Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(107)

Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.cc

Issue 1695263002: Move direct use of VideoCapturer::VideoAdapter to VideoSinkWants. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Tried to address pthatchers comments. Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
11 #include "webrtc/media/engine/webrtcvideoengine2.h" 11 #include "webrtc/media/engine/webrtcvideoengine2.h"
12 12
13 #include <algorithm> 13 #include <algorithm>
14 #include <set> 14 #include <set>
15 #include <string> 15 #include <string>
16 16
17 #include "webrtc/base/asyncinvoker.h"
17 #include "webrtc/base/buffer.h" 18 #include "webrtc/base/buffer.h"
18 #include "webrtc/base/logging.h" 19 #include "webrtc/base/logging.h"
19 #include "webrtc/base/stringutils.h" 20 #include "webrtc/base/stringutils.h"
20 #include "webrtc/base/timeutils.h" 21 #include "webrtc/base/timeutils.h"
21 #include "webrtc/base/trace_event.h" 22 #include "webrtc/base/trace_event.h"
22 #include "webrtc/call.h" 23 #include "webrtc/call.h"
23 #include "webrtc/media/base/videocapturer.h" 24 #include "webrtc/media/base/videocapturer.h"
24 #include "webrtc/media/base/videorenderer.h" 25 #include "webrtc/media/base/videorenderer.h"
25 #include "webrtc/media/engine/constants.h" 26 #include "webrtc/media/engine/constants.h"
26 #include "webrtc/media/engine/simulcast.h" 27 #include "webrtc/media/engine/simulcast.h"
(...skipping 280 matching lines...) Expand 10 before | Expand all | Expand 10 after
307 if (width * height <= 320 * 240) { 308 if (width * height <= 320 * 240) {
308 return 600; 309 return 600;
309 } else if (width * height <= 640 * 480) { 310 } else if (width * height <= 640 * 480) {
310 return 1700; 311 return 1700;
311 } else if (width * height <= 960 * 540) { 312 } else if (width * height <= 960 * 540) {
312 return 2000; 313 return 2000;
313 } else { 314 } else {
314 return 2500; 315 return 2500;
315 } 316 }
316 } 317 }
318
317 } // namespace 319 } // namespace
318 320
319 // Constants defined in webrtc/media/engine/constants.h 321 // Constants defined in webrtc/media/engine/constants.h
320 // TODO(pbos): Move these to a separate constants.cc file. 322 // TODO(pbos): Move these to a separate constants.cc file.
321 const int kMinVideoBitrate = 30; 323 const int kMinVideoBitrate = 30;
322 const int kStartVideoBitrate = 300; 324 const int kStartVideoBitrate = 300;
323 325
324 const int kVideoMtu = 1200; 326 const int kVideoMtu = 1200;
325 const int kVideoRtpBufferSize = 65536; 327 const int kVideoRtpBufferSize = 65536;
326 328
327 // This constant is really an on/off, lower-level configurable NACK history 329 // This constant is really an on/off, lower-level configurable NACK history
328 // duration hasn't been implemented. 330 // duration hasn't been implemented.
329 static const int kNackHistoryMs = 1000; 331 static const int kNackHistoryMs = 1000;
330 332
331 static const int kDefaultQpMax = 56; 333 static const int kDefaultQpMax = 56;
332 334
333 static const int kDefaultRtcpReceiverReportSsrc = 1; 335 static const int kDefaultRtcpReceiverReportSsrc = 1;
334 336
337 // Reduce wanted maximum number of pixels by at most 4.
338 static const int kMaxCpuDownScale = 4;
339
335 std::vector<VideoCodec> DefaultVideoCodecList() { 340 std::vector<VideoCodec> DefaultVideoCodecList() {
336 std::vector<VideoCodec> codecs; 341 std::vector<VideoCodec> codecs;
337 codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType, 342 codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
338 kVp8CodecName)); 343 kVp8CodecName));
339 codecs.push_back( 344 codecs.push_back(
340 VideoCodec::CreateRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType)); 345 VideoCodec::CreateRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType));
341 if (CodecIsInternallySupported(kVp9CodecName)) { 346 if (CodecIsInternallySupported(kVp9CodecName)) {
342 codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType, 347 codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType,
343 kVp9CodecName)); 348 kVp9CodecName));
344 codecs.push_back( 349 codecs.push_back(
(...skipping 642 matching lines...) Expand 10 before | Expand all | Expand 10 after
987 992
988 rtc::CritScope stream_lock(&stream_crit_); 993 rtc::CritScope stream_lock(&stream_crit_);
989 994
990 if (!ValidateSendSsrcAvailability(sp)) 995 if (!ValidateSendSsrcAvailability(sp))
991 return false; 996 return false;
992 997
993 for (uint32_t used_ssrc : sp.ssrcs) 998 for (uint32_t used_ssrc : sp.ssrcs)
994 send_ssrcs_.insert(used_ssrc); 999 send_ssrcs_.insert(used_ssrc);
995 1000
996 webrtc::VideoSendStream::Config config(this); 1001 webrtc::VideoSendStream::Config config(this);
997 config.overuse_callback = this;
998
999 WebRtcVideoSendStream* stream = new WebRtcVideoSendStream( 1002 WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
1000 call_, sp, config, external_encoder_factory_, options_, 1003 call_, sp, config, external_encoder_factory_, options_,
1001 bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_, 1004 signal_cpu_adaptation_, bitrate_config_.max_bitrate_bps, send_codec_,
1002 send_params_); 1005 send_rtp_extensions_, send_params_);
1003
1004 uint32_t ssrc = sp.first_ssrc(); 1006 uint32_t ssrc = sp.first_ssrc();
1005 RTC_DCHECK(ssrc != 0); 1007 RTC_DCHECK(ssrc != 0);
1006 send_streams_[ssrc] = stream; 1008 send_streams_[ssrc] = stream;
1007 1009
1008 if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) { 1010 if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
1009 rtcp_receiver_report_ssrc_ = ssrc; 1011 rtcp_receiver_report_ssrc_ = ssrc;
1010 LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added " 1012 LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
1011 "a send stream."; 1013 "a send stream.";
1012 for (auto& kv : receive_streams_) 1014 for (auto& kv : receive_streams_)
1013 kv.second->SetLocalSsrc(ssrc); 1015 kv.second->SetLocalSsrc(ssrc);
(...skipping 377 matching lines...) Expand 10 before | Expand all | Expand 10 after
1391 1393
1392 // Speculative change to increase the outbound socket buffer size. 1394 // Speculative change to increase the outbound socket buffer size.
1393 // In b/15152257, we are seeing a significant number of packets discarded 1395 // In b/15152257, we are seeing a significant number of packets discarded
1394 // due to lack of socket buffer space, although it's not yet clear what the 1396 // due to lack of socket buffer space, although it's not yet clear what the
1395 // ideal value should be. 1397 // ideal value should be.
1396 MediaChannel::SetOption(NetworkInterface::ST_RTP, 1398 MediaChannel::SetOption(NetworkInterface::ST_RTP,
1397 rtc::Socket::OPT_SNDBUF, 1399 rtc::Socket::OPT_SNDBUF,
1398 kVideoRtpBufferSize); 1400 kVideoRtpBufferSize);
1399 } 1401 }
1400 1402
1401 void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
1402 // OnLoadUpdate can not take any locks that are held while creating streams
1403 // etc. Doing so establishes lock-order inversions between the webrtc process
1404 // thread on stream creation and locks such as stream_crit_ while calling out.
1405 rtc::CritScope stream_lock(&capturer_crit_);
1406 if (!signal_cpu_adaptation_)
1407 return;
1408 // Do not adapt resolution for screen content as this will likely result in
1409 // blurry and unreadable text.
1410 for (auto& kv : capturers_) {
1411 if (kv.second != nullptr
1412 && !kv.second->IsScreencast()
1413 && kv.second->video_adapter() != nullptr) {
1414 kv.second->video_adapter()->OnCpuResolutionRequest(
1415 load == kOveruse ? CoordinatedVideoAdapter::DOWNGRADE
1416 : CoordinatedVideoAdapter::UPGRADE);
1417 }
1418 }
1419 }
1420
1421 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, 1403 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
1422 size_t len, 1404 size_t len,
1423 const webrtc::PacketOptions& options) { 1405 const webrtc::PacketOptions& options) {
1424 rtc::Buffer packet(data, len, kMaxRtpPacketLen); 1406 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
1425 rtc::PacketOptions rtc_options; 1407 rtc::PacketOptions rtc_options;
1426 rtc_options.packet_id = options.packet_id; 1408 rtc_options.packet_id = options.packet_id;
1427 return MediaChannel::SendPacket(&packet, rtc_options); 1409 return MediaChannel::SendPacket(&packet, rtc_options);
1428 } 1410 }
1429 1411
1430 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) { 1412 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
1469 external_encoder(nullptr), 1451 external_encoder(nullptr),
1470 type(type), 1452 type(type),
1471 external(external) { 1453 external(external) {
1472 if (external) { 1454 if (external) {
1473 external_encoder = encoder; 1455 external_encoder = encoder;
1474 this->encoder = 1456 this->encoder =
1475 new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder); 1457 new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
1476 } 1458 }
1477 } 1459 }
1478 1460
1461 // Proxy class used for marshalling calls to webrtc::LoadObserver::OnLoadUpdate
1462 // from a media engine thread to the worker thread.
1463 class WebRtcVideoChannel2::WebRtcVideoSendStream::LoadObserverProxy {
1464 public:
1465 explicit LoadObserverProxy(webrtc::LoadObserver* observer) {
1466 helper_ = new rtc::RefCountedObject<Helper>(observer);
1467 }
1468 ~LoadObserverProxy() { helper_->Detach(); }
1469
1470 webrtc::LoadObserver* proxy() { return helper_; }
1471
1472 private:
1473 class Helper : public webrtc::LoadObserver, public rtc::RefCountInterface {
1474 public:
1475 explicit Helper(webrtc::LoadObserver* observer)
1476 : thread_(rtc::Thread::Current()), observer_(observer) {}
1477 void Detach() {
1478 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1479 observer_ = nullptr;
1480 }
1481 void OnLoadUpdate(webrtc::LoadObserver::Load load) override {
1482 if (rtc::Thread::Current() == thread_) {
1483 observer_->OnLoadUpdate(load);
1484 return;
1485 }
1486 invoker_.AsyncInvoke<void>(
1487 thread_, rtc::Bind(&Helper::OnLoadUpdateOnCorrectThread, this, load));
1488 }
1489 void OnLoadUpdateOnCorrectThread(webrtc::LoadObserver::Load load) {
1490 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1491 if (observer_) {
1492 observer_->OnLoadUpdate(load);
1493 }
1494 }
1495
1496 private:
1497 rtc::ThreadChecker thread_checker_;
1498 rtc::AsyncInvoker invoker_;
1499 rtc::Thread* thread_;
1500 webrtc::LoadObserver* observer_;
1501 };
1502
1503 rtc::scoped_refptr<Helper> helper_;
1504 };
1505
1479 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( 1506 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
1480 webrtc::Call* call, 1507 webrtc::Call* call,
1481 const StreamParams& sp, 1508 const StreamParams& sp,
1482 const webrtc::VideoSendStream::Config& config, 1509 const webrtc::VideoSendStream::Config& config,
1483 WebRtcVideoEncoderFactory* external_encoder_factory, 1510 WebRtcVideoEncoderFactory* external_encoder_factory,
1484 const VideoOptions& options, 1511 const VideoOptions& options,
1512 bool enable_cpu_overuse_detection,
1485 int max_bitrate_bps, 1513 int max_bitrate_bps,
1486 const rtc::Optional<VideoCodecSettings>& codec_settings, 1514 const rtc::Optional<VideoCodecSettings>& codec_settings,
1487 const std::vector<webrtc::RtpExtension>& rtp_extensions, 1515 const std::vector<webrtc::RtpExtension>& rtp_extensions,
1488 // TODO(deadbeef): Don't duplicate information between send_params, 1516 // TODO(deadbeef): Don't duplicate information between send_params,
1489 // rtp_extensions, options, etc. 1517 // rtp_extensions, options, etc.
1490 const VideoSendParameters& send_params) 1518 const VideoSendParameters& send_params)
1491 : ssrcs_(sp.ssrcs), 1519 : ssrcs_(sp.ssrcs),
1492 ssrc_groups_(sp.ssrc_groups), 1520 ssrc_groups_(sp.ssrc_groups),
1493 call_(call), 1521 call_(call),
1522 load_proxy_(new LoadObserverProxy(this)),
1494 external_encoder_factory_(external_encoder_factory), 1523 external_encoder_factory_(external_encoder_factory),
1495 stream_(NULL), 1524 stream_(NULL),
1496 parameters_(config, options, max_bitrate_bps, codec_settings), 1525 parameters_(config, options, max_bitrate_bps, codec_settings),
1497 pending_encoder_reconfiguration_(false), 1526 pending_encoder_reconfiguration_(false),
1498 allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false), 1527 allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
1499 capturer_(NULL), 1528 capturer_(NULL),
1500 sending_(false), 1529 sending_(false),
1501 muted_(false), 1530 muted_(false),
1502 old_adapt_changes_(0), 1531 old_adapt_changes_(0),
1503 first_frame_timestamp_ms_(0), 1532 first_frame_timestamp_ms_(0),
1504 last_frame_timestamp_ms_(0) { 1533 last_frame_timestamp_ms_(0) {
1505 parameters_.config.rtp.max_packet_size = kVideoMtu; 1534 parameters_.config.rtp.max_packet_size = kVideoMtu;
1506 1535
1507 sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs); 1536 sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
1508 sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, 1537 sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
1509 &parameters_.config.rtp.rtx.ssrcs); 1538 &parameters_.config.rtp.rtx.ssrcs);
1510 parameters_.config.rtp.c_name = sp.cname; 1539 parameters_.config.rtp.c_name = sp.cname;
1511 parameters_.config.rtp.extensions = rtp_extensions; 1540 parameters_.config.rtp.extensions = rtp_extensions;
1512 parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size 1541 parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
1513 ? webrtc::RtcpMode::kReducedSize 1542 ? webrtc::RtcpMode::kReducedSize
1514 : webrtc::RtcpMode::kCompound; 1543 : webrtc::RtcpMode::kCompound;
1544 parameters_.config.overuse_callback =
1545 enable_cpu_overuse_detection ? load_proxy_->proxy() : nullptr;
1515 1546
1516 if (codec_settings) { 1547 if (codec_settings) {
1517 SetCodecAndOptions(*codec_settings, parameters_.options); 1548 SetCodecAndOptions(*codec_settings, parameters_.options);
1518 } 1549 }
1519 } 1550 }
1520 1551
1521 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() { 1552 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
1522 DisconnectCapturer(); 1553 DisconnectCapturer();
1523 if (stream_ != NULL) { 1554 if (stream_ != NULL) {
1524 call_->DestroyVideoSendStream(stream_); 1555 call_->DestroyVideoSendStream(stream_);
(...skipping 103 matching lines...) Expand 10 before | Expand all | Expand 10 after
1628 muted_ = mute; 1659 muted_ = mute;
1629 } 1660 }
1630 1661
1631 bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() { 1662 bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
1632 cricket::VideoCapturer* capturer; 1663 cricket::VideoCapturer* capturer;
1633 { 1664 {
1634 rtc::CritScope cs(&lock_); 1665 rtc::CritScope cs(&lock_);
1635 if (capturer_ == NULL) 1666 if (capturer_ == NULL)
1636 return false; 1667 return false;
1637 1668
1638 if (capturer_->video_adapter() != nullptr)
1639 old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes();
1640
1641 capturer = capturer_; 1669 capturer = capturer_;
1642 capturer_ = NULL; 1670 capturer_ = NULL;
1643 } 1671 }
1644 capturer->RemoveSink(this); 1672 capturer->RemoveSink(this);
1645 1673
1646 return true; 1674 return true;
1647 } 1675 }
1648 1676
1649 const std::vector<uint32_t>& 1677 const std::vector<uint32_t>&
1650 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const { 1678 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
(...skipping 271 matching lines...) Expand 10 before | Expand all | Expand 10 after
1922 } 1950 }
1923 1951
1924 void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() { 1952 void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
1925 rtc::CritScope cs(&lock_); 1953 rtc::CritScope cs(&lock_);
1926 if (stream_ != NULL) { 1954 if (stream_ != NULL) {
1927 stream_->Stop(); 1955 stream_->Stop();
1928 } 1956 }
1929 sending_ = false; 1957 sending_ = false;
1930 } 1958 }
1931 1959
1960 void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
1961 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1962 LOG(LS_INFO) << "OnLoadUpdate " << load;
1963 rtc::CritScope cs(&lock_);
1964 if (!capturer_) {
1965 return;
1966 }
1967
1968 if (load == kOveruse) {
1969 if (cpu_downscale_ < kMaxCpuDownScale) {
1970 cpu_downscale_ *= 2;
1971 sink_wants_.max_number_of_pixels = rtc::Optional<int>(
1972 (last_dimensions_.height * last_dimensions_.width) / 2);
1973 }
1974 } else {
1975 if (cpu_downscale_ > 1) {
1976 cpu_downscale_ /= 2;
1977 sink_wants_.max_number_of_pixels = rtc::Optional<int>(
1978 (last_dimensions_.height * last_dimensions_.width) * 2);
1979 }
1980 }
1981 capturer_->AddOrUpdateSink(this, sink_wants_);
1982 }
1983
1932 VideoSenderInfo 1984 VideoSenderInfo
1933 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() { 1985 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
1934 VideoSenderInfo info; 1986 VideoSenderInfo info;
1935 webrtc::VideoSendStream::Stats stats; 1987 webrtc::VideoSendStream::Stats stats;
1936 { 1988 {
1937 rtc::CritScope cs(&lock_); 1989 rtc::CritScope cs(&lock_);
1938 for (uint32_t ssrc : parameters_.config.rtp.ssrcs) 1990 for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
1939 info.add_ssrc(ssrc); 1991 info.add_ssrc(ssrc);
1940 1992
1941 if (parameters_.codec_settings) 1993 if (parameters_.codec_settings)
(...skipping 18 matching lines...) Expand all
1960 2012
1961 if (capturer_ != NULL) { 2013 if (capturer_ != NULL) {
1962 if (!capturer_->IsMuted()) { 2014 if (!capturer_->IsMuted()) {
1963 VideoFormat last_captured_frame_format; 2015 VideoFormat last_captured_frame_format;
1964 capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops, 2016 capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops,
1965 &info.capturer_frame_time, 2017 &info.capturer_frame_time,
1966 &last_captured_frame_format); 2018 &last_captured_frame_format);
1967 info.input_frame_width = last_captured_frame_format.width; 2019 info.input_frame_width = last_captured_frame_format.width;
1968 info.input_frame_height = last_captured_frame_format.height; 2020 info.input_frame_height = last_captured_frame_format.height;
1969 } 2021 }
1970 if (capturer_->video_adapter() != nullptr) {
1971 info.adapt_changes += capturer_->video_adapter()->adaptation_changes();
1972 info.adapt_reason = capturer_->video_adapter()->adapt_reason();
1973 }
1974 } 2022 }
1975 } 2023 }
1976 2024
1977 // Get bandwidth limitation info from stream_->GetStats(). 2025 // Get bandwidth limitation info from stream_->GetStats().
1978 // Input resolution (output from video_adapter) can be further scaled down or 2026 // Input resolution (output from video_adapter) can be further scaled down or
1979 // higher video layer(s) can be dropped due to bitrate constraints. 2027 // higher video layer(s) can be dropped due to bitrate constraints.
1980 // Note, adapt_changes only include changes from the video_adapter. 2028 // Note, adapt_changes only include changes from the video_adapter.
1981 if (stats.bw_limited_resolution) 2029 if (stats.bw_limited_resolution)
1982 info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH; 2030 info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH;
1983 2031
(...skipping 499 matching lines...) Expand 10 before | Expand all | Expand 10 after
2483 rtx_mapping[video_codecs[i].codec.id] != 2531 rtx_mapping[video_codecs[i].codec.id] !=
2484 fec_settings.red_payload_type) { 2532 fec_settings.red_payload_type) {
2485 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2533 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2486 } 2534 }
2487 } 2535 }
2488 2536
2489 return video_codecs; 2537 return video_codecs;
2490 } 2538 }
2491 2539
2492 } // namespace cricket 2540 } // namespace cricket
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698