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

Issue 1527003002: [rtp_rtcp] RtcpReceiverTest rewritten using public available interface (observers) instead intermid… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 10 months 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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218 } 218 }
219 219
220 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type, 220 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
221 int associated_payload_type) { 221 int associated_payload_type) {
222 rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type); 222 rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
223 } 223 }
224 224
225 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket( 225 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
226 const uint8_t* rtcp_packet, 226 const uint8_t* rtcp_packet,
227 const size_t length) { 227 const size_t length) {
228 // Allow receive of non-compound RTCP packets. 228 return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
229 RTCPUtility::RTCPParserV2 rtcp_parser(rtcp_packet, length, true);
230
231 const bool valid_rtcpheader = rtcp_parser.IsValid();
232 if (!valid_rtcpheader) {
233 LOG(LS_WARNING) << "Incoming invalid RTCP packet";
234 return -1;
235 }
236 RTCPHelp::RTCPPacketInformation rtcp_packet_information;
237 int32_t ret_val = rtcp_receiver_.IncomingRTCPPacket(
238 rtcp_packet_information, &rtcp_parser);
239 if (ret_val == 0) {
240 rtcp_receiver_.TriggerCallbacksFromRTCPPacket(rtcp_packet_information);
241 }
242 return ret_val;
243 } 229 }
244 230
245 int32_t ModuleRtpRtcpImpl::RegisterSendPayload( 231 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
246 const CodecInst& voice_codec) { 232 const CodecInst& voice_codec) {
247 return rtp_sender_.RegisterPayload( 233 return rtp_sender_.RegisterPayload(
248 voice_codec.plname, 234 voice_codec.plname,
249 voice_codec.pltype, 235 voice_codec.pltype,
250 voice_codec.plfreq, 236 voice_codec.plfreq,
251 voice_codec.channels, 237 voice_codec.channels,
252 (voice_codec.rate < 0) ? 0 : voice_codec.rate); 238 (voice_codec.rate < 0) ? 0 : voice_codec.rate);
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978 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 964 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
979 StreamDataCountersCallback* callback) { 965 StreamDataCountersCallback* callback) {
980 rtp_sender_.RegisterRtpStatisticsCallback(callback); 966 rtp_sender_.RegisterRtpStatisticsCallback(callback);
981 } 967 }
982 968
983 StreamDataCountersCallback* 969 StreamDataCountersCallback*
984 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 970 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
985 return rtp_sender_.GetRtpStatisticsCallback(); 971 return rtp_sender_.GetRtpStatisticsCallback();
986 } 972 }
987 } // namespace webrtc 973 } // namespace webrtc
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