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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: Rebased Created 4 years, 9 months ago
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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
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307 if (width * height <= 320 * 240) { 307 if (width * height <= 320 * 240) {
308 return 600; 308 return 600;
309 } else if (width * height <= 640 * 480) { 309 } else if (width * height <= 640 * 480) {
310 return 1700; 310 return 1700;
311 } else if (width * height <= 960 * 540) { 311 } else if (width * height <= 960 * 540) {
312 return 2000; 312 return 2000;
313 } else { 313 } else {
314 return 2500; 314 return 2500;
315 } 315 }
316 } 316 }
317
317 } // namespace 318 } // namespace
318 319
319 // Constants defined in webrtc/media/engine/constants.h 320 // Constants defined in webrtc/media/engine/constants.h
320 // TODO(pbos): Move these to a separate constants.cc file. 321 // TODO(pbos): Move these to a separate constants.cc file.
321 const int kMinVideoBitrate = 30; 322 const int kMinVideoBitrate = 30;
322 const int kStartVideoBitrate = 300; 323 const int kStartVideoBitrate = 300;
323 324
324 const int kVideoMtu = 1200; 325 const int kVideoMtu = 1200;
325 const int kVideoRtpBufferSize = 65536; 326 const int kVideoRtpBufferSize = 65536;
326 327
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993 994
994 rtc::CritScope stream_lock(&stream_crit_); 995 rtc::CritScope stream_lock(&stream_crit_);
995 996
996 if (!ValidateSendSsrcAvailability(sp)) 997 if (!ValidateSendSsrcAvailability(sp))
997 return false; 998 return false;
998 999
999 for (uint32_t used_ssrc : sp.ssrcs) 1000 for (uint32_t used_ssrc : sp.ssrcs)
1000 send_ssrcs_.insert(used_ssrc); 1001 send_ssrcs_.insert(used_ssrc);
1001 1002
1002 webrtc::VideoSendStream::Config config(this); 1003 webrtc::VideoSendStream::Config config(this);
1003 config.overuse_callback = this; 1004 WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
1004 1005 call_, sp, config, external_encoder_factory_, signal_cpu_adaptation_,
1005 WebRtcVideoSendStream* stream = 1006 bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
1006 new WebRtcVideoSendStream(call_, sp, config, external_encoder_factory_, 1007 send_params_);
1007 bitrate_config_.max_bitrate_bps, send_codec_,
1008 send_rtp_extensions_, send_params_);
1009 1008
1010 uint32_t ssrc = sp.first_ssrc(); 1009 uint32_t ssrc = sp.first_ssrc();
1011 RTC_DCHECK(ssrc != 0); 1010 RTC_DCHECK(ssrc != 0);
1012 send_streams_[ssrc] = stream; 1011 send_streams_[ssrc] = stream;
1013 1012
1014 if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) { 1013 if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
1015 rtcp_receiver_report_ssrc_ = ssrc; 1014 rtcp_receiver_report_ssrc_ = ssrc;
1016 LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added " 1015 LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
1017 "a send stream."; 1016 "a send stream.";
1018 for (auto& kv : receive_streams_) 1017 for (auto& kv : receive_streams_)
(...skipping 257 matching lines...) Expand 10 before | Expand all | Expand 10 after
1276 rtc::CritScope stream_lock(&stream_crit_); 1275 rtc::CritScope stream_lock(&stream_crit_);
1277 const auto& kv = send_streams_.find(ssrc); 1276 const auto& kv = send_streams_.find(ssrc);
1278 if (kv == send_streams_.end()) { 1277 if (kv == send_streams_.end()) {
1279 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; 1278 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
1280 return false; 1279 return false;
1281 } 1280 }
1282 if (!kv->second->SetCapturer(capturer)) { 1281 if (!kv->second->SetCapturer(capturer)) {
1283 return false; 1282 return false;
1284 } 1283 }
1285 } 1284 }
1286 {
1287 rtc::CritScope lock(&capturer_crit_);
1288 capturers_[ssrc] = capturer;
1289 }
1290 return true; 1285 return true;
1291 } 1286 }
1292 1287
1293 void WebRtcVideoChannel2::OnPacketReceived( 1288 void WebRtcVideoChannel2::OnPacketReceived(
1294 rtc::Buffer* packet, 1289 rtc::Buffer* packet,
1295 const rtc::PacketTime& packet_time) { 1290 const rtc::PacketTime& packet_time) {
1296 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp, 1291 const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
1297 packet_time.not_before); 1292 packet_time.not_before);
1298 const webrtc::PacketReceiver::DeliveryStatus delivery_result = 1293 const webrtc::PacketReceiver::DeliveryStatus delivery_result =
1299 call_->Receiver()->DeliverPacket( 1294 call_->Receiver()->DeliverPacket(
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1405 1400
1406 // Speculative change to increase the outbound socket buffer size. 1401 // Speculative change to increase the outbound socket buffer size.
1407 // In b/15152257, we are seeing a significant number of packets discarded 1402 // In b/15152257, we are seeing a significant number of packets discarded
1408 // due to lack of socket buffer space, although it's not yet clear what the 1403 // due to lack of socket buffer space, although it's not yet clear what the
1409 // ideal value should be. 1404 // ideal value should be.
1410 MediaChannel::SetOption(NetworkInterface::ST_RTP, 1405 MediaChannel::SetOption(NetworkInterface::ST_RTP,
1411 rtc::Socket::OPT_SNDBUF, 1406 rtc::Socket::OPT_SNDBUF,
1412 kVideoRtpBufferSize); 1407 kVideoRtpBufferSize);
1413 } 1408 }
1414 1409
1415 void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
1416 // OnLoadUpdate can not take any locks that are held while creating streams
1417 // etc. Doing so establishes lock-order inversions between the webrtc process
1418 // thread on stream creation and locks such as stream_crit_ while calling out.
1419 rtc::CritScope stream_lock(&capturer_crit_);
1420 if (!signal_cpu_adaptation_)
1421 return;
1422 // Do not adapt resolution for screen content as this will likely result in
1423 // blurry and unreadable text.
1424 for (auto& kv : capturers_) {
1425 if (kv.second != nullptr
1426 && !kv.second->IsScreencast()
1427 && kv.second->video_adapter() != nullptr) {
1428 kv.second->video_adapter()->OnCpuResolutionRequest(
1429 load == kOveruse ? CoordinatedVideoAdapter::DOWNGRADE
1430 : CoordinatedVideoAdapter::UPGRADE);
1431 }
1432 }
1433 }
1434
1435 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data, 1410 bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
1436 size_t len, 1411 size_t len,
1437 const webrtc::PacketOptions& options) { 1412 const webrtc::PacketOptions& options) {
1438 rtc::Buffer packet(data, len, kMaxRtpPacketLen); 1413 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
1439 rtc::PacketOptions rtc_options; 1414 rtc::PacketOptions rtc_options;
1440 rtc_options.packet_id = options.packet_id; 1415 rtc_options.packet_id = options.packet_id;
1441 return MediaChannel::SendPacket(&packet, rtc_options); 1416 return MediaChannel::SendPacket(&packet, rtc_options);
1442 } 1417 }
1443 1418
1444 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) { 1419 bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
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1488 this->encoder = 1463 this->encoder =
1489 new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder); 1464 new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
1490 } 1465 }
1491 } 1466 }
1492 1467
1493 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream( 1468 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
1494 webrtc::Call* call, 1469 webrtc::Call* call,
1495 const StreamParams& sp, 1470 const StreamParams& sp,
1496 const webrtc::VideoSendStream::Config& config, 1471 const webrtc::VideoSendStream::Config& config,
1497 WebRtcVideoEncoderFactory* external_encoder_factory, 1472 WebRtcVideoEncoderFactory* external_encoder_factory,
1473 bool enable_cpu_overuse_detection,
1498 int max_bitrate_bps, 1474 int max_bitrate_bps,
1499 const rtc::Optional<VideoCodecSettings>& codec_settings, 1475 const rtc::Optional<VideoCodecSettings>& codec_settings,
1500 const std::vector<webrtc::RtpExtension>& rtp_extensions, 1476 const std::vector<webrtc::RtpExtension>& rtp_extensions,
1501 // TODO(deadbeef): Don't duplicate information between send_params, 1477 // TODO(deadbeef): Don't duplicate information between send_params,
1502 // rtp_extensions, options, etc. 1478 // rtp_extensions, options, etc.
1503 const VideoSendParameters& send_params) 1479 const VideoSendParameters& send_params)
1504 : ssrcs_(sp.ssrcs), 1480 : worker_thread_(rtc::Thread::Current()),
1481 ssrcs_(sp.ssrcs),
1505 ssrc_groups_(sp.ssrc_groups), 1482 ssrc_groups_(sp.ssrc_groups),
1506 call_(call), 1483 call_(call),
1484 cpu_restricted_counter_(0),
1485 number_of_cpu_adapt_changes_(0),
1486 capturer_(nullptr),
1507 external_encoder_factory_(external_encoder_factory), 1487 external_encoder_factory_(external_encoder_factory),
1508 stream_(NULL), 1488 stream_(nullptr),
1509 parameters_(config, send_params.options, max_bitrate_bps, codec_settings), 1489 parameters_(config, send_params.options, max_bitrate_bps, codec_settings),
1510 pending_encoder_reconfiguration_(false), 1490 pending_encoder_reconfiguration_(false),
1511 allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false), 1491 allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
1512 capturer_(NULL), 1492 capturer_is_screencast_(false),
1513 sending_(false), 1493 sending_(false),
1514 muted_(false), 1494 muted_(false),
1515 old_adapt_changes_(0),
1516 first_frame_timestamp_ms_(0), 1495 first_frame_timestamp_ms_(0),
1517 last_frame_timestamp_ms_(0) { 1496 last_frame_timestamp_ms_(0) {
1518 parameters_.config.rtp.max_packet_size = kVideoMtu; 1497 parameters_.config.rtp.max_packet_size = kVideoMtu;
1519 parameters_.conference_mode = send_params.conference_mode; 1498 parameters_.conference_mode = send_params.conference_mode;
1520 1499
1521 sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs); 1500 sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
1522 sp.GetFidSsrcs(parameters_.config.rtp.ssrcs, 1501 sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
1523 &parameters_.config.rtp.rtx.ssrcs); 1502 &parameters_.config.rtp.rtx.ssrcs);
1524 parameters_.config.rtp.c_name = sp.cname; 1503 parameters_.config.rtp.c_name = sp.cname;
1525 parameters_.config.rtp.extensions = rtp_extensions; 1504 parameters_.config.rtp.extensions = rtp_extensions;
1526 parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size 1505 parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
1527 ? webrtc::RtcpMode::kReducedSize 1506 ? webrtc::RtcpMode::kReducedSize
1528 : webrtc::RtcpMode::kCompound; 1507 : webrtc::RtcpMode::kCompound;
1508 parameters_.config.overuse_callback =
1509 enable_cpu_overuse_detection ? this : nullptr;
1529 1510
1530 if (codec_settings) { 1511 if (codec_settings) {
1531 SetCodecAndOptions(*codec_settings, parameters_.options); 1512 SetCodecAndOptions(*codec_settings, parameters_.options);
1532 } 1513 }
1533 } 1514 }
1534 1515
1535 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() { 1516 WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
1536 DisconnectCapturer(); 1517 DisconnectCapturer();
1537 if (stream_ != NULL) { 1518 if (stream_ != NULL) {
1538 call_->DestroyVideoSendStream(stream_); 1519 call_->DestroyVideoSendStream(stream_);
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1582 int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec; 1563 int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
1583 // frame->GetTimeStamp() is essentially a delta, align to webrtc time 1564 // frame->GetTimeStamp() is essentially a delta, align to webrtc time
1584 if (first_frame_timestamp_ms_ == 0) { 1565 if (first_frame_timestamp_ms_ == 0) {
1585 first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms; 1566 first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
1586 } 1567 }
1587 1568
1588 last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms; 1569 last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
1589 video_frame.set_render_time_ms(last_frame_timestamp_ms_); 1570 video_frame.set_render_time_ms(last_frame_timestamp_ms_);
1590 // Reconfigure codec if necessary. 1571 // Reconfigure codec if necessary.
1591 SetDimensions(video_frame.width(), video_frame.height(), 1572 SetDimensions(video_frame.width(), video_frame.height(),
1592 capturer_->IsScreencast()); 1573 capturer_is_screencast_);
1593 last_rotation_ = video_frame.rotation(); 1574 last_rotation_ = video_frame.rotation();
1594 1575
1595 stream_->Input()->IncomingCapturedFrame(video_frame); 1576 stream_->Input()->IncomingCapturedFrame(video_frame);
1596 } 1577 }
1597 1578
1598 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer( 1579 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
1599 VideoCapturer* capturer) { 1580 VideoCapturer* capturer) {
1600 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer"); 1581 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
1582 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1601 if (!DisconnectCapturer() && capturer == NULL) { 1583 if (!DisconnectCapturer() && capturer == NULL) {
1602 return false; 1584 return false;
1603 } 1585 }
1604 1586
1605 { 1587 {
1606 rtc::CritScope cs(&lock_); 1588 rtc::CritScope cs(&lock_);
1607 1589
1608 // Reset timestamps to realign new incoming frames to a webrtc timestamp. A 1590 // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
1609 // new capturer may have a different timestamp delta than the previous one. 1591 // new capturer may have a different timestamp delta than the previous one.
1610 first_frame_timestamp_ms_ = 0; 1592 first_frame_timestamp_ms_ = 0;
(...skipping 12 matching lines...) Expand all
1623 // necessary to give this black frame a larger timestamp than the 1605 // necessary to give this black frame a larger timestamp than the
1624 // previous one. 1606 // previous one.
1625 last_frame_timestamp_ms_ += 1; 1607 last_frame_timestamp_ms_ += 1;
1626 black_frame.set_render_time_ms(last_frame_timestamp_ms_); 1608 black_frame.set_render_time_ms(last_frame_timestamp_ms_);
1627 stream_->Input()->IncomingCapturedFrame(black_frame); 1609 stream_->Input()->IncomingCapturedFrame(black_frame);
1628 } 1610 }
1629 1611
1630 capturer_ = NULL; 1612 capturer_ = NULL;
1631 return true; 1613 return true;
1632 } 1614 }
1633 1615 capturer_is_screencast_ = capturer->IsScreencast();
1634 capturer_ = capturer;
1635 capturer_->AddOrUpdateSink(this, sink_wants_);
1636 } 1616 }
1617 capturer_ = capturer;
1618 capturer_->AddOrUpdateSink(this, sink_wants_);
1637 return true; 1619 return true;
1638 } 1620 }
1639 1621
1640 void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) { 1622 void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
1641 rtc::CritScope cs(&lock_); 1623 rtc::CritScope cs(&lock_);
1642 muted_ = mute; 1624 muted_ = mute;
1643 } 1625 }
1644 1626
1645 bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() { 1627 bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
1646 cricket::VideoCapturer* capturer; 1628 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1647 { 1629 if (capturer_ == NULL) {
1648 rtc::CritScope cs(&lock_); 1630 return false;
1649 if (capturer_ == NULL) 1631 }
1650 return false;
1651 1632
1652 if (capturer_->video_adapter() != nullptr) 1633 capturer_->RemoveSink(this);
1653 old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes(); 1634 capturer_ = NULL;
1654 1635 // Reset |cpu_restricted_counter_| if the capturer is changed. It is not
1655 capturer = capturer_; 1636 // possible to know if the video resolution is restricted by CPU usage after
1656 capturer_ = NULL; 1637 // the capturer is changed since the next capturer might be screen capture
1657 } 1638 // with another resolution and frame rate.
1658 capturer->RemoveSink(this); 1639 cpu_restricted_counter_ = 0;
1659
1660 return true; 1640 return true;
1661 } 1641 }
1662 1642
1663 const std::vector<uint32_t>& 1643 const std::vector<uint32_t>&
1664 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const { 1644 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
1665 return ssrcs_; 1645 return ssrcs_;
1666 } 1646 }
1667 1647
1668 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions( 1648 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
1669 const VideoOptions& options) { 1649 const VideoOptions& options) {
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1815 params.options ? *params.options : parameters_.options); 1795 params.options ? *params.options : parameters_.options);
1816 return; 1796 return;
1817 } else if (params.options) { 1797 } else if (params.options) {
1818 // Reconfigure if codecs are already set. 1798 // Reconfigure if codecs are already set.
1819 if (parameters_.codec_settings) { 1799 if (parameters_.codec_settings) {
1820 SetCodecAndOptions(*parameters_.codec_settings, *params.options); 1800 SetCodecAndOptions(*parameters_.codec_settings, *params.options);
1821 return; 1801 return;
1822 } else { 1802 } else {
1823 parameters_.options = *params.options; 1803 parameters_.options = *params.options;
1824 } 1804 }
1825 } 1805 } else if (params.conference_mode && parameters_.codec_settings) {
1826 else if (params.conference_mode && parameters_.codec_settings) {
1827 SetCodecAndOptions(*parameters_.codec_settings, parameters_.options); 1806 SetCodecAndOptions(*parameters_.codec_settings, parameters_.options);
1828 return; 1807 return;
1829 } 1808 }
1830 if (recreate_stream) { 1809 if (recreate_stream) {
1831 LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters"; 1810 LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
1832 RecreateWebRtcStream(); 1811 RecreateWebRtcStream();
1833 } 1812 }
1834 } 1813 }
1835 1814
1836 webrtc::VideoEncoderConfig 1815 webrtc::VideoEncoderConfig
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1943 } 1922 }
1944 1923
1945 void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() { 1924 void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
1946 rtc::CritScope cs(&lock_); 1925 rtc::CritScope cs(&lock_);
1947 if (stream_ != NULL) { 1926 if (stream_ != NULL) {
1948 stream_->Stop(); 1927 stream_->Stop();
1949 } 1928 }
1950 sending_ = false; 1929 sending_ = false;
1951 } 1930 }
1952 1931
1932 void WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate(Load load) {
1933 if (worker_thread_ != rtc::Thread::Current()) {
1934 invoker_.AsyncInvoke<void>(
1935 worker_thread_,
1936 rtc::Bind(&WebRtcVideoChannel2::WebRtcVideoSendStream::OnLoadUpdate,
1937 this, load));
1938 return;
1939 }
1940 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1941 LOG(LS_INFO) << "OnLoadUpdate " << load;
1942 if (!capturer_) {
1943 return;
1944 }
1945 {
1946 rtc::CritScope cs(&lock_);
1947 // Do not adapt resolution for screen content as this will likely result in
1948 // blurry and unreadable text.
1949 if (capturer_is_screencast_)
1950 return;
1951
1952 rtc::Optional<int> max_pixel_count;
1953 rtc::Optional<int> max_pixel_count_step_up;
1954 if (load == kOveruse) {
1955 max_pixel_count = rtc::Optional<int>(
1956 (last_dimensions_.height * last_dimensions_.width) / 2);
1957 // Increase |number_of_cpu_adapt_changes_| if
1958 // sink_wants_.max_pixel_count will be changed since
1959 // last time |capturer_->AddOrUpdateSink| was called. That is, this will
1960 // result in a new request for the capturer to change resolution.
1961 if (!sink_wants_.max_pixel_count ||
1962 *sink_wants_.max_pixel_count > *max_pixel_count) {
1963 ++number_of_cpu_adapt_changes_;
1964 ++cpu_restricted_counter_;
1965 }
1966 } else {
1967 RTC_DCHECK(load == kUnderuse);
1968 max_pixel_count_step_up = rtc::Optional<int>(last_dimensions_.height *
1969 last_dimensions_.width);
1970 // Increase |number_of_cpu_adapt_changes_| if
1971 // sink_wants_.max_pixel_count_step_up will be changed since
1972 // last time |capturer_->AddOrUpdateSink| was called. That is, this will
1973 // result in a new request for the capturer to change resolution.
1974 if (sink_wants_.max_pixel_count ||
1975 (sink_wants_.max_pixel_count_step_up &&
1976 *sink_wants_.max_pixel_count_step_up < *max_pixel_count_step_up)) {
1977 ++number_of_cpu_adapt_changes_;
1978 --cpu_restricted_counter_;
1979 }
1980 }
1981 sink_wants_.max_pixel_count = max_pixel_count;
1982 sink_wants_.max_pixel_count_step_up = max_pixel_count_step_up;
1983 }
1984 capturer_->AddOrUpdateSink(this, sink_wants_);
1985 }
1986
1953 VideoSenderInfo 1987 VideoSenderInfo
1954 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() { 1988 WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
1955 VideoSenderInfo info; 1989 VideoSenderInfo info;
1956 webrtc::VideoSendStream::Stats stats; 1990 webrtc::VideoSendStream::Stats stats;
1991 RTC_DCHECK(thread_checker_.CalledOnValidThread());
1957 { 1992 {
1958 rtc::CritScope cs(&lock_); 1993 rtc::CritScope cs(&lock_);
1959 for (uint32_t ssrc : parameters_.config.rtp.ssrcs) 1994 for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
1960 info.add_ssrc(ssrc); 1995 info.add_ssrc(ssrc);
1961 1996
1962 if (parameters_.codec_settings) 1997 if (parameters_.codec_settings)
1963 info.codec_name = parameters_.codec_settings->codec.name; 1998 info.codec_name = parameters_.codec_settings->codec.name;
1964 for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) { 1999 for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
1965 if (i == parameters_.encoder_config.streams.size() - 1) { 2000 if (i == parameters_.encoder_config.streams.size() - 1) {
1966 info.preferred_bitrate += 2001 info.preferred_bitrate +=
1967 parameters_.encoder_config.streams[i].max_bitrate_bps; 2002 parameters_.encoder_config.streams[i].max_bitrate_bps;
1968 } else { 2003 } else {
1969 info.preferred_bitrate += 2004 info.preferred_bitrate +=
1970 parameters_.encoder_config.streams[i].target_bitrate_bps; 2005 parameters_.encoder_config.streams[i].target_bitrate_bps;
1971 } 2006 }
1972 } 2007 }
1973 2008
1974 if (stream_ == NULL) 2009 if (stream_ == NULL)
1975 return info; 2010 return info;
1976 2011
1977 stats = stream_->GetStats(); 2012 stats = stream_->GetStats();
2013 }
2014 info.adapt_changes = number_of_cpu_adapt_changes_;
2015 info.adapt_reason = cpu_restricted_counter_ <= 0
2016 ? CoordinatedVideoAdapter::ADAPTREASON_NONE
2017 : CoordinatedVideoAdapter::ADAPTREASON_CPU;
1978 2018
1979 info.adapt_changes = old_adapt_changes_; 2019 if (capturer_) {
1980 info.adapt_reason = CoordinatedVideoAdapter::ADAPTREASON_NONE; 2020 if (!capturer_->IsMuted()) {
1981 2021 VideoFormat last_captured_frame_format;
1982 if (capturer_ != NULL) { 2022 capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops,
1983 if (!capturer_->IsMuted()) { 2023 &info.capturer_frame_time,
1984 VideoFormat last_captured_frame_format; 2024 &last_captured_frame_format);
1985 capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops, 2025 info.input_frame_width = last_captured_frame_format.width;
1986 &info.capturer_frame_time, 2026 info.input_frame_height = last_captured_frame_format.height;
1987 &last_captured_frame_format);
1988 info.input_frame_width = last_captured_frame_format.width;
1989 info.input_frame_height = last_captured_frame_format.height;
1990 }
1991 if (capturer_->video_adapter() != nullptr) {
1992 info.adapt_changes += capturer_->video_adapter()->adaptation_changes();
1993 info.adapt_reason = capturer_->video_adapter()->adapt_reason();
1994 }
1995 } 2027 }
1996 } 2028 }
1997 2029
1998 // Get bandwidth limitation info from stream_->GetStats(). 2030 // Get bandwidth limitation info from stream_->GetStats().
1999 // Input resolution (output from video_adapter) can be further scaled down or 2031 // Input resolution (output from video_adapter) can be further scaled down or
2000 // higher video layer(s) can be dropped due to bitrate constraints. 2032 // higher video layer(s) can be dropped due to bitrate constraints.
2001 // Note, adapt_changes only include changes from the video_adapter. 2033 // Note, adapt_changes only include changes from the video_adapter.
2002 if (stats.bw_limited_resolution) 2034 if (stats.bw_limited_resolution)
2003 info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH; 2035 info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH;
2004 2036
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2504 rtx_mapping[video_codecs[i].codec.id] != 2536 rtx_mapping[video_codecs[i].codec.id] !=
2505 fec_settings.red_payload_type) { 2537 fec_settings.red_payload_type) {
2506 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2538 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2507 } 2539 }
2508 } 2540 }
2509 2541
2510 return video_codecs; 2542 return video_codecs;
2511 } 2543 }
2512 2544
2513 } // namespace cricket 2545 } // namespace cricket
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