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Issue 1458853002: Remove spammy logging of RTCP delivery failures. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 1 month ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2014 Google Inc. 3 * Copyright 2014 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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1471 LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery."; 1471 LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
1472 return; 1472 return;
1473 } 1473 }
1474 } 1474 }
1475 1475
1476 void WebRtcVideoChannel2::OnRtcpReceived( 1476 void WebRtcVideoChannel2::OnRtcpReceived(
1477 rtc::Buffer* packet, 1477 rtc::Buffer* packet,
1478 const rtc::PacketTime& packet_time) { 1478 const rtc::PacketTime& packet_time) {
1479 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, 1479 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
1480 packet_time.not_before); 1480 packet_time.not_before);
1481 if (call_->Receiver()->DeliverPacket( 1481 // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
1482 webrtc::MediaType::VIDEO, 1482 // for both audio and video on the same path. Since BundleFilter doesn't
1483 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(), 1483 // filter RTCP anymore incoming RTCP packets could've been going to audio (so
1484 webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) { 1484 // logging failures spam the log).
1485 LOG(LS_WARNING) << "Failed to deliver RTCP packet."; 1485 call_->Receiver()->DeliverPacket(
1486 } 1486 webrtc::MediaType::VIDEO,
1487 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
1488 webrtc_packet_time);
1487 } 1489 }
1488 1490
1489 void WebRtcVideoChannel2::OnReadyToSend(bool ready) { 1491 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
1490 LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); 1492 LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
1491 call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); 1493 call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
1492 } 1494 }
1493 1495
1494 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) { 1496 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
1495 LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> " 1497 LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
1496 << (mute ? "mute" : "unmute"); 1498 << (mute ? "mute" : "unmute");
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2742 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2744 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2743 } 2745 }
2744 } 2746 }
2745 2747
2746 return video_codecs; 2748 return video_codecs;
2747 } 2749 }
2748 2750
2749 } // namespace cricket 2751 } // namespace cricket
2750 2752
2751 #endif // HAVE_WEBRTC_VIDEO 2753 #endif // HAVE_WEBRTC_VIDEO
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