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This exposes the stats selection algorithm[1] on the PeerConnection.
Per-spec, there are four flavors of getStats():
1. RTCPeerConnection.getStats().
2. RTCPeerConnection.getStats(MediaStreamTrack selector).
3. RTCRtpSender.getStats().
4. RTCRtpReceiver.getStats().
1) is the parameterless getStats() which is already shipped.
2) is the same as 3) and 4) except the track is used to look up the
corresponding sender/receiver to use as the selector.
3) and 4) perform stats collection with a filter, which is implemented
in RTCStatsCollector.GetStatsReport(selector).
For technical reasons, it is easier to place GetStats() on the
PeerConnection where the RTCStatsCollector lives than to place it on the
sender/receiver. Passing the selector as an argument or as a "this"
makes little difference other than style. Wiring Chrome up such that the
JavaScript APIs is like the spec is trivial after GetStats() is added to
PeerConnectionInterface.
This CL also adds comments documenting our intent to deprecate and
remove the legacy GetStats() APIs some time in the future.
[1] https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
Bug: chromium:680172
Change-Id: I09316ba6f20b25d4f9c11785d0a1a1262d6062a1
Reviewed-on: https://webrtc-review.googlesource.com/62900
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22602}
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This implements the stats selection algorithm[1] in RTCStatsCollector by
obtaining the selector's inbound-rtp/outbound-rtp stats and performing
the stats traversal algorithm (TakeReferencedStats)[2] on a copy of the
cached report with the rtps as starting point.
Changes:
- RTCStatsCollector.GetStatsReport() with selector arguments added.
- RequestInfo added, "callbacks_" is replaced by "requests_".
- RTCStatsReport.Copy() added.
- New test for sender selector and receiver selector,
RTCStatsCollectorTest.GetStatsWithSelector.
[1] https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
[2] https://cs.chromium.org/chromium/src/third_party/webrtc/pc/rtcstatstraversal.h
Bug: chromium:680172
Change-Id: I9eff00738a1f24c94c9c8ecd13c1304452e962cf
Reviewed-on: https://webrtc-review.googlesource.com/62141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22499}
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This is part of the work to add a selector argument to getStats().
Changes:
- TakeReferencedStats() added, which traverses the stats graph and takes
any stats from the report that are directly or indirectly accessible
from the starting stats objects in the stats graph. The result is
returned as a stats report.
- GetStatsReferencedIds(), an efficient helper function for getting
neighbor stats object IDs.
- RTCStatsReport::Take(), removed the stats object with the given ID and
returns ownership of it (so that it can be added to another report).
TakeReferencedStats() is tested with a bunch of sample stats graphs.
GetStatsReferencedIds() is tested in the rtcstats_integrationttest.cc,
making sure the expected IDs are returned. The expected IDs are the
values of the stats object members with the "Id" or "Ids" suffix.
Design doc:
https://docs.google.com/document/d/18BywbtXgHCjsbR5nWBedpzqDjAfXrFSTJNiADnzoK0w/edit?usp=sharing
Bug: chromium:680172
Change-Id: I5da9da8250da0cb05adb864015901393a4290776
Reviewed-on: https://webrtc-review.googlesource.com/60869
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22381}
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This is a reland of f9f71b91ae073fdd2b89ff9df1204835aa3137eb
after the change in chromium tests.
Chromium change done here:
https://chromium-review.googlesource.com/c/chromium/src/+/950776
Original reviewed on: https://webrtc-review.googlesource.com/c/src/+/54420
No changes to the original patchset were done.
TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,solenberg@webrtc.org
Bug: webrtc:8901
Change-Id: Ic88c3cb963dceea0426eb90519743e3c1a4533c1
Reviewed-on: https://webrtc-review.googlesource.com/60140
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22310}
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This reverts commit f9f71b91ae073fdd2b89ff9df1204835aa3137eb.
Reason for revert: Looks like it's breaking WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Mac%20Tester/48322 (win and lin testers are also failing on the same test).
[ RUN ] WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise
[12743:4099:0305/082149.300326:WARNING:notification_platform_bridge_mac.mm(510)] AlertNotificationService: XPC connection invalidated.
[12743:88323:0305/082150.773242:WARNING:embedded_test_server.cc(228)] Request not handled. Returning 404: /favicon.ico
[12743:775:0305/082150.774044:INFO:CONSOLE(13)] "Requesting doGetUserMedia: constraints: {"audio":true,"video":true}", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082150.969262:INFO:CONSOLE(13)] "Returning request-callback-granted to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082150.983959:INFO:CONSOLE(13)] "Returning ok-got-stream to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.741587:INFO:CONSOLE(13)] "Requesting doGetUserMedia: constraints: {"audio":true,"video":true}", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.749225:INFO:CONSOLE(13)] "Returning request-callback-granted to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.754982:INFO:CONSOLE(13)] "Returning ok-got-stream to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.761516:INFO:CONSOLE(13)] "Returning ok-peerconnection-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:775:0305/082151.762047:WARNING:RTCPeerConnection.cpp(1151)] mediaConstraints is not a supported argument to addStream.
[12752:775:0305/082151.762096:WARNING:RTCPeerConnection.cpp(1153)] mediaConstraints was
[12743:775:0305/082151.762953:INFO:CONSOLE(13)] "Added local stream.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.763010:INFO:CONSOLE(13)] "Returning ok-added to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.767078:INFO:CONSOLE(13)] "Returning ok-peerconnection-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12755:775:0305/082151.767614:WARNING:RTCPeerConnection.cpp(1151)] mediaConstraints is not a supported argument to addStream.
[12755:775:0305/082151.767660:WARNING:RTCPeerConnection.cpp(1153)] mediaConstraints was
[12743:775:0305/082151.768452:INFO:CONSOLE(13)] "Added local stream.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.768523:INFO:CONSOLE(13)] "Returning ok-added to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.776171:INFO:CONSOLE(13)] "Returning ok-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.777197:INFO:CONSOLE(13)] "Returning ok-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:42755:0305/082151.777736:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 104 to 127
[12752:42755:0305/082151.777766:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 106 to 125
[12752:42755:0305/082151.777829:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 103 to 124
[12752:42755:0305/082151.777850:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 105 to 123
[12743:775:0305/082151.778835:INFO:CONSOLE(13)] "createOffer(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.779780:INFO:CONSOLE(13)] "Returning ok-{"type":"offer","sdp":"v=0\r\no=- 3491235150284933882 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:3632917417 cname:J9N+OjIJeArKjXXh\r\na=ssrc:3632917417 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 6e608085-751b-4945-8982-6f4aedf7bef6\r\na=ssrc:3632917417 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:3632917417 label:6e608085-751b-4945-8982-6f4aedf7bef6\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 124 127 123 125 107 108\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420032\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=127\r\na=rtpmap:125 red/90000\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 ulpfec/90000\r\na=ssrc-group:FID 1955312265 3021315394\r\na=ssrc:1955312265 cname:J9N+OjIJeArKjXXh\r\na=ssrc:1955312265 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:1955312265 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:1955312265 label:7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:3021315394 cname:J9N+OjIJeArKjXXh\r\na=ssrc:3021315394 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:3021315394 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:3021315394 label:7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\nm=application 9 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n"} to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.781514:INFO:CONSOLE(13)] "setLocalDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12755:41731:0305/082151.782411:WARNING:channel.cc(1039)] Trying to cache the Absolute Send Time extension id but the SRTP is not active.
[12752:43011:0305/082151.884258:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620da600:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12752:43011:0305/082151.884438:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(1, 65536) failed: 0
[12752:43011:0305/082151.884481:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(2, 65536) failed: 0
[12752:43011:0305/082151.884513:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12755:41731:0305/082151.922410:WARNING:channel.cc(1039)] Trying to cache the Absolute Send Time extension id but the SRTP is not active.
[12743:775:0305/082151.924626:INFO:CONSOLE(13)] "createAnswer(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.925506:INFO:CONSOLE(13)] "Returning ok-{"type":"answer","sdp":"v=0\r\no=- 6096510228474213355 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS 7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:3234277340 cname:PfS0qqt1exijuETX\r\na=ssrc:3234277340 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 9ddb9a77-20aa-42ba-8540-9e32f3dbb0af\r\na=ssrc:3234277340 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:3234277340 label:9ddb9a77-20aa-42ba-8540-9e32f3dbb0af\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 124 127 123 125 107 108\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420032\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=127\r\na=rtpmap:125 red/90000\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 ulpfec/90000\r\na=ssrc-group:FID 3517790794 302440277\r\na=ssrc:3517790794 cname:PfS0qqt1exijuETX\r\na=ssrc:3517790794 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:3517790794 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:3517790794 label:2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:302440277 cname:PfS0qqt1exijuETX\r\na=ssrc:302440277 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:302440277 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:302440277 label:2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\nm=application 9 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\nb=AS:30\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n"} to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.925954:INFO:CONSOLE(13)] "Receiving remote stream...", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.926204:INFO:CONSOLE(13)] "setRemoteDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.939935:INFO:CONSOLE(13)] "Returning ok-verified to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.940232:INFO:CONSOLE(13)] "setLocalDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:43011:0305/082151.942049:WARNING:p2ptransportchannel.cc(1093)] SetOption(1, 65536) failed: 0
[12752:43011:0305/082151.942084:WARNING:p2ptransportchannel.cc(1093)] SetOption(2, 65536) failed: 0
[12743:775:0305/082151.946009:INFO:CONSOLE(13)] "Receiving remote stream...", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.946327:INFO:CONSOLE(13)] "setRemoteDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.946367:INFO:CONSOLE(13)] "Returning ok-accepted-answer to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.950048:INFO:CONSOLE(368)] "Still ICE gathering - waiting...", source: http://127.0.0.1:50666/webrtc/peerconnection.js (368)
[12755:41731:0305/082152.030690:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(1, 65536) failed: 0
[12755:41731:0305/082152.030759:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(2, 65536) failed: 0
[12755:41731:0305/082152.030785:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12743:775:0305/082152.048464:INFO:CONSOLE(13)] "Returning [{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 65179 typ host generation 0 ufrag X54X network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag X54X network-id 1","sdpMid":"audio","sdpMLineIndex":0}] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.049868:INFO:CONSOLE(13)] "Returning ok-received-candidates to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.050468:INFO:CONSOLE(13)] "Returning [{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 60484 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 62030 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"video","sdpMLineIndex":1},{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 50175 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"data","sdpMLineIndex":2},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"video","sdpMLineIndex":1},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"data","sdpMLineIndex":2}] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.052841:INFO:CONSOLE(13)] "Returning ok-received-candidates to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.053385:INFO:CONSOLE(13)] "Returning ["codec","inbound-rtp","outbound-rtp","peer-connection","stream","track","data-channel","transport","local-candidate","remote-candidate","candidate-pair","certificate"] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.061797:INFO:CONSOLE(13)] "Returning Test failed: Error: stats.hugeFramesSent is not a whitelisted member: 0
at failTest (http://127.0.0.1:50666/webrtc/test_functions.js:46:15)
at verifyStatsIsWhitelisted_ (http://127.0.0.1:50666/webrtc/peerconnection_getstats.js:386:13)
at http://127.0.0.1:50666/webrtc/peerconnection_getstats.js:273:9 to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc:533: Failure
Value of: base::StartsWith(result, "ok-", base::CompareCase::SENSITIVE)
Actual: false
Expected: true
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc:138: Failure
Value of: value
Actual: false
Expected: true
BrowserTestBase received signal: Segmentation fault: 11. Backtrace:
0 browser_tests 0x0000000105c700cc base::debug::StackTrace::StackTrace(unsigned long) + 28
1 browser_tests 0x0000000106271902 content::(anonymous namespace)::DumpStackTraceSignalHandler(int) + 226
2 libsystem_platform.dylib 0x00007fffa63ccb3a _sigtramp + 26
3 ??? 0x0000000000000000 0x0 + 0
4 browser_tests 0x0000000102ee29e3 WebRtcTestBase::VerifyStatsGeneratedPromise(content::WebContents*) const + 467
5 browser_tests 0x0000000102edb4d1 WebRtcBrowserTest_RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise_Test::RunTestOnMainThread() + 817
6 browser_tests 0x000000010627162d content::BrowserTestBase::ProxyRunTestOnMainThreadLoop() + 557
7 browser_tests 0x0000000105da0d23 ChromeBrowserMainParts::PreMainMessageLoopRunImpl() + 4227
8 browser_tests 0x0000000105d9fb9e ChromeBrowserMainParts::PreMainMessageLoopRun() + 62
9 browser_tests 0x0000000104a3a3d3 content::BrowserMainLoop::PreMainMessageLoopRun() + 67
10 browser_tests 0x0000000104df0dc7 content::StartupTaskRunner::RunAllTasksNow() + 39
11 browser_tests 0x0000000104a38d35 content::BrowserMainLoop::CreateStartupTasks() + 661
12 browser_tests 0x0000000104a3c8f0 content::BrowserMainRunnerImpl::Initialize(content::MainFunctionParams const&) + 96
13 browser_tests 0x0000000104a36c94 content::BrowserMain(content::MainFunctionParams const&) + 180
14 browser_tests 0x0000000105c3ebb9 content::ContentMainRunnerImpl::Run() + 377
15 browser_tests 0x000000010784a8f4 service_manager::Main(service_manager::MainParams const&) + 2324
16 browser_tests 0x0000000105c3e094 content::ContentMain(content::ContentMainParams const&) + 68
17 browser_tests 0x0000000106271216 content::BrowserTestBase::SetUp() + 2550
18 browser_tests 0x0000000105d2993e InProcessBrowserTest::SetUp() + 398
19 browser_tests 0x0000000104032b51 testing::Test::Run() + 97
20 browser_tests 0x0000000104033770 testing::TestInfo::Run() + 288
21 browser_tests 0x0000000104033cd7 testing::TestCase::Run() + 263
22 browser_tests 0x000000010403b167 testing::internal::UnitTestImpl::RunAllTests() + 903
23 browser_tests 0x000000010403adb3 testing::UnitTest::Run() + 163
24 browser_tests 0x0000000105d41c67 base::TestSuite::Run() + 167
25 browser_tests 0x0000000105c63755 ChromeTestSuiteRunner::RunTestSuite(int, char**) + 37
26 browser_tests 0x00000001062b6597 content::LaunchTests(content::TestLauncherDelegate*, unsigned long, int, char**) + 391
27 browser_tests 0x0000000105c63c3c LaunchChromeTests(unsigned long, content::TestLauncherDelegate*, int, char**) + 348
28 browser_tests 0x0000000105c636ce main + 94
29 libdyld.dylib 0x00007fffa61bd235 start + 1
30 ??? 0x000000000000000a 0x0 + 10
Original change's description:
> Add hugeFramesSent GetStats metric
>
> Bug: webrtc:8901
> Change-Id: I36021c1160c3426d3bfa0f37ff0adaa35710b93e
> Reviewed-on: https://webrtc-review.googlesource.com/54420
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22290}
TBR=solenberg@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,hta@webrtc.org
Change-Id: I6a7501c46f928281d357da37f9232bb92c5a4f19
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8901
Reviewed-on: https://webrtc-review.googlesource.com/60120
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22303}
|
|
Bug: webrtc:8901
Change-Id: I36021c1160c3426d3bfa0f37ff0adaa35710b93e
Reviewed-on: https://webrtc-review.googlesource.com/54420
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22290}
|
|
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.
Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).
This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.
Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.
TBR=phoglund@webrtc.org
Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
|
|
"/*"""
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.
Reason for revert:
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660
Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
>
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
>
> Reason for revert:
>
> Breaks Chrome FYI:
>
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
> -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
> static_library(target_name) {
> ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
> //third_party/webrtc/*
> //third_party/webrtc_overrides/*
> ]
>
> https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
>
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> >
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> >
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> >
> > BUG=webrtc:8254
> >
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
>
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org
Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
|
|
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
Reason for revert:
Breaks Chrome FYI:
/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
-> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
static_library(target_name) {
^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
//third_party/webrtc/*
//third_party/webrtc_overrides/*
]
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
>
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
>
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
>
> BUG=webrtc:8254
>
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
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This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
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This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Id521896c3c43595349021c857bec216e429a0c8d
Reviewed-on: https://webrtc-review.googlesource.com/32780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21264}
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This reverts commit a9e0924fa7688c4e4558e179c6608ce1093e15f8.
Reason for revert: Breaks because of RTC_LAST_SYSTEM_ERROR
Original change's description:
> Fix circular dependencies between optional, array_view, and rtc_base.
>
> This splits things out of rtc_base and makes dependencies explicit.
>
> Bug: webrtc:6828
> Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
> Reviewed-on: https://webrtc-review.googlesource.com/31940
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21245}
TBR=phoglund@webrtc.org,kwiberg@webrtc.org
Change-Id: I1a5dcf2223f00ae7c46f9f2a12b990ab3a84397d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6828
Reviewed-on: https://webrtc-review.googlesource.com/32760
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21251}
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This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
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Bug: None
Change-Id: Id73fd4e4f83f2942cb1b394908ccc729933977cf
Reviewed-on: https://webrtc-review.googlesource.com/29020
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21044}
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Details:
* Add RTCNetworkType enum
* Add network_type to local ice candidate stats
Bug: webrtc:8435
Change-Id: Idb872849c09ad49c8f759d02afdc825e397afa07
Reviewed-on: https://webrtc-review.googlesource.com/14680
Commit-Queue: Gary Liu <qinghualiu@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20822}
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Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay
Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
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The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.
Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
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In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
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