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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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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1335 if (last_stats_log_ms_ == -1 || | 1335 if (last_stats_log_ms_ == -1 || |
1336 now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) { | 1336 now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) { |
1337 last_stats_log_ms_ = now_ms; | 1337 last_stats_log_ms_ = now_ms; |
1338 log_stats = true; | 1338 log_stats = true; |
1339 } | 1339 } |
1340 | 1340 |
1341 info->Clear(); | 1341 info->Clear(); |
1342 FillSenderStats(info, log_stats); | 1342 FillSenderStats(info, log_stats); |
1343 FillReceiverStats(info, log_stats); | 1343 FillReceiverStats(info, log_stats); |
1344 FillSendAndReceiveCodecStats(info); | 1344 FillSendAndReceiveCodecStats(info); |
1345 // TODO(holmer): We should either have rtt available as a metric on | |
1346 // VideoSend/ReceiveStreams, or we should remove rtt from VideoSenderInfo. | |
1345 webrtc::Call::Stats stats = call_->GetStats(); | 1347 webrtc::Call::Stats stats = call_->GetStats(); |
hbos
2017/05/09 12:48:14
What are these stats, which contain send/recv_band
holmer
2017/05/30 14:44:28
The only reason I have to call Call::GetStats() is
hbos
2017/05/31 14:31:58
Here call_->GetStats() is invoked from WebRtcVideo
stefan-webrtc
2017/05/31 16:48:03
Ideally we should probably call Call::GetStats on
| |
1346 FillBandwidthEstimationStats(stats, info); | |
1347 if (stats.rtt_ms != -1) { | 1348 if (stats.rtt_ms != -1) { |
1348 for (size_t i = 0; i < info->senders.size(); ++i) { | 1349 for (size_t i = 0; i < info->senders.size(); ++i) { |
1349 info->senders[i].rtt_ms = stats.rtt_ms; | 1350 info->senders[i].rtt_ms = stats.rtt_ms; |
1350 } | 1351 } |
1351 } | 1352 } |
1352 | 1353 |
1353 if (log_stats) | 1354 if (log_stats) |
1354 LOG(LS_INFO) << stats.ToString(now_ms); | 1355 LOG(LS_INFO) << stats.ToString(now_ms); |
1355 | 1356 |
1356 return true; | 1357 return true; |
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1371 bool log_stats) { | 1372 bool log_stats) { |
1372 rtc::CritScope stream_lock(&stream_crit_); | 1373 rtc::CritScope stream_lock(&stream_crit_); |
1373 for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = | 1374 for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it = |
1374 receive_streams_.begin(); | 1375 receive_streams_.begin(); |
1375 it != receive_streams_.end(); ++it) { | 1376 it != receive_streams_.end(); ++it) { |
1376 video_media_info->receivers.push_back( | 1377 video_media_info->receivers.push_back( |
1377 it->second->GetVideoReceiverInfo(log_stats)); | 1378 it->second->GetVideoReceiverInfo(log_stats)); |
1378 } | 1379 } |
1379 } | 1380 } |
1380 | 1381 |
1381 void WebRtcVideoChannel2::FillBandwidthEstimationStats( | 1382 void WebRtcVideoChannel2::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) { |
hbos
2017/05/09 12:48:13
nit: Do we still have a need for a bwe pointer or
holmer
2017/05/30 14:44:28
We could, but it'd require that we copy the Bandwi
hbos
2017/05/31 14:31:58
Acknowledged.
| |
1382 const webrtc::Call::Stats& stats, | |
1383 VideoMediaInfo* video_media_info) { | |
1384 BandwidthEstimationInfo bwe_info; | |
1385 bwe_info.available_send_bandwidth = stats.send_bandwidth_bps; | |
1386 bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps; | |
1387 bwe_info.bucket_delay = stats.pacer_delay_ms; | |
1388 | |
1389 // Get send stream bitrate stats. | |
1390 rtc::CritScope stream_lock(&stream_crit_); | 1383 rtc::CritScope stream_lock(&stream_crit_); |
1391 for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream = | 1384 for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream = |
1392 send_streams_.begin(); | 1385 send_streams_.begin(); |
1393 stream != send_streams_.end(); ++stream) { | 1386 stream != send_streams_.end(); ++stream) { |
1394 stream->second->FillBandwidthEstimationInfo(&bwe_info); | 1387 stream->second->FillBitrateInfo(bwe_info); |
1395 } | 1388 } |
1396 video_media_info->bw_estimations.push_back(bwe_info); | |
1397 } | 1389 } |
1398 | 1390 |
1399 void WebRtcVideoChannel2::FillSendAndReceiveCodecStats( | 1391 void WebRtcVideoChannel2::FillSendAndReceiveCodecStats( |
1400 VideoMediaInfo* video_media_info) { | 1392 VideoMediaInfo* video_media_info) { |
1401 for (const VideoCodec& codec : send_params_.codecs) { | 1393 for (const VideoCodec& codec : send_params_.codecs) { |
1402 webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); | 1394 webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters(); |
1403 video_media_info->send_codecs.insert( | 1395 video_media_info->send_codecs.insert( |
1404 std::make_pair(codec_params.payload_type, std::move(codec_params))); | 1396 std::make_pair(codec_params.payload_type, std::move(codec_params))); |
1405 } | 1397 } |
1406 for (const VideoCodec& codec : recv_params_.codecs) { | 1398 for (const VideoCodec& codec : recv_params_.codecs) { |
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2093 webrtc::VideoSendStream::StreamStats first_stream_stats = | 2085 webrtc::VideoSendStream::StreamStats first_stream_stats = |
2094 stats.substreams.begin()->second; | 2086 stats.substreams.begin()->second; |
2095 info.fraction_lost = | 2087 info.fraction_lost = |
2096 static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) / | 2088 static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) / |
2097 (1 << 8); | 2089 (1 << 8); |
2098 } | 2090 } |
2099 | 2091 |
2100 return info; | 2092 return info; |
2101 } | 2093 } |
2102 | 2094 |
2103 void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBandwidthEstimationInfo( | 2095 void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBitrateInfo( |
2104 BandwidthEstimationInfo* bwe_info) { | 2096 BandwidthEstimationInfo* bwe_info) { |
2105 RTC_DCHECK_RUN_ON(&thread_checker_); | 2097 RTC_DCHECK_RUN_ON(&thread_checker_); |
2106 if (stream_ == NULL) { | 2098 if (stream_ == NULL) { |
2107 return; | 2099 return; |
2108 } | 2100 } |
2109 webrtc::VideoSendStream::Stats stats = stream_->GetStats(); | 2101 webrtc::VideoSendStream::Stats stats = stream_->GetStats(); |
2110 for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it = | 2102 for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it = |
2111 stats.substreams.begin(); | 2103 stats.substreams.begin(); |
2112 it != stats.substreams.end(); ++it) { | 2104 it != stats.substreams.end(); ++it) { |
2113 bwe_info->transmit_bitrate += it->second.total_bitrate_bps; | 2105 bwe_info->transmit_bitrate += it->second.total_bitrate_bps; |
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2599 rtx_mapping[video_codecs[i].codec.id] != | 2591 rtx_mapping[video_codecs[i].codec.id] != |
2600 ulpfec_config.red_payload_type) { | 2592 ulpfec_config.red_payload_type) { |
2601 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; | 2593 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; |
2602 } | 2594 } |
2603 } | 2595 } |
2604 | 2596 |
2605 return video_codecs; | 2597 return video_codecs; |
2606 } | 2598 } |
2607 | 2599 |
2608 } // namespace cricket | 2600 } // namespace cricket |
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