Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(6)

Side by Side Diff: webrtc/video/video_receive_stream.cc

Issue 1412143002: Remove the video channel id completely. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Remove string. Created 5 years, 2 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/video/video_receive_stream.h ('k') | webrtc/video/video_send_stream.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 122 matching lines...) Expand 10 before | Expand all | Expand 10 after
133 codec.height = 180; 133 codec.height = 180;
134 codec.startBitrate = codec.minBitrate = codec.maxBitrate = 134 codec.startBitrate = codec.minBitrate = codec.maxBitrate =
135 Call::Config::kDefaultStartBitrateBps / 1000; 135 Call::Config::kDefaultStartBitrateBps / 1000;
136 136
137 return codec; 137 return codec;
138 } 138 }
139 } // namespace 139 } // namespace
140 140
141 VideoReceiveStream::VideoReceiveStream(int num_cpu_cores, 141 VideoReceiveStream::VideoReceiveStream(int num_cpu_cores,
142 ChannelGroup* channel_group, 142 ChannelGroup* channel_group,
143 int channel_id,
144 const VideoReceiveStream::Config& config, 143 const VideoReceiveStream::Config& config,
145 webrtc::VoiceEngine* voice_engine, 144 webrtc::VoiceEngine* voice_engine,
146 ProcessThread* process_thread) 145 ProcessThread* process_thread)
147 : transport_adapter_(config.rtcp_send_transport), 146 : transport_adapter_(config.rtcp_send_transport),
148 encoded_frame_proxy_(config.pre_decode_callback), 147 encoded_frame_proxy_(config.pre_decode_callback),
149 config_(config), 148 config_(config),
150 clock_(Clock::GetRealTimeClock()), 149 clock_(Clock::GetRealTimeClock()),
151 channel_group_(channel_group) { 150 channel_group_(channel_group) {
152 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); 151 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
153 152
(...skipping 212 matching lines...) Expand 10 before | Expand all | Expand 10 after
366 return 0; 365 return 0;
367 } 366 }
368 367
369 void VideoReceiveStream::SignalNetworkState(NetworkState state) { 368 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
370 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode 369 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
371 : RtcpMode::kOff); 370 : RtcpMode::kOff);
372 } 371 }
373 372
374 } // namespace internal 373 } // namespace internal
375 } // namespace webrtc 374 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/video/video_receive_stream.h ('k') | webrtc/video/video_send_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698