Chromium Code Reviews| Index: webrtc/voice_engine/channel.cc | 
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc | 
| index d630575cbcb292d5787326a9c364095fa5891d1e..9158d362c684104a913d2dde6e7c26877e98f926 100644 | 
| --- a/webrtc/voice_engine/channel.cc | 
| +++ b/webrtc/voice_engine/channel.cc | 
| @@ -373,16 +373,17 @@ class VoERtcpObserver : public RtcpBandwidthObserver { | 
| // to calculate the number of RTP packets this report refers to. Ignore if | 
| // we haven't seen this SSRC before. | 
| std::map<uint32_t, uint32_t>::iterator seq_num_it = | 
| - extended_max_sequence_number_.find(block_it->sourceSSRC); | 
| + extended_max_sequence_number_.find(block_it->source_ssrc); | 
| int number_of_packets = 0; | 
| if (seq_num_it != extended_max_sequence_number_.end()) { | 
| - number_of_packets = block_it->extendedHighSeqNum - seq_num_it->second; | 
| + number_of_packets = | 
| + block_it->extended_highest_sequence_number - seq_num_it->second; | 
| } | 
| - fraction_lost_aggregate += number_of_packets * block_it->fractionLost; | 
| + fraction_lost_aggregate += number_of_packets * block_it->fraction_lost; | 
| total_number_of_packets += number_of_packets; | 
| - extended_max_sequence_number_[block_it->sourceSSRC] = | 
| - block_it->extendedHighSeqNum; | 
| + extended_max_sequence_number_[block_it->source_ssrc] = | 
| + block_it->extended_highest_sequence_number; | 
| } | 
| int weighted_fraction_lost = 0; | 
| if (total_number_of_packets > 0) { | 
| @@ -2648,14 +2649,15 @@ int Channel::GetRemoteRTCPReportBlocks( | 
| std::vector<RTCPReportBlock>::const_iterator it = rtcp_report_blocks.begin(); | 
| for (; it != rtcp_report_blocks.end(); ++it) { | 
| ReportBlock report_block; | 
| - report_block.sender_SSRC = it->remoteSSRC; | 
| - report_block.source_SSRC = it->sourceSSRC; | 
| - report_block.fraction_lost = it->fractionLost; | 
| - report_block.cumulative_num_packets_lost = it->cumulativeLost; | 
| - report_block.extended_highest_sequence_number = it->extendedHighSeqNum; | 
| + report_block.sender_SSRC = it->sender_ssrc; | 
| + report_block.source_SSRC = it->source_ssrc; | 
| + report_block.fraction_lost = it->fraction_lost; | 
| + report_block.cumulative_num_packets_lost = it->packets_lost; | 
| + report_block.extended_highest_sequence_number = | 
| + it->extended_highest_sequence_number; | 
| report_block.interarrival_jitter = it->jitter; | 
| - report_block.last_SR_timestamp = it->lastSR; | 
| - report_block.delay_since_last_SR = it->delaySinceLastSR; | 
| + report_block.last_SR_timestamp = it->last_sender_report_timestamp; | 
| + report_block.delay_since_last_SR = it->delay_since_last_sender_report; | 
| report_blocks->push_back(report_block); | 
| } | 
| return 0; | 
| @@ -3145,7 +3147,7 @@ int64_t Channel::GetRTT(bool allow_associate_channel) const { | 
| uint32_t remoteSSRC = rtp_receiver_->SSRC(); | 
| 
 
stefan-webrtc
2017/08/07 10:43:24
Would be nice to also change this to sender_ssrc.
 
srte
2017/08/07 12:17:30
While that would not be wrong, I'm not sure it'd a
 
 | 
| std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin(); | 
| for (; it != report_blocks.end(); ++it) { | 
| - if (it->remoteSSRC == remoteSSRC) | 
| + if (it->sender_ssrc == remoteSSRC) | 
| break; | 
| } | 
| if (it == report_blocks.end()) { | 
| @@ -3153,7 +3155,7 @@ int64_t Channel::GetRTT(bool allow_associate_channel) const { | 
| // To calculate RTT we try with the SSRC of the first report block. | 
| // This is very important for send-only channels where we don't know | 
| // the SSRC of the other end. | 
| - remoteSSRC = report_blocks[0].remoteSSRC; | 
| + remoteSSRC = report_blocks[0].sender_ssrc; | 
| } | 
| int64_t avg_rtt = 0; |