OLD | NEW |
1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 |
(...skipping 1285 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1296 } | 1296 } |
1297 if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { | 1297 if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { |
1298 ReadLockScoped read_lock(*send_crit_); | 1298 ReadLockScoped read_lock(*send_crit_); |
1299 for (auto& kv : audio_send_ssrcs_) { | 1299 for (auto& kv : audio_send_ssrcs_) { |
1300 if (kv.second->DeliverRtcp(packet, length)) | 1300 if (kv.second->DeliverRtcp(packet, length)) |
1301 rtcp_delivered = true; | 1301 rtcp_delivered = true; |
1302 } | 1302 } |
1303 } | 1303 } |
1304 | 1304 |
1305 if (rtcp_delivered) | 1305 if (rtcp_delivered) |
1306 event_log_->LogRtcpPacket(kIncomingPacket, packet, length); | 1306 event_log_->LogIncomingRtcpPacket(rtc::MakeArrayView(packet, length)); |
1307 | 1307 |
1308 return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; | 1308 return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
1309 } | 1309 } |
1310 | 1310 |
1311 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, | 1311 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
1312 const uint8_t* packet, | 1312 const uint8_t* packet, |
1313 size_t length, | 1313 size_t length, |
1314 const PacketTime& packet_time) { | 1314 const PacketTime& packet_time) { |
1315 TRACE_EVENT0("webrtc", "Call::DeliverRtp"); | 1315 TRACE_EVENT0("webrtc", "Call::DeliverRtp"); |
1316 | 1316 |
(...skipping 28 matching lines...) Expand all Loading... |
1345 return DELIVERY_UNKNOWN_SSRC; | 1345 return DELIVERY_UNKNOWN_SSRC; |
1346 } | 1346 } |
1347 parsed_packet->IdentifyExtensions(it->second.extensions); | 1347 parsed_packet->IdentifyExtensions(it->second.extensions); |
1348 | 1348 |
1349 NotifyBweOfReceivedPacket(*parsed_packet, media_type); | 1349 NotifyBweOfReceivedPacket(*parsed_packet, media_type); |
1350 | 1350 |
1351 if (media_type == MediaType::AUDIO) { | 1351 if (media_type == MediaType::AUDIO) { |
1352 if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { | 1352 if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
1353 received_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1353 received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1354 received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1354 received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1355 event_log_->LogRtpHeader(kIncomingPacket, packet, length); | 1355 event_log_->LogIncomingRtpHeader(*parsed_packet); |
1356 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); | 1356 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
1357 if (!first_received_rtp_audio_ms_) { | 1357 if (!first_received_rtp_audio_ms_) { |
1358 first_received_rtp_audio_ms_.emplace(arrival_time_ms); | 1358 first_received_rtp_audio_ms_.emplace(arrival_time_ms); |
1359 } | 1359 } |
1360 last_received_rtp_audio_ms_.emplace(arrival_time_ms); | 1360 last_received_rtp_audio_ms_.emplace(arrival_time_ms); |
1361 return DELIVERY_OK; | 1361 return DELIVERY_OK; |
1362 } | 1362 } |
1363 } else if (media_type == MediaType::VIDEO) { | 1363 } else if (media_type == MediaType::VIDEO) { |
1364 if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { | 1364 if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
1365 received_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1365 received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1366 received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1366 received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1367 event_log_->LogRtpHeader(kIncomingPacket, packet, length); | 1367 event_log_->LogIncomingRtpHeader(*parsed_packet); |
1368 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); | 1368 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
1369 if (!first_received_rtp_video_ms_) { | 1369 if (!first_received_rtp_video_ms_) { |
1370 first_received_rtp_video_ms_.emplace(arrival_time_ms); | 1370 first_received_rtp_video_ms_.emplace(arrival_time_ms); |
1371 } | 1371 } |
1372 last_received_rtp_video_ms_.emplace(arrival_time_ms); | 1372 last_received_rtp_video_ms_.emplace(arrival_time_ms); |
1373 return DELIVERY_OK; | 1373 return DELIVERY_OK; |
1374 } | 1374 } |
1375 } | 1375 } |
1376 return DELIVERY_UNKNOWN_SSRC; | 1376 return DELIVERY_UNKNOWN_SSRC; |
1377 } | 1377 } |
(...skipping 61 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
1439 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1439 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1440 receive_side_cc_.OnReceivedPacket( | 1440 receive_side_cc_.OnReceivedPacket( |
1441 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1441 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1442 header); | 1442 header); |
1443 } | 1443 } |
1444 } | 1444 } |
1445 | 1445 |
1446 } // namespace internal | 1446 } // namespace internal |
1447 | 1447 |
1448 } // namespace webrtc | 1448 } // namespace webrtc |
OLD | NEW |