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Issue 1655793003: Make cricket::VideoCapturer implement VideoSourceInterface (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed Android 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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1011 if (enable && options) { 1011 if (enable && options) {
1012 VideoSendParameters new_params = send_params_; 1012 VideoSendParameters new_params = send_params_;
1013 new_params.options.SetAll(*options); 1013 new_params.options.SetAll(*options);
1014 SetSendParameters(send_params_); 1014 SetSendParameters(send_params_);
1015 } 1015 }
1016 return true; 1016 return true;
1017 } 1017 }
1018 1018
1019 bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( 1019 bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
1020 const StreamParams& sp) const { 1020 const StreamParams& sp) const {
1021 for (uint32_t ssrc: sp.ssrcs) { 1021 for (uint32_t ssrc : sp.ssrcs) {
nisse-webrtc 2016/02/03 09:16:34 Unrelated change.
perkj_webrtc 2016/02/08 14:32:01 yes. git cl format, git cl lint.
1022 if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) { 1022 if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
1023 LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists."; 1023 LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
1024 return false; 1024 return false;
1025 } 1025 }
1026 } 1026 }
1027 return true; 1027 return true;
1028 } 1028 }
1029 1029
1030 bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability( 1030 bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability(
1031 const StreamParams& sp) const { 1031 const StreamParams& sp) const {
1032 for (uint32_t ssrc: sp.ssrcs) { 1032 for (uint32_t ssrc : sp.ssrcs) {
1033 if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) { 1033 if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) {
1034 LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc 1034 LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc
1035 << "' already exists."; 1035 << "' already exists.";
1036 return false; 1036 return false;
1037 } 1037 }
1038 } 1038 }
1039 return true; 1039 return true;
1040 } 1040 }
1041 1041
1042 bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) { 1042 bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
(...skipping 285 matching lines...) Expand 10 before | Expand all | Expand 10 after
1328 { 1328 {
1329 rtc::CritScope stream_lock(&stream_crit_); 1329 rtc::CritScope stream_lock(&stream_crit_);
1330 if (send_streams_.find(ssrc) == send_streams_.end()) { 1330 if (send_streams_.find(ssrc) == send_streams_.end()) {
1331 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; 1331 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
1332 return false; 1332 return false;
1333 } 1333 }
1334 if (!send_streams_[ssrc]->SetCapturer(capturer)) { 1334 if (!send_streams_[ssrc]->SetCapturer(capturer)) {
1335 return false; 1335 return false;
1336 } 1336 }
1337 } 1337 }
1338
1339 if (capturer) {
1340 capturer->SetApplyRotation(!ContainsHeaderExtension(
1341 send_rtp_extensions_, kRtpVideoRotationHeaderExtension));
1342 }
1343 { 1338 {
1344 rtc::CritScope lock(&capturer_crit_); 1339 rtc::CritScope lock(&capturer_crit_);
1345 capturers_[ssrc] = capturer; 1340 capturers_[ssrc] = capturer;
1346 } 1341 }
1347 return true; 1342 return true;
1348 } 1343 }
1349 1344
1350 bool WebRtcVideoChannel2::SendIntraFrame() { 1345 bool WebRtcVideoChannel2::SendIntraFrame() {
1351 // TODO(pbos): Implement. 1346 // TODO(pbos): Implement.
1352 LOG(LS_VERBOSE) << "SendIntraFrame()."; 1347 LOG(LS_VERBOSE) << "SendIntraFrame().";
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1616 video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2, 1611 video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
1617 (width + 1) / 2); 1612 (width + 1) / 2);
1618 memset(video_frame->buffer(webrtc::kYPlane), 16, 1613 memset(video_frame->buffer(webrtc::kYPlane), 16,
1619 video_frame->allocated_size(webrtc::kYPlane)); 1614 video_frame->allocated_size(webrtc::kYPlane));
1620 memset(video_frame->buffer(webrtc::kUPlane), 128, 1615 memset(video_frame->buffer(webrtc::kUPlane), 128,
1621 video_frame->allocated_size(webrtc::kUPlane)); 1616 video_frame->allocated_size(webrtc::kUPlane));
1622 memset(video_frame->buffer(webrtc::kVPlane), 128, 1617 memset(video_frame->buffer(webrtc::kVPlane), 128,
1623 video_frame->allocated_size(webrtc::kVPlane)); 1618 video_frame->allocated_size(webrtc::kVPlane));
1624 } 1619 }
1625 1620
1626 void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( 1621 void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame(
1627 VideoCapturer* capturer, 1622 const VideoFrame& frame) {
1628 const VideoFrame* frame) { 1623 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame");
1629 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame"); 1624 webrtc::VideoFrame video_frame(frame.GetVideoFrameBuffer(), 0, 0,
1630 webrtc::VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0, 1625 frame.GetVideoRotation());
1631 frame->GetVideoRotation());
1632 rtc::CritScope cs(&lock_); 1626 rtc::CritScope cs(&lock_);
1633 if (stream_ == NULL) { 1627 if (stream_ == NULL) {
1634 // Frame input before send codecs are configured, dropping frame. 1628 // Frame input before send codecs are configured, dropping frame.
1635 return; 1629 return;
1636 } 1630 }
1637 1631
1638 // Not sending, abort early to prevent expensive reconfigurations while 1632 // Not sending, abort early to prevent expensive reconfigurations while
1639 // setting up codecs etc. 1633 // setting up codecs etc.
1640 if (!sending_) 1634 if (!sending_)
1641 return; 1635 return;
1642 1636
1643 if (format_.width == 0) { // Dropping frames. 1637 if (format_.width == 0) { // Dropping frames.
1644 RTC_DCHECK(format_.height == 0); 1638 RTC_DCHECK(format_.height == 0);
1645 LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame."; 1639 LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
1646 return; 1640 return;
1647 } 1641 }
1648 if (muted_) { 1642 if (muted_) {
1649 // Create a black frame to transmit instead. 1643 // Create a black frame to transmit instead.
1650 CreateBlackFrame(&video_frame, 1644 CreateBlackFrame(&video_frame, static_cast<int>(frame.GetWidth()),
1651 static_cast<int>(frame->GetWidth()), 1645 static_cast<int>(frame.GetHeight()));
1652 static_cast<int>(frame->GetHeight()));
1653 } 1646 }
1654 1647
1655 int64_t frame_delta_ms = frame->GetTimeStamp() / rtc::kNumNanosecsPerMillisec; 1648 int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
1656 // frame->GetTimeStamp() is essentially a delta, align to webrtc time 1649 // frame->GetTimeStamp() is essentially a delta, align to webrtc time
1657 if (first_frame_timestamp_ms_ == 0) { 1650 if (first_frame_timestamp_ms_ == 0) {
1658 first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms; 1651 first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
1659 } 1652 }
1660 1653
1661 last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms; 1654 last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
1662 video_frame.set_render_time_ms(last_frame_timestamp_ms_); 1655 video_frame.set_render_time_ms(last_frame_timestamp_ms_);
1663 // Reconfigure codec if necessary. 1656 // Reconfigure codec if necessary.
1664 SetDimensions( 1657 SetDimensions(video_frame.width(), video_frame.height(),
1665 video_frame.width(), video_frame.height(), capturer->IsScreencast()); 1658 capturer_->IsScreencast());
1666 1659
1667 stream_->Input()->IncomingCapturedFrame(video_frame); 1660 stream_->Input()->IncomingCapturedFrame(video_frame);
1668 } 1661 }
1669 1662
1670 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer( 1663 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
1671 VideoCapturer* capturer) { 1664 VideoCapturer* capturer) {
1672 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer"); 1665 TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
1673 if (!DisconnectCapturer() && capturer == NULL) { 1666 if (!DisconnectCapturer() && capturer == NULL) {
1674 return false; 1667 return false;
1675 } 1668 }
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1698 format_.interval / rtc::kNumNanosecsPerMillisec; 1691 format_.interval / rtc::kNumNanosecsPerMillisec;
1699 black_frame.set_render_time_ms(last_frame_timestamp_ms_); 1692 black_frame.set_render_time_ms(last_frame_timestamp_ms_);
1700 stream_->Input()->IncomingCapturedFrame(black_frame); 1693 stream_->Input()->IncomingCapturedFrame(black_frame);
1701 } 1694 }
1702 1695
1703 capturer_ = NULL; 1696 capturer_ = NULL;
1704 return true; 1697 return true;
1705 } 1698 }
1706 1699
1707 capturer_ = capturer; 1700 capturer_ = capturer;
1701 capturer_->AddSink(this, sink_capabilities_);
pthatcher1 2016/02/03 15:38:36 I think it would be better to add capabilities() t
perkj_webrtc 2016/02/08 14:32:01 PTAL. I think I have now changed this to be as we
1708 } 1702 }
1709 // Lock cannot be held while connecting the capturer to prevent lock-order
1710 // violations.
1711 capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame);
1712 return true; 1703 return true;
1713 } 1704 }
1714 1705
1715 // TODO(pbos): Apply this on the VideoAdapter instead! 1706 // TODO(pbos): Apply this on the VideoAdapter instead!
1716 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat( 1707 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat(
1717 const VideoFormat& format) { 1708 const VideoFormat& format) {
1718 if ((format.width == 0 || format.height == 0) && 1709 if ((format.width == 0 || format.height == 0) &&
1719 format.width != format.height) { 1710 format.width != format.height) {
1720 LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not " 1711 LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not "
1721 "both, 0x0 drops frames)."; 1712 "both, 0x0 drops frames).";
(...skipping 27 matching lines...) Expand all
1749 rtc::CritScope cs(&lock_); 1740 rtc::CritScope cs(&lock_);
1750 if (capturer_ == NULL) 1741 if (capturer_ == NULL)
1751 return false; 1742 return false;
1752 1743
1753 if (capturer_->video_adapter() != nullptr) 1744 if (capturer_->video_adapter() != nullptr)
1754 old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes(); 1745 old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes();
1755 1746
1756 capturer = capturer_; 1747 capturer = capturer_;
1757 capturer_ = NULL; 1748 capturer_ = NULL;
1758 } 1749 }
1759 capturer->SignalVideoFrame.disconnect(this); 1750 capturer->RemoveSink(this);
1760 return true; 1751 return true;
1761 } 1752 }
1762 1753
1763 const std::vector<uint32_t>& 1754 const std::vector<uint32_t>&
1764 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const { 1755 WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
1765 return ssrcs_; 1756 return ssrcs_;
1766 } 1757 }
1767 1758
1768 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions( 1759 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
1769 const VideoOptions& options) { 1760 const VideoOptions& options) {
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1891 rtc::CritScope cs(&lock_); 1882 rtc::CritScope cs(&lock_);
1892 // |recreate_stream| means construction-time parameters have changed and the 1883 // |recreate_stream| means construction-time parameters have changed and the
1893 // sending stream needs to be reset with the new config. 1884 // sending stream needs to be reset with the new config.
1894 bool recreate_stream = false; 1885 bool recreate_stream = false;
1895 if (params.rtcp_mode) { 1886 if (params.rtcp_mode) {
1896 parameters_.config.rtp.rtcp_mode = *params.rtcp_mode; 1887 parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
1897 recreate_stream = true; 1888 recreate_stream = true;
1898 } 1889 }
1899 if (params.rtp_header_extensions) { 1890 if (params.rtp_header_extensions) {
1900 parameters_.config.rtp.extensions = *params.rtp_header_extensions; 1891 parameters_.config.rtp.extensions = *params.rtp_header_extensions;
1892 sink_capabilities_.can_apply_rotation = ContainsHeaderExtension(
1893 *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension);
1901 if (capturer_) { 1894 if (capturer_) {
1902 capturer_->SetApplyRotation(!ContainsHeaderExtension( 1895 capturer_->RemoveSink(this);
1903 *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension)); 1896 capturer_->AddSink(this, sink_capabilities_);
pthatcher1 2016/02/03 15:38:36 Why not just fire SignalCapabilitiesChanged()?
perkj_webrtc 2016/02/08 14:32:01 PTAL
1904 } 1897 }
1905 recreate_stream = true; 1898 recreate_stream = true;
1906 } 1899 }
1907 if (params.max_bandwidth_bps) { 1900 if (params.max_bandwidth_bps) {
1908 // Max bitrate has changed, reconfigure encoder settings on the next frame 1901 // Max bitrate has changed, reconfigure encoder settings on the next frame
1909 // or stream recreation. 1902 // or stream recreation.
1910 parameters_.max_bitrate_bps = *params.max_bandwidth_bps; 1903 parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
1911 pending_encoder_reconfiguration_ = true; 1904 pending_encoder_reconfiguration_ = true;
1912 } 1905 }
1913 // Set codecs and options. 1906 // Set codecs and options.
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2605 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2598 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2606 } 2599 }
2607 } 2600 }
2608 2601
2609 return video_codecs; 2602 return video_codecs;
2610 } 2603 }
2611 2604
2612 } // namespace cricket 2605 } // namespace cricket
2613 2606
2614 #endif // HAVE_WEBRTC_VIDEO 2607 #endif // HAVE_WEBRTC_VIDEO
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