Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(317)

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

Issue 2741413003: Delete method ModuleRtpRtcpImpl::SendPayloadType. (Closed)
Patch Set: Created 3 years, 9 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_rtcp_impl.h ('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 241 matching lines...) Expand 10 before | Expand all | Expand 10 after
252 void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type, 252 void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
253 const char* payload_name) { 253 const char* payload_name) {
254 RTC_CHECK_EQ( 254 RTC_CHECK_EQ(
255 0, rtp_sender_.RegisterPayload(payload_name, payload_type, 90000, 0, 0)); 255 0, rtp_sender_.RegisterPayload(payload_name, payload_type, 90000, 0, 0));
256 } 256 }
257 257
258 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) { 258 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
259 return rtp_sender_.DeRegisterSendPayload(payload_type); 259 return rtp_sender_.DeRegisterSendPayload(payload_type);
260 } 260 }
261 261
262 int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
263 return rtp_sender_.SendPayloadType();
264 }
265
266 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const { 262 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
267 return rtp_sender_.TimestampOffset(); 263 return rtp_sender_.TimestampOffset();
268 } 264 }
269 265
270 // Configure start timestamp, default is a random number. 266 // Configure start timestamp, default is a random number.
271 void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) { 267 void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
272 rtcp_sender_.SetTimestampOffset(timestamp); 268 rtcp_sender_.SetTimestampOffset(timestamp);
273 rtp_sender_.SetTimestampOffset(timestamp); 269 rtp_sender_.SetTimestampOffset(timestamp);
274 } 270 }
275 271
(...skipping 39 matching lines...) Expand 10 before | Expand all | Expand 10 after
315 } 311 }
316 312
317 // TODO(pbos): Handle media and RTX streams separately (separate RTCP 313 // TODO(pbos): Handle media and RTX streams separately (separate RTCP
318 // feedbacks). 314 // feedbacks).
319 RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() { 315 RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
320 StreamDataCounters rtp_stats; 316 StreamDataCounters rtp_stats;
321 StreamDataCounters rtx_stats; 317 StreamDataCounters rtx_stats;
322 rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats); 318 rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats);
323 319
324 RTCPSender::FeedbackState state; 320 RTCPSender::FeedbackState state;
325 state.send_payload_type = SendPayloadType(); 321 state.send_payload_type = rtp_sender_.SendPayloadType();
326 state.packets_sent = rtp_stats.transmitted.packets + 322 state.packets_sent = rtp_stats.transmitted.packets +
327 rtx_stats.transmitted.packets; 323 rtx_stats.transmitted.packets;
328 state.media_bytes_sent = rtp_stats.transmitted.payload_bytes + 324 state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
329 rtx_stats.transmitted.payload_bytes; 325 rtx_stats.transmitted.payload_bytes;
330 state.module = this; 326 state.module = this;
331 327
332 LastReceivedNTP(&state.last_rr_ntp_secs, 328 LastReceivedNTP(&state.last_rr_ntp_secs,
333 &state.last_rr_ntp_frac, 329 &state.last_rr_ntp_frac,
334 &state.remote_sr); 330 &state.remote_sr);
335 331
(...skipping 542 matching lines...) Expand 10 before | Expand all | Expand 10 after
878 StreamDataCountersCallback* 874 StreamDataCountersCallback*
879 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 875 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
880 return rtp_sender_.GetRtpStatisticsCallback(); 876 return rtp_sender_.GetRtpStatisticsCallback();
881 } 877 }
882 878
883 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( 879 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
884 const BitrateAllocation& bitrate) { 880 const BitrateAllocation& bitrate) {
885 rtcp_sender_.SetVideoBitrateAllocation(bitrate); 881 rtcp_sender_.SetVideoBitrateAllocation(bitrate);
886 } 882 }
887 } // namespace webrtc 883 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698