Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(774)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

Issue 2431253003: Remove now unused code in RtpHeaderExtensionMap (Closed)
Patch Set: Created 4 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/modules/rtp_rtcp/source/rtp_sender_unittest.cc ('k') | no next file » | 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) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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 229 matching lines...) Expand 10 before | Expand all | Expand 10 after
240 // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ 240 // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
241 // ts_126114v120700p.pdf Section 7.4.5: 241 // ts_126114v120700p.pdf Section 7.4.5:
242 // The MTSI client shall add the payload bytes as defined in this clause 242 // The MTSI client shall add the payload bytes as defined in this clause
243 // onto the last RTP packet in each group of packets which make up a key 243 // onto the last RTP packet in each group of packets which make up a key
244 // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265 244 // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
245 // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP 245 // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
246 // packet in each group of packets which make up another type of frame 246 // packet in each group of packets which make up another type of frame
247 // (e.g. a P-Frame) only if the current value is different from the previous 247 // (e.g. a P-Frame) only if the current value is different from the previous
248 // value sent. 248 // value sent.
249 // Here we are adding it to every packet of every frame at this point. 249 // Here we are adding it to every packet of every frame at this point.
250 if (video_header && video_header->rotation != kVideoRotation_0) { 250 if (video_header && video_header->rotation != kVideoRotation_0)
sprang_webrtc 2016/10/24 13:48:10 Just to be clear, how exactly is this rotation ext
danilchap 2016/10/24 14:27:25 No rotation extension means it is 0. Before my ref
sprang_webrtc 2016/10/25 07:47:00 Acknowledged. I interpreted the previous "rtp_sen
251 // TODO(danilchap): Remove next call together with concept
252 // of inactive extension. Now it helps to calulate total maximum size
253 // or rtp header extensions that is used in FECPacketOverhead() function.
254 rtp_sender_->ActivateCVORtpHeaderExtension();
255 rtp_header->SetExtension<VideoOrientation>(video_header->rotation); 251 rtp_header->SetExtension<VideoOrientation>(video_header->rotation);
256 }
257 252
258 size_t packet_capacity = rtp_sender_->MaxPayloadLength() - 253 size_t packet_capacity = rtp_sender_->MaxPayloadLength() -
259 FecPacketOverhead() - 254 FecPacketOverhead() -
260 (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0); 255 (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
261 RTC_DCHECK_LE(packet_capacity, rtp_header->capacity()); 256 RTC_DCHECK_LE(packet_capacity, rtp_header->capacity());
262 RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size()); 257 RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size());
263 size_t max_data_payload_length = packet_capacity - rtp_header->headers_size(); 258 size_t max_data_payload_length = packet_capacity - rtp_header->headers_size();
264 259
265 std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create( 260 std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
266 video_type, max_data_payload_length, 261 video_type, max_data_payload_length,
(...skipping 74 matching lines...) Expand 10 before | Expand all | Expand 10 after
341 rtc::CritScope cs(&crit_); 336 rtc::CritScope cs(&crit_);
342 return retransmission_settings_; 337 return retransmission_settings_;
343 } 338 }
344 339
345 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) { 340 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
346 rtc::CritScope cs(&crit_); 341 rtc::CritScope cs(&crit_);
347 retransmission_settings_ = settings; 342 retransmission_settings_ = settings;
348 } 343 }
349 344
350 } // namespace webrtc 345 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698