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

Issue 2491293002: Make FlexFEC packets paceable through RTPSender. (Closed)
Patch Set: Feedback response 3. Created 4 years, 1 month ago
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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
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222 222
223 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) { 223 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
224 rtp_sender_.SetRtxSsrc(ssrc); 224 rtp_sender_.SetRtxSsrc(ssrc);
225 } 225 }
226 226
227 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type, 227 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
228 int associated_payload_type) { 228 int associated_payload_type) {
229 rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type); 229 rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
230 } 230 }
231 231
232 rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
233 return rtp_sender_.FlexfecSsrc();
234 }
235
232 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket( 236 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
233 const uint8_t* rtcp_packet, 237 const uint8_t* rtcp_packet,
234 const size_t length) { 238 const size_t length) {
235 return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1; 239 return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
236 } 240 }
237 241
238 int32_t ModuleRtpRtcpImpl::RegisterSendPayload( 242 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
239 const CodecInst& voice_codec) { 243 const CodecInst& voice_codec) {
240 return rtp_sender_.RegisterPayload( 244 return rtp_sender_.RegisterPayload(
241 voice_codec.plname, voice_codec.pltype, voice_codec.plfreq, 245 voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
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393 return rtp_sender_.SendOutgoingData( 397 return rtp_sender_.SendOutgoingData(
394 frame_type, payload_type, time_stamp, capture_time_ms, payload_data, 398 frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
395 payload_size, fragmentation, rtp_video_header, transport_frame_id_out); 399 payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
396 } 400 }
397 401
398 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, 402 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
399 uint16_t sequence_number, 403 uint16_t sequence_number,
400 int64_t capture_time_ms, 404 int64_t capture_time_ms,
401 bool retransmission, 405 bool retransmission,
402 int probe_cluster_id) { 406 int probe_cluster_id) {
403 if (SendingMedia() && ssrc == rtp_sender_.SSRC()) { 407 return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
404 return rtp_sender_.TimeToSendPacket(sequence_number, capture_time_ms, 408 retransmission, probe_cluster_id);
405 retransmission, probe_cluster_id);
406 }
407 // No RTP sender is interested in sending this packet.
408 return true;
409 } 409 }
410 410
411 size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes, 411 size_t ModuleRtpRtcpImpl::TimeToSendPadding(size_t bytes,
412 int probe_cluster_id) { 412 int probe_cluster_id) {
413 return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id); 413 return rtp_sender_.TimeToSendPadding(bytes, probe_cluster_id);
414 } 414 }
415 415
416 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const { 416 uint16_t ModuleRtpRtcpImpl::MaxPayloadLength() const {
417 return rtp_sender_.MaxPayloadLength(); 417 return rtp_sender_.MaxPayloadLength();
418 } 418 }
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945 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 945 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
946 StreamDataCountersCallback* callback) { 946 StreamDataCountersCallback* callback) {
947 rtp_sender_.RegisterRtpStatisticsCallback(callback); 947 rtp_sender_.RegisterRtpStatisticsCallback(callback);
948 } 948 }
949 949
950 StreamDataCountersCallback* 950 StreamDataCountersCallback*
951 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 951 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
952 return rtp_sender_.GetRtpStatisticsCallback(); 952 return rtp_sender_.GetRtpStatisticsCallback();
953 } 953 }
954 } // namespace webrtc 954 } // namespace webrtc
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