49ac5959c2
Add GetSources to VideoRtpReceiver
...
BUG=webrtc:9770
Change-Id: I16143fce6eb727bbab0f6c621aa5b51bc6d28d6b
Reviewed-on: https://webrtc-review.googlesource.com/101600
Reviewed-by: Seth Hampson <shampson@webrtc.org >
Reviewed-by: Åsa Persson <asapersson@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24858}
2018-09-27 10:00:40 +00:00
84df1c724e
Make fewer copies when using StringBuilder.
...
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00
bfd412ef71
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
...
This change passes a pointer (non-owning) down to the MediaChannel when set
in the RtpSender / RtpReceiver. This currently is not used to encrypt frames.
Bug: webrtc:9681
Change-Id: I385fa8b948427803cd3f9cef918c31d7754d1b4f
Reviewed-on: https://webrtc-review.googlesource.com/97000
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Emad Omara <emadomara@webrtc.org >
Commit-Queue: Benjamin Wright <benwright@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24694}
2018-09-11 20:10:44 +00:00
366a50c4ef
Remove simple stringstream usages.
...
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
f5f5373372
Delete unused member MediaSenderInfo::packets_cached.
...
Bug: None
Change-Id: I06e1a2010cc0af4b8a4ea726078fea6b67fa84d5
Reviewed-on: https://webrtc-review.googlesource.com/93281
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24263}
2018-08-10 13:13:54 +00:00
e41c433502
Move sigslot to proper third_party directory
...
Extract sigslot into separate target and move it to proper third_party
directory.
Bug: webrtc:8366
Change-Id: Id2e0712bd020bfad811947803c94553dce06d976
Reviewed-on: https://webrtc-review.googlesource.com/84141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 14:53:33 +00:00
665174fdbb
Reformat the WebRTC code base
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Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
00c7183614
Replace rtc::Optional with absl::optional in media, ortc, p2p
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This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'media ortc p2p':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I19167714af7cc1436d34cfcba6c8b3718d8e677b
Reviewed-on: https://webrtc-review.googlesource.com/83731
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23638}
2018-06-16 07:09:59 +00:00
97e04884bd
Delete unused stats for preferred_bitrate.
...
Bug: webrtc:8830
Change-Id: Iaa30488255f2e09e269274136d370740cd030902
Reviewed-on: https://webrtc-review.googlesource.com/78880
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23529}
2018-06-07 08:11:07 +00:00
0327c2ddc1
Move VideoStreamEncoderInterface to api/.
...
Bug: webrtc:8830
Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0
Reviewed-on: https://webrtc-review.googlesource.com/74481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 19:50:37 +00:00
65ec0fc81e
Delete unneeded includes of basictypes.h.
...
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
c948fe62fd
Delete unneeded includes of call/video_config.h.
...
Bug: webrtc:8830
Change-Id: I6114b47e5524a6d2450108388236478b1ceafb67
Reviewed-on: https://webrtc-review.googlesource.com/77425
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23295}
2018-05-18 09:00:56 +00:00
c6ce9c5938
New file api/video/BUILD.gn
...
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.
Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
11b34f4d08
Remove chromium clang style errors affecting sdk/android/media_jni
...
Bug: webrtc:163
Change-Id: I1e98174817ca032ee13f9a6a386803382843389d
Reviewed-on: https://webrtc-review.googlesource.com/67360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Paulina Hensman <phensman@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22796}
2018-04-09 13:55:49 +00:00
ff40b142c0
Delete obsolete enable argument to SetVideoSend.
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This argument was previously used to implement track muting
(black frames) in the video engine, but that now happens in
the VideoTrack/VideoBroadcaster upstream.
Bug: webrtc:6983
Change-Id: Ib721b297d9fbe55b641c56690dbbd37a52edbb2f
Reviewed-on: https://webrtc-review.googlesource.com/67341
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22785}
2018-04-09 08:45:29 +00:00
5897a6ec6a
Adds support for signaling a=msid lines without a=ssrc lines.
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Currently in the SDP we require an a=ssrc line in the m= section in
order for a StreamParams object to be created with that
MediaContentDescription. This change creates a StreamParams object
without ssrcs in the case that a=msid lines are signaled, but ssrcs
are not. When the remote description is set, this allows us to store
the "unsignaled" StreamParams object in the media channel to later
be used when the first packet is received and we create the
receive stream.
Bug: webrtc:7932, webrtc:7933
Change-Id: Ib6734abeee62b8ed688a8208722c402134c074ef
Reviewed-on: https://webrtc-review.googlesource.com/63141
Commit-Queue: Seth Hampson <shampson@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22712}
2018-04-03 21:21:11 +00:00
003930a3ce
Fix MID not always getting set with audio
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Bug: webrtc:4050
Change-Id: I543a9f70c6c7fd10cd177ce16eba6c335db367ec
Reviewed-on: https://webrtc-review.googlesource.com/65020
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Zhi Huang <zhihuang@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22681}
2018-03-29 20:22:28 +00:00
bb50ce5bb6
Wire up MID send value to the PeerConnection API
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Bug: webrtc:4050
Change-Id: I522cf8621e2cb639f54be2402174befd23e4af59
Reviewed-on: https://webrtc-review.googlesource.com/60962
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22610}
2018-03-26 18:14:30 +00:00
70473fcac4
Reland "Add hugeFramesSent GetStats metric"
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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}
2018-03-06 13:38:11 +00:00
8ddc2e6258
Revert "Add hugeFramesSent GetStats metric"
...
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}
2018-03-06 08:28:52 +00:00
f9f71b91ae
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}
2018-03-05 15:09:12 +00:00
f9c5cf65f6
Renamed rtc::RateLimiter to rtrc::DataRateLimiter.
...
This removes a confusing name collision between webrtc::RateLimiter
and rtc::RateLimiter where the header file names were separated only by
an underscore.
Bug: None
Change-Id: Ifcf0a4e62b2bf3bd9057714d7c536f7609ad1b79
Reviewed-on: https://webrtc-review.googlesource.com/58741
Commit-Queue: Sebastian Jansson <srte@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#22237}
2018-02-28 19:58:24 +00:00
f120cba82d
Delete AudioMonitor and related code.
...
Bug: webrtc:8760
Change-Id: I0b11ec66b0f2576f52866864ba046191034a4d2d
Reviewed-on: https://webrtc-review.googlesource.com/39003
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Noah Richards <noahric@chromium.org >
Cr-Commit-Position: refs/heads/master@{#21801}
2018-01-30 09:48:29 +00:00
ba37b4b075
Change return type of RtpSenderInterface::SetParameters from bool to RTCError
...
Note that RTCErrorTypes are mapped to the following DOMException names:
INTERNAL_ERROR -> OperationError
UNSUPPORTED_PARAMETER -> OperationError
INVALID_STATE -> InvalidStateError
INVALID_MODIFICATION -> InvalidModificationError
INVALID_RANGE -> RangeError
Bug: webrtc:8772
Change-Id: I44e3fe2456b007b8fb227d37d74b07ba226a19e4
Reviewed-on: https://webrtc-review.googlesource.com/37141
Commit-Queue: Zach Stein <zstein@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21766}
2018-01-25 19:06:04 +00:00
b8e1201020
Generate track stats when SSRC=0
...
This will generate an all-zeroes track stat when the sender
has not yet been connected (SSRC has not been assigned).
Bug: webrtc:8673
Change-Id: Id59e6941bc87eba6bb33b4d2a8fd808d985052c7
Reviewed-on: https://webrtc-review.googlesource.com/43080
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21734}
2018-01-23 16:15:58 +00:00
6daa278156
Move MediaConfig to its own header file and target.
...
To eliminate circular dependencies, we need to eliminate the include
of media/base/mediachannel.h from api/peerconnectioninterface.h.
MediaConfig is one of the types the PeerConnection api depends on,
since it's part of PeerConnectionInterface::RTCConfiguration. It's
formally a public member, but the intention is that applications should use
accessor mehtods on RTCConfiguration and never access the contents of
MediaConfig directly.
Bug: webrtc:7504
Change-Id: Idfab6f69132d6b90d1628fa4543a393e22db79ac
Reviewed-on: https://webrtc-review.googlesource.com/41260
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Patrik Höglund <phoglund@google.com >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21731}
2018-01-23 11:02:16 +00:00
1d7ecd29c7
Rename a few MediaConfig::Video flags for consistency.
...
enable_cpu_overuse_detection --> enable_cpu_adaptation
disable_prerenderer_smoothing --> enable_prerenderer_smoothing
where the latter also gets opposite meaning.
Bug: none
Change-Id: Ic10de0871a87e86a899aefa72ecb7e46fcdeaa65
Reviewed-on: https://webrtc-review.googlesource.com/40280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21726}
2018-01-22 17:32:58 +00:00
a6fe261b97
Move AudioOptions to its own header file and target.
...
It is part of our api.
With the intention to later delete the inclusion of mediachannel.h from
api/peerconnectioninterface.h, and eliminate circular dependencies.
Bug: webrtc:7504
Change-Id: If44efd14d85675530e457760a1c4a1d338f931b7
Reviewed-on: https://webrtc-review.googlesource.com/41281
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21694}
2018-01-19 13:00:32 +00:00
6539f69746
Add VideoSendStream::Config::EncoderSettings::experiment_cpu_load_estimator.
...
And wire it up to methods on RTCConfiguration, via MediaConfig::Video.
Bug: webrtc:8504
Change-Id: I30805ee20c11d1d2fe552eb81f16d514db0ba4a8
Reviewed-on: https://webrtc-review.googlesource.com/39786
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21670}
2018-01-18 10:42:07 +00:00
adc1e9bf94
Remove old videosourceinterface files.
...
I have one downstream CL that needs to land first before landing this.
Bug: webrtc:6828
Change-Id: Ib6f3ae78f83775278e4c2e4d34a93fe3748fb851
Reviewed-on: https://webrtc-review.googlesource.com/38340
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21567}
2018-01-11 08:23:33 +00:00
002f921c5d
Inline default constructors for MediaChannel structs
...
Bug: None
Change-Id: I72b534c49d3f26e988d1c92aae09435a9483a930
Reviewed-on: https://webrtc-review.googlesource.com/37143
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21540}
2018-01-10 01:31:40 +00:00
dc8b5ab350
Remove dead code for media channel errors
...
Bug: None
Change-Id: Ifb8f2cd42a5e24ce8386eff97435890766bbd5fc
Reviewed-on: https://webrtc-review.googlesource.com/37142
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org >
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21507}
2018-01-06 00:25:29 +00:00
be214a26f8
Move videosinkinterface.h to common_video to solve a circular dep.
...
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Commit-Queue: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
7622048be2
Add an AudioOptions field to force software echo cancellation on iOS.
...
This is a temporary hack for the iPad Pro 12.9" gen2, which has
non-functional echo cancellation.
Bug: webrtc:8682
Change-Id: I646deeeb4723c4accac6f364c5c76a015791e202
Reviewed-on: https://webrtc-review.googlesource.com/35680
Commit-Queue: Jonathan Yu <yujo@chromium.org >
Reviewed-by: Henrik Andreassson <henrika@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21417}
2017-12-22 00:04:43 +00:00
3e113438b1
Fix circular dependencies in webrtc_common.
...
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.
I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.
Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00
3a233744eb
Reland "Remove the aec_quality_min metric."
...
This is a reland of 99b1bd1553d442ef7d27755567594ac7e65c53b7
Original change's description:
> Remove the aec_quality_min metric.
>
> Removing this unused metric.
>
> Bug: webrtc:8563
> Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
> Reviewed-on: https://webrtc-review.googlesource.com/30720
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org >
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#21158}
Bug: webrtc:8563
Change-Id: I622df96528cd6e54e252b22a315840e12d521c7f
Reviewed-on: https://webrtc-review.googlesource.com/31780
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21201}
2017-12-11 10:36:55 +00:00
9a44f96ea7
Delete rtc_base/window.h.
...
Bug: webrtc:6424
Change-Id: Iaed83b07dd469a9990f48fe41fcdff5e7493eb31
Reviewed-on: https://webrtc-review.googlesource.com/31480
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21194}
2017-12-11 07:59:35 +00:00
a3fad93d87
Revert "Remove the aec_quality_min metric."
...
This reverts commit 99b1bd1553d442ef7d27755567594ac7e65c53b7.
Reason for revert: breaks downstream projects.
Original change's description:
> Remove the aec_quality_min metric.
>
> Removing this unused metric.
>
> Bug: webrtc:8563
> Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
> Reviewed-on: https://webrtc-review.googlesource.com/30720
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org >
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#21158}
TBR=solenberg@webrtc.org ,gustaf@webrtc.org
Change-Id: I90f16915d517123e4bfba39db64424cdcc4ef03f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8563
Reviewed-on: https://webrtc-review.googlesource.com/31360
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21160}
2017-12-08 11:45:50 +00:00
99b1bd1553
Remove the aec_quality_min metric.
...
Removing this unused metric.
Bug: webrtc:8563
Change-Id: I47446d6aaf5dcc3a8ea57f9248576d68bbe2a304
Reviewed-on: https://webrtc-review.googlesource.com/30720
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21158}
2017-12-08 10:48:49 +00:00
606a5971e3
Remove adjust_agc_delta from WebRtcVoiceEngine
...
The setting is no longer used anywhere.
Bug: None
Change-Id: Id4143ca0a565472a4f08905c06f5d3f7d5dfb756
Reviewed-on: https://webrtc-review.googlesource.com/31100
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org >
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#21151}
2017-12-07 23:06:19 +00:00
aba85d1f53
Resolve circular dependency in rtc_media_base.
...
This one was pretty straightforward fortunately.
Bug: webrtc:6828
Change-Id: Ie7b5e71f1298c409dbca2c74eaa09c0986e41d8f
Reviewed-on: https://webrtc-review.googlesource.com/25821
Commit-Queue: Patrik Höglund <phoglund@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20914}
2017-11-28 15:28:58 +00:00
56d460902e
Use the new AudioProcessing statistics everywhere.
...
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.
Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 18:17:39 +00:00
c3ed630560
Add stats googHasEnteredLowResolution.
...
Indicates if the forced sw fallback has had an effect (or would have had an effect if it had been
enabled).
Bug: webrtc:6634
Change-Id: I574b9001a2fae650fb894a1caa0d0f84257658e3
Reviewed-on: https://webrtc-review.googlesource.com/23300
Reviewed-by: Stefan Holmer <stefan@webrtc.org >
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Åsa Persson <asapersson@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20729}
2017-11-17 13:02:07 +00:00
c97cf03ede
Removes unused sample-rate APIs from the ADM.
...
The following four methods are removed:
SetRecordingSampleRate(const uint32_t samplesPerSec)
RecordingSampleRate(uint32_t* samplesPerSec) const
SetPlayoutSampleRate(const uint32_t samplesPerSec)
PlayoutSampleRate(uint32_t* samplesPerSec) const
Bug: webrtc:7306
Change-Id: I2c3c2e7bd3fb1264da197699fd5de15ab6c35c1b
Reviewed-on: https://webrtc-review.googlesource.com/22001
Commit-Queue: Henrik Andreassson <henrika@webrtc.org >
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20703}
2017-11-16 08:59:53 +00:00
5f5918f4ef
Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks.
...
OnTransportOverChanged is merged into OnNetworkRouteChanged in MediaChannel
because the transport overhead will be added to rtc::NetworkRoute structure.
This CL depends on https://webrtc-review.googlesource.com/c/src/+/13520
Bug: None
Change-Id: I6ed6583f6c91db4ce61a89406de39774239f3a04
Reviewed-on: https://webrtc-review.googlesource.com/15200
Commit-Queue: Zhi Huang <zhihuang@webrtc.org >
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org >
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20678}
2017-11-14 20:42:36 +00:00
e78bcb97c3
Enable cpplint in media/
...
Bug: webrtc:5584
Change-Id: I2fd1395d35596d9002e19cc90fcda3a5d4cde9e7
Reviewed-on: https://webrtc-review.googlesource.com/16564
Commit-Queue: Steve Anton <steveanton@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20504}
2017-10-31 17:46:42 +00:00
b0a0207838
Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
...
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}
2017-10-02 10:47:00 +00:00
9a2e906b0c
Added RTCMediaStreamTrackStats.concealmentEvents
...
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}
2017-09-18 08:58:06 +00:00
92ea95e34a
Fixing WebRTC after moving from src/webrtc to src/
...
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}
2017-09-15 05:02:56 +00:00
bb547203bf
Moving src/webrtc into src/.
...
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}
2017-09-15 04:25:06 +00:00
