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Issue 1658533003: Remove unimplemented VideoChannel code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months 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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1340 capturer->SetApplyRotation(!ContainsHeaderExtension( 1340 capturer->SetApplyRotation(!ContainsHeaderExtension(
1341 send_rtp_extensions_, kRtpVideoRotationHeaderExtension)); 1341 send_rtp_extensions_, kRtpVideoRotationHeaderExtension));
1342 } 1342 }
1343 { 1343 {
1344 rtc::CritScope lock(&capturer_crit_); 1344 rtc::CritScope lock(&capturer_crit_);
1345 capturers_[ssrc] = capturer; 1345 capturers_[ssrc] = capturer;
1346 } 1346 }
1347 return true; 1347 return true;
1348 } 1348 }
1349 1349
1350 bool WebRtcVideoChannel2::SendIntraFrame() {
1351 // TODO(pbos): Implement.
1352 LOG(LS_VERBOSE) << "SendIntraFrame().";
1353 return true;
1354 }
1355
1356 bool WebRtcVideoChannel2::RequestIntraFrame() {
1357 // TODO(pbos): Implement.
1358 LOG(LS_VERBOSE) << "SendIntraFrame().";
1359 return true;
1360 }
1361
1362 void WebRtcVideoChannel2::OnPacketReceived( 1350 void WebRtcVideoChannel2::OnPacketReceived(
1363 rtc::Buffer* packet, 1351 rtc::Buffer* packet,
1364 const rtc::PacketTime& packet_time) { 1352 const rtc::PacketTime& packet_time) {
1365 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, 1353 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
1366 packet_time.not_before); 1354 packet_time.not_before);
1367 const webrtc::PacketReceiver::DeliveryStatus delivery_result = 1355 const webrtc::PacketReceiver::DeliveryStatus delivery_result =
1368 call_->Receiver()->DeliverPacket( 1356 call_->Receiver()->DeliverPacket(
1369 webrtc::MediaType::VIDEO, 1357 webrtc::MediaType::VIDEO,
1370 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(), 1358 reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
1371 webrtc_packet_time); 1359 webrtc_packet_time);
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1467 1455
1468 // Speculative change to increase the outbound socket buffer size. 1456 // Speculative change to increase the outbound socket buffer size.
1469 // In b/15152257, we are seeing a significant number of packets discarded 1457 // In b/15152257, we are seeing a significant number of packets discarded
1470 // due to lack of socket buffer space, although it's not yet clear what the 1458 // due to lack of socket buffer space, although it's not yet clear what the
1471 // ideal value should be. 1459 // ideal value should be.
1472 MediaChannel::SetOption(NetworkInterface::ST_RTP, 1460 MediaChannel::SetOption(NetworkInterface::ST_RTP,
1473 rtc::Socket::OPT_SNDBUF, 1461 rtc::Socket::OPT_SNDBUF,
1474 kVideoRtpBufferSize); 1462 kVideoRtpBufferSize);
1475 } 1463 }
1476 1464
1477 void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) {
1478 // TODO(pbos): Implement.
1479 }
1480
1481 void WebRtcVideoChannel2::OnMessage(rtc::Message* msg) {
1482 // Ignored.
1483 }
1484
1485 void WebRtcVideoChannel2::OnLoadUpdate(Load load) { 1465 void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
1486 // OnLoadUpdate can not take any locks that are held while creating streams 1466 // OnLoadUpdate can not take any locks that are held while creating streams
1487 // etc. Doing so establishes lock-order inversions between the webrtc process 1467 // etc. Doing so establishes lock-order inversions between the webrtc process
1488 // thread on stream creation and locks such as stream_crit_ while calling out. 1468 // thread on stream creation and locks such as stream_crit_ while calling out.
1489 rtc::CritScope stream_lock(&capturer_crit_); 1469 rtc::CritScope stream_lock(&capturer_crit_);
1490 if (!signal_cpu_adaptation_) 1470 if (!signal_cpu_adaptation_)
1491 return; 1471 return;
1492 // Do not adapt resolution for screen content as this will likely result in 1472 // Do not adapt resolution for screen content as this will likely result in
1493 // blurry and unreadable text. 1473 // blurry and unreadable text.
1494 for (auto& kv : capturers_) { 1474 for (auto& kv : capturers_) {
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2605 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2585 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2606 } 2586 }
2607 } 2587 }
2608 2588
2609 return video_codecs; 2589 return video_codecs;
2610 } 2590 }
2611 2591
2612 } // namespace cricket 2592 } // namespace cricket
2613 2593
2614 #endif // HAVE_WEBRTC_VIDEO 2594 #endif // HAVE_WEBRTC_VIDEO
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