTrack(
- const std::string& id) {
- for (size_t i = 0; i < media_streams_.size(); ++i) {
- MediaStreamTrackInterface* track = media_streams_[i]->FindAudioTrack(id);
- if (track) {
- return track;
- }
- }
- return NULL;
- }
-
- virtual MediaStreamTrackInterface* FindVideoTrack(
- const std::string& id) {
- for (size_t i = 0; i < media_streams_.size(); ++i) {
- MediaStreamTrackInterface* track = media_streams_[i]->FindVideoTrack(id);
- if (track) {
- return track;
- }
- }
- return NULL;
- }
-
- void AddStream(MediaStreamInterface* stream) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(stream->label()) == 0)
- return;
- }
- media_streams_.push_back(stream);
- }
-
- void RemoveStream(MediaStreamInterface* remove_stream) {
- for (StreamVector::iterator it = media_streams_.begin();
- it != media_streams_.end(); ++it) {
- if ((*it)->label().compare(remove_stream->label()) == 0) {
- media_streams_.erase(it);
- break;
- }
- }
- }
-
- protected:
- StreamCollection() {}
- explicit StreamCollection(StreamCollection* original)
- : media_streams_(original->media_streams_) {
- }
- typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
- StreamVector;
- StreamVector media_streams_;
-};
-
-} // namespace webrtc
-
-#endif // TALK_APP_WEBRTC_STREAMCOLLECTION_H_
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