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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 |
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583 received_bytes_per_second_counter_.GetStats(); | 583 received_bytes_per_second_counter_.GetStats(); |
584 if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { | 584 if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
585 RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps", | 585 RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps", |
586 recv_bytes_per_sec.average * 8 / 1000); | 586 recv_bytes_per_sec.average * 8 / 1000); |
587 LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, " | 587 LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, " |
588 << recv_bytes_per_sec.ToStringWithMultiplier(8); | 588 << recv_bytes_per_sec.ToStringWithMultiplier(8); |
589 } | 589 } |
590 } | 590 } |
591 | 591 |
592 PacketReceiver* Call::Receiver() { | 592 PacketReceiver* Call::Receiver() { |
593 // TODO(solenberg): Some test cases in EndToEndTest use this from a different | 593 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
594 // thread. Re-enable once that is fixed. | |
595 // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | |
596 return this; | 594 return this; |
597 } | 595 } |
598 | 596 |
599 webrtc::AudioSendStream* Call::CreateAudioSendStream( | 597 webrtc::AudioSendStream* Call::CreateAudioSendStream( |
600 const webrtc::AudioSendStream::Config& config) { | 598 const webrtc::AudioSendStream::Config& config) { |
601 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); | 599 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
602 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 600 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
603 event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config)); | 601 event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config)); |
604 | 602 |
605 rtc::Optional<RtpState> suspended_rtp_state; | 603 rtc::Optional<RtpState> suspended_rtp_state; |
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1365 } | 1363 } |
1366 } | 1364 } |
1367 return DELIVERY_UNKNOWN_SSRC; | 1365 return DELIVERY_UNKNOWN_SSRC; |
1368 } | 1366 } |
1369 | 1367 |
1370 PacketReceiver::DeliveryStatus Call::DeliverPacket( | 1368 PacketReceiver::DeliveryStatus Call::DeliverPacket( |
1371 MediaType media_type, | 1369 MediaType media_type, |
1372 const uint8_t* packet, | 1370 const uint8_t* packet, |
1373 size_t length, | 1371 size_t length, |
1374 const PacketTime& packet_time) { | 1372 const PacketTime& packet_time) { |
1375 // TODO(solenberg): Tests call this function on a network thread, libjingle | 1373 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
1376 // calls on the worker thread. We should move towards always using a network | |
1377 // thread. Then this check can be enabled. | |
1378 // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | |
1379 if (RtpHeaderParser::IsRtcp(packet, length)) | 1374 if (RtpHeaderParser::IsRtcp(packet, length)) |
1380 return DeliverRtcp(media_type, packet, length); | 1375 return DeliverRtcp(media_type, packet, length); |
1381 | 1376 |
1382 return DeliverRtp(media_type, packet, length, packet_time); | 1377 return DeliverRtp(media_type, packet, length, packet_time); |
1383 } | 1378 } |
1384 | 1379 |
1385 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) { | 1380 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) { |
1386 rtc::Optional<RtpPacketReceived> parsed_packet = | 1381 rtc::Optional<RtpPacketReceived> parsed_packet = |
1387 ParseRtpPacket(packet, length, nullptr); | 1382 ParseRtpPacket(packet, length, nullptr); |
1388 if (!parsed_packet) | 1383 if (!parsed_packet) |
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1433 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1428 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1434 receive_side_cc_.OnReceivedPacket( | 1429 receive_side_cc_.OnReceivedPacket( |
1435 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1430 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1436 header); | 1431 header); |
1437 } | 1432 } |
1438 } | 1433 } |
1439 | 1434 |
1440 } // namespace internal | 1435 } // namespace internal |
1441 | 1436 |
1442 } // namespace webrtc | 1437 } // namespace webrtc |
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