Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(3)

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

Issue 2969623003: Update includes for webrtc/{base => rtc_base} rename (2/3) (Closed)
Patch Set: Rebased onto 224e65939af87443addfc5bb500fbf434728bd1c and restored sorting in clock.cc Created 3 years, 5 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/payload_router.h ('k') | webrtc/video/quality_threshold.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) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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
11 #include "webrtc/video/payload_router.h" 11 #include "webrtc/video/payload_router.h"
12 12
13 #include "webrtc/base/checks.h"
14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" 13 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
16 #include "webrtc/modules/video_coding/include/video_codec_interface.h" 15 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
16 #include "webrtc/rtc_base/checks.h"
17 17
18 namespace webrtc { 18 namespace webrtc {
19 19
20 namespace { 20 namespace {
21 // Map information from info into rtp. 21 // Map information from info into rtp.
22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) { 22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
23 RTC_DCHECK(info); 23 RTC_DCHECK(info);
24 switch (info->codecType) { 24 switch (info->codecType) {
25 case kVideoCodecVP8: { 25 case kVideoCodecVP8: {
26 rtp->codec = kRtpVideoVp8; 26 rtp->codec = kRtpVideoVp8;
(...skipping 151 matching lines...) Expand 10 before | Expand all | Expand 10 after
178 BitrateAllocation layer_bitrate; 178 BitrateAllocation layer_bitrate;
179 for (int tl = 0; tl < kMaxTemporalStreams; ++tl) 179 for (int tl = 0; tl < kMaxTemporalStreams; ++tl)
180 layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl)); 180 layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl));
181 rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate); 181 rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate);
182 } 182 }
183 } 183 }
184 } 184 }
185 } 185 }
186 186
187 } // namespace webrtc 187 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/video/payload_router.h ('k') | webrtc/video/quality_threshold.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698