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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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93 rtcp_receiver_(configuration.clock, | 93 rtcp_receiver_(configuration.clock, |
94 configuration.receiver_only, | 94 configuration.receiver_only, |
95 configuration.rtcp_packet_type_counter_observer, | 95 configuration.rtcp_packet_type_counter_observer, |
96 configuration.bandwidth_callback, | 96 configuration.bandwidth_callback, |
97 configuration.intra_frame_callback, | 97 configuration.intra_frame_callback, |
98 configuration.transport_feedback_callback, | 98 configuration.transport_feedback_callback, |
99 configuration.bitrate_allocation_observer, | 99 configuration.bitrate_allocation_observer, |
100 this), | 100 this), |
101 clock_(configuration.clock), | 101 clock_(configuration.clock), |
102 audio_(configuration.audio), | 102 audio_(configuration.audio), |
| 103 collision_detected_(false), |
103 last_process_time_(configuration.clock->TimeInMilliseconds()), | 104 last_process_time_(configuration.clock->TimeInMilliseconds()), |
104 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()), | 105 last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()), |
105 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()), | 106 last_rtt_process_time_(configuration.clock->TimeInMilliseconds()), |
106 packet_overhead_(28), // IPV4 UDP. | 107 packet_overhead_(28), // IPV4 UDP. |
107 nack_last_time_sent_full_(0), | 108 nack_last_time_sent_full_(0), |
108 nack_last_time_sent_full_prev_(0), | 109 nack_last_time_sent_full_prev_(0), |
109 nack_last_seq_number_sent_(0), | 110 nack_last_seq_number_sent_(0), |
110 key_frame_req_method_(kKeyFrameReqPliRtcp), | 111 key_frame_req_method_(kKeyFrameReqPliRtcp), |
111 remote_bitrate_(configuration.remote_bitrate_estimator), | 112 remote_bitrate_(configuration.remote_bitrate_estimator), |
112 rtt_stats_(configuration.rtt_stats), | 113 rtt_stats_(configuration.rtt_stats), |
113 rtt_ms_(0) { | 114 rtt_ms_(0) { |
| 115 // Make sure that RTCP objects are aware of our SSRC. |
| 116 uint32_t SSRC = rtp_sender_.SSRC(); |
| 117 rtcp_sender_.SetSSRC(SSRC); |
| 118 SetRtcpReceiverSsrcs(SSRC); |
| 119 |
114 // Make sure rtcp sender use same timestamp offset as rtp sender. | 120 // Make sure rtcp sender use same timestamp offset as rtp sender. |
115 rtcp_sender_.SetTimestampOffset(rtp_sender_.TimestampOffset()); | 121 rtcp_sender_.SetTimestampOffset(rtp_sender_.TimestampOffset()); |
116 | 122 |
117 // Set default packet size limit. | 123 // Set default packet size limit. |
118 // TODO(nisse): Kind-of duplicates | 124 // TODO(nisse): Kind-of duplicates |
119 // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize. | 125 // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize. |
120 const size_t kTcpOverIpv4HeaderSize = 40; | 126 const size_t kTcpOverIpv4HeaderSize = 40; |
121 SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize); | 127 SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize); |
122 } | 128 } |
123 | 129 |
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342 BitrateSent(&state.send_bitrate, &tmp, &tmp, &tmp); | 348 BitrateSent(&state.send_bitrate, &tmp, &tmp, &tmp); |
343 return state; | 349 return state; |
344 } | 350 } |
345 | 351 |
346 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { | 352 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { |
347 if (rtcp_sender_.Sending() != sending) { | 353 if (rtcp_sender_.Sending() != sending) { |
348 // Sends RTCP BYE when going from true to false | 354 // Sends RTCP BYE when going from true to false |
349 if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { | 355 if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { |
350 LOG(LS_WARNING) << "Failed to send RTCP BYE"; | 356 LOG(LS_WARNING) << "Failed to send RTCP BYE"; |
351 } | 357 } |
| 358 |
| 359 collision_detected_ = false; |
| 360 |
| 361 // Generate a new SSRC for the next "call" if false |
| 362 rtp_sender_.SetSendingStatus(sending); |
| 363 |
| 364 // Make sure that RTCP objects are aware of our SSRC (it could have changed |
| 365 // Due to collision) |
| 366 uint32_t SSRC = rtp_sender_.SSRC(); |
| 367 rtcp_sender_.SetSSRC(SSRC); |
| 368 SetRtcpReceiverSsrcs(SSRC); |
| 369 |
| 370 return 0; |
352 } | 371 } |
353 return 0; | 372 return 0; |
354 } | 373 } |
355 | 374 |
356 bool ModuleRtpRtcpImpl::Sending() const { | 375 bool ModuleRtpRtcpImpl::Sending() const { |
357 return rtcp_sender_.Sending(); | 376 return rtcp_sender_.Sending(); |
358 } | 377 } |
359 | 378 |
360 void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { | 379 void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { |
361 rtp_sender_.SetSendingMediaStatus(sending); | 380 rtp_sender_.SetSendingMediaStatus(sending); |
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768 bool ModuleRtpRtcpImpl::SetFecParameters( | 787 bool ModuleRtpRtcpImpl::SetFecParameters( |
769 const FecProtectionParams& delta_params, | 788 const FecProtectionParams& delta_params, |
770 const FecProtectionParams& key_params) { | 789 const FecProtectionParams& key_params) { |
771 return rtp_sender_.SetFecParameters(delta_params, key_params); | 790 return rtp_sender_.SetFecParameters(delta_params, key_params); |
772 } | 791 } |
773 | 792 |
774 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { | 793 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { |
775 // Inform about the incoming SSRC. | 794 // Inform about the incoming SSRC. |
776 rtcp_sender_.SetRemoteSSRC(ssrc); | 795 rtcp_sender_.SetRemoteSSRC(ssrc); |
777 rtcp_receiver_.SetRemoteSSRC(ssrc); | 796 rtcp_receiver_.SetRemoteSSRC(ssrc); |
| 797 |
| 798 // Check for a SSRC collision. |
| 799 if (rtp_sender_.SSRC() == ssrc && !collision_detected_) { |
| 800 // If we detect a collision change the SSRC but only once. |
| 801 collision_detected_ = true; |
| 802 uint32_t new_ssrc = rtp_sender_.GenerateNewSSRC(); |
| 803 if (new_ssrc == 0) { |
| 804 // Configured via API ignore. |
| 805 return; |
| 806 } |
| 807 if (RtcpMode::kOff != rtcp_sender_.Status()) { |
| 808 // Send RTCP bye on the current SSRC. |
| 809 SendRTCP(kRtcpBye); |
| 810 } |
| 811 // Change local SSRC and inform all objects about the new SSRC. |
| 812 rtcp_sender_.SetSSRC(new_ssrc); |
| 813 SetRtcpReceiverSsrcs(new_ssrc); |
| 814 } |
778 } | 815 } |
779 | 816 |
780 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, | 817 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, |
781 uint32_t* video_rate, | 818 uint32_t* video_rate, |
782 uint32_t* fec_rate, | 819 uint32_t* fec_rate, |
783 uint32_t* nack_rate) const { | 820 uint32_t* nack_rate) const { |
784 *total_rate = rtp_sender_.BitrateSent(); | 821 *total_rate = rtp_sender_.BitrateSent(); |
785 *video_rate = rtp_sender_.VideoBitrateSent(); | 822 *video_rate = rtp_sender_.VideoBitrateSent(); |
786 *fec_rate = rtp_sender_.FecOverheadRate(); | 823 *fec_rate = rtp_sender_.FecOverheadRate(); |
787 *nack_rate = rtp_sender_.NackOverheadRate(); | 824 *nack_rate = rtp_sender_.NackOverheadRate(); |
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876 StreamDataCountersCallback* | 913 StreamDataCountersCallback* |
877 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 914 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
878 return rtp_sender_.GetRtpStatisticsCallback(); | 915 return rtp_sender_.GetRtpStatisticsCallback(); |
879 } | 916 } |
880 | 917 |
881 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( | 918 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( |
882 const BitrateAllocation& bitrate) { | 919 const BitrateAllocation& bitrate) { |
883 rtcp_sender_.SetVideoBitrateAllocation(bitrate); | 920 rtcp_sender_.SetVideoBitrateAllocation(bitrate); |
884 } | 921 } |
885 } // namespace webrtc | 922 } // namespace webrtc |
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