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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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