Commit Graph

2471 Commits

Author SHA1 Message Date
5c532d3774 Robustification of the echo suppression behavior during headset usage.
This CL robustifies the echo removal behavior when headsets are used.
In particular it:
-Introduces a secondary, more refined alignment when no alignment can
be found using the delay estimator.
-Changes decision logic for when to use the linear filter output.
-Changes the decision logic for when to be transparent.
-Changes the way that the transparent mode works.
-Makes the nonlinear mode less aggressive.
-Removes the detector for non-audible echoes.
-Makes the attenuation when there are signals with strong narrowband
characteristics more mild in scenarios with low render.

Furthermore the CL:
-Removes the input of external echo leakage information.


Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406
Reviewed-on: https://webrtc-review.googlesource.com/60866
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22548}
2018-03-22 00:23:23 +00:00
96114428bd Remove unneeded migration helper.
The migration has completed and this is no longer needed.

Bug: None
Change-Id: I2ef262e78cad618e9bb664baa239d446fe8bd69d
Reviewed-on: https://webrtc-review.googlesource.com/63320
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22524}
2018-03-21 08:20:08 +00:00
db67ba1c81 Report SRTP error codes to UMA
Bug: webrtc:8996
Change-Id: I75de77ed15c2829425c00f57ebd07109803425db
Reviewed-on: https://webrtc-review.googlesource.com/63122
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22521}
2018-03-20 18:37:49 +00:00
2e0da5aa38 Remove EncodedFrame picture_id/spatial_layer references.
Bug: webrtc:8909
Change-Id: I8499968829aaa8069b92b4a3cda81bfea759f2a4
Reviewed-on: https://webrtc-review.googlesource.com/63262
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22512}
2018-03-20 12:17:35 +00:00
5b3541f9af RTCStatsCollector::GetStatsReport() with optional selector argument.
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}
2018-03-19 15:32:16 +00:00
0fa82a60e9 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId.
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.

Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
2018-03-19 15:13:11 +00:00
5f1a31c565 Adding a smooth transition from the startup phase parameter set in AEC3
This CL ensures a smooth transition from the parameters used during
the startup phase in the call to the parameters used in the rest of the
call. This is achieved by slowly transitioning between the parameter
sets via interpolation.

Bug: chromium:819240,webrtc:8983
Change-Id: Ifbac4b93fc6ad6efc441f41fb88ef09e8ee3d669
Reviewed-on: https://webrtc-review.googlesource.com/60360
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22451}
2018-03-15 13:38:16 +00:00
2cb4105224 Moved audioproc_f interface into api directory.
The interface of the audioproc_f tool should be located in the api/ directory, so it becomes visible to the outside world.

Bug: webrtc:8732
Change-Id: Ia7475883aeb0e1f7a6afa5e791204b38dc53a8b8
Reviewed-on: https://webrtc-review.googlesource.com/61801
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22449}
2018-03-15 12:31:37 +00:00
13b8bad235 Final name changing of MediaStreamInterface.label() to id().
Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().

Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
2018-03-14 20:30:52 +00:00
22e623ad68 Add configurable threshold for writability state update.
Add configurable parameters in RTCConfiguration with the default value
given by the constants CONNECTION_WRITE_CONNECT_TIME and
CONNECTION_WRITE_CONNECT_FAILURES in the ICE implementation. These two
parameters define the time period for which a candidate pair must wait
for ping response and the minimum number of connectivity checks that
the pair must send without response before its state becomes unreliable
from writable as defined in the current ICE implementation.

Bug: webrtc:8988
Change-Id: I484599b7d776489a87741ffea8926df766095da9
Reviewed-on: https://webrtc-review.googlesource.com/60704
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22411}
2018-03-13 18:54:03 +00:00
d6fbf2a4b1 Tests: Pass codec ID argument to audio codecs
Bug: webrtc:8941
Change-Id: Ia6d51dcbf7d69b38f3615e01d3f7031b8f5c31d0
Reviewed-on: https://webrtc-review.googlesource.com/58092
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22383}
2018-03-12 13:25:29 +00:00
b619936dee Stats traversal algorithm added.
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}
2018-03-12 10:54:09 +00:00
5081c0cc6d Change error handlers for Set*Description to use RTCError
Needed in order to return error codes to Chromium.

Bug: chromium:819629, chromium:589455
Change-Id: Iab22250db62a348eee21c6d8bfc44020a7380586
Reviewed-on: https://webrtc-review.googlesource.com/60522
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22367}
2018-03-09 15:37:34 +00:00
ab1aee0be4 Reland "Deprecate the adaptive level controller"
This is a reland of 6f37ed78d99daa36e964ff0a65b205f0916d9949

CQ dry run OK except for missing iOS swarming bots.
NOTRY=True

Original change's description:
> Deprecate the adaptive level controller
>
> Level control handled by default-on AGC.
>
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}

Bug: none
Change-Id: I0b9b8e2f3457d5efd4603efbfbbc6b84651df315
Reviewed-on: https://webrtc-review.googlesource.com/60720
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22352}
2018-03-09 09:42:13 +00:00
e6826d2461 Add configurable connectivity check intervals.
The connectivity check intervals for candidate pairs with strong and
weak connectivity are currently constants in the ICE implementation. A
set of suboptimal value of these constants for a given application may
result in undesirable behavior including excessive network switching
latency. This CL adds these intervals to RTCConfiguration that is
available to applications to configure, while maintaining the original
constants as their default value for compatibility with existing
applications.

Bug: webrtc:8988
Change-Id: I804b0f4cf7881be7d3c8aec2776bc9596de72482
Reviewed-on: https://webrtc-review.googlesource.com/60585
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22351}
2018-03-09 08:09:43 +00:00
12edf4ce34 Separate build target for rtc_base/numerics/safe_minmax.h
So that we can avoid dependency cycles.

Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:12:00 +00:00
513449eab9 Changes name of RtpTransceiverInit's stream_labels to stream_ids.
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.

Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
2018-03-06 23:42:01 +00:00
8e20f17d1d Report UMA metrics for received SDP format
This change adds UMA stats that record the format of the remote offered
SDP. There are three classifications for the SDP format:
- Simple: No more than one audio and one video. Should be compatible
    with both Plan B and Unified Plan endpoints.
- ComplexPlanB: More than one audio or more than one video in the Plan B
    format (e.g., one audio mline with multiple streams).
- ComplexUnifiedPlan: More than one audio or more than one video in the
    Unified Plan format (e.g., two audio mlines).

Bug: chromium:811683
Change-Id: If46394edfa6a812ef313d632e27ec27516c18867
Reviewed-on: https://webrtc-review.googlesource.com/57220
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22315}
2018-03-06 21:22:51 +00:00
70473fcac4 Reland "Add hugeFramesSent GetStats metric"
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
52f8188f5d Revert "Deprecate the adaptive level controller"
This reverts commit 6f37ed78d99daa36e964ff0a65b205f0916d9949.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Deprecate the adaptive level controller
> 
> Level control handled by default-on AGC.
> 
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}

TBR=solenberg@webrtc.org,saza@webrtc.org,aleloi@webrtc.org

Change-Id: Ic52f41fcbebfd2291a51b17ac788313e1ceef163
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/60240
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22308}
2018-03-06 11:54:22 +00:00
6f37ed78d9 Deprecate the adaptive level controller
Level control handled by default-on AGC.

Bug: none
Change-Id: I405daeceece12c896d41156b649fcfd556726f77
Reviewed-on: https://webrtc-review.googlesource.com/59682
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22305}
2018-03-06 10:20:01 +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
6b54228e8b Adapt audio codec factory templates to the new codec pair ID arguments
We use template magic to let them handle both the presence and absence
of the new argument. This will be removed in a later CL, when we can
assume that new argument is always present.

Bug: webrtc:8941
Change-Id: I2d47f7c8572a9f03e742401dcf491b948b161f63
Reviewed-on: https://webrtc-review.googlesource.com/58081
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22301}
2018-03-06 01:13:10 +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
f9d0f1d215 UMA counters for SDES x media type
These counters will register whether the media sections
used with SDES are for audio, video or data.

Bug: chromium:804275
Change-Id: I1da3bb6625af755c0897bf4cd349655cb283fbb6
Reviewed-on: https://webrtc-review.googlesource.com/59400
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22287}
2018-03-05 13:46:43 +00:00
3f693b9e75 Delete unused method SetPeriodicKeyFrames.
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.

Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
2018-03-05 08:54:32 +00:00
845e87877e Name change from stream label to stream id for spec compliance.
Bug: webrtc:7932
Change-Id: I66f33597342394083256f050cac2a00a68042302
Reviewed-on: https://webrtc-review.googlesource.com/59280
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@{#22276}
2018-03-02 20:44:48 +00:00
17668ec4a5 Audio codec implementations: Take optional codec ID argument
None of the built-in codecs do anything with the ID, but callers will
soon require them to accept it.

Bug: webrtc:8941
Change-Id: I0eb77db82d72c7d34cff639fecb67c1e6ec421bf
Reviewed-on: https://webrtc-review.googlesource.com/58089
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22259}
2018-03-01 19:28:38 +00:00
98900740ad Audio codec factories: Pass a codec pair ID to new codecs
Currently ignored by all implementations and callers, but future CLs
will remedy that.

Bug: webrtc:8941
Change-Id: I59a3af78fefcf35af3e5ef37d2adf1165ce5751e
Reviewed-on: https://webrtc-review.googlesource.com/58080
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22248}
2018-03-01 12:23:28 +00:00
8447e91429 Add a hysteresis for the API call skew detection to better handle jittery platforms
Bug: webrtc:8954,chromium:817313
Change-Id: I940d52ac96e5bddf886d47be089a1991ae24b51b
Reviewed-on: https://webrtc-review.googlesource.com/58640
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22228}
2018-02-28 14:02:43 +00:00
0efa941d2f Move EchoCanceller3Factory to api/auido
The AEC3 factory is now part of the WebRTC API.

Bug: webrtc:8844
Change-Id: If6f419b4ca0354e2d346c0e6474086e456ba747e
Reviewed-on: https://webrtc-review.googlesource.com/57141
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22204}
2018-02-27 14:09:59 +00:00
8c913efc1b Removed using FrameObject = EncodedFrame.
Bug: webrtc:8909
Change-Id: Ia825466c52bdd2b0e070c3949cadaba05af7c1bc
Reviewed-on: https://webrtc-review.googlesource.com/57140
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22201}
2018-02-27 12:35:19 +00:00
461cdf0dd1 Added AEC3 parameter for specifying that the setup has clock drift
Bug: webrtc:8940
Change-Id: Ic29af185ecb212d7d27bafc8432033bdd35c6d3b
Reviewed-on: https://webrtc-review.googlesource.com/58060
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22199}
2018-02-27 08:59:03 +00:00
ab6ea6b7b1 Miscellaneous updates to PeerConnectionInterface comments
NoTry: true
Bug: None
Change-Id: I9cf4f77e4775ff9b97c71f6a9bdb2d7f159e7048
Reviewed-on: https://webrtc-review.googlesource.com/57240
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22192}
2018-02-26 22:27:52 +00:00
1807d57ab8 Add application_data field(s) to RtpPacketToSend and PacketOptions.
Pass pointer to application_data from RtpPacketToSend arriving via RtpSender::SendToNetwork through to Transport::SendRtp, in PacketOptions.

Bug: webrtc:8906
Change-Id: Ie75013ed472710f4efcfbcc160e46a6119a1f41d
Reviewed-on: https://webrtc-review.googlesource.com/55600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22174}
2018-02-23 17:20:46 +00:00
f35c6667d6 Separate build targets for aec3 and aec3_unittests
Bug: webrtc:8844
Change-Id: Id6a98eae19aaedc87c3f402a004f58f0290d5c28
Reviewed-on: https://webrtc-review.googlesource.com/56580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22173}
2018-02-23 13:16:16 +00:00
225c787c6e Move default thresholds from QualityScaler to encoders.
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.

Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
2018-02-23 13:12:36 +00:00
e7c891f953 Renamed FrameObject to EncodedFrame.
The plan is to:
 1. Move FrameObject to api/video.
 2. Rename FrameObject to EncodedFrame.
 3. Move EncodedFrame out of the video_coding namespace.

This is the 2nd CL.

Bug: webrtc:8909
Change-Id: I5e76a0a3b306156b8bc1de67834b4adf14bebef9
Reviewed-on: https://webrtc-review.googlesource.com/56182
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22158}
2018-02-22 16:12:48 +00:00
5d436ac0bf Removed Die mock from MockAudioEncoder
MockAudioEncoder was calling a mocked Die function on itself in its
destructor. This outputs "Uninteresting mock function call" warning if
the Die call was not expected. This is true even if a NiceMock is used
to suppress the warnings.

The purpose of testing that the destructor is called might be to protect
against memory leaks when audio encoder ownership is transferred using a
raw pointer. However, this case is already covered by msan checks.

Bug: None
Change-Id: I0603c417b4b239027859228e05ebcf83ff5aaf18
Reviewed-on: https://webrtc-review.googlesource.com/56183
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22146}
2018-02-22 12:53:38 +00:00
d5a272ff51 Create public EncodedFrame interface.
The plan is to:
 1. Move FrameObject to api/video.
 2. Rename FrameObject to EncodedFrame.
 3. Move EncodedFrame out of the video_coding namespace.

This is the 1st CL.

Bug: webrtc:8909
Change-Id: I2e5100eda6c51bcefb32295e03b73cf1f5c213a4
Reviewed-on: https://webrtc-review.googlesource.com/55560
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22135}
2018-02-21 16:24:15 +00:00
72a43a1d2c Collect packet loss and RTT stats of STUN binding requests.
STUN candidates use STUN binding requests to keep NAT bindings open.
Related stats including packet loss and RTT can be now collected via the
legacy GetStats in PeerConnection.

Bug: None
Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c
Reviewed-on: https://webrtc-review.googlesource.com/54100
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22113}
2018-02-21 00:49:26 +00:00
db53f8e604 Add configurable STUN binding request interval.
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.

TBR=sakal@webrtc.org

Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
2018-02-20 23:32:46 +00:00
b6b00dc180 Safe behavior of the initial echo removal in AEC3
This CL adds functionality to allow removal of any echo occurring
before the render and capture signals have been properly aligned.
The functionality is added in such a manner that the transparency
to nearend is maintained as much as possible.


Bug: webrtc:8883
Change-Id: I813cbbc4c48822e7dffcd9ab6233be4c222089de
Reviewed-on: https://webrtc-review.googlesource.com/49941
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22107}
2018-02-20 22:01:36 +00:00
0dd1b0a4b2 Revert "Revert "Enables PeerConnectionFactory using external fec controller""
This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122.

Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before.

Original change's description:
> Revert "Enables PeerConnectionFactory using external fec controller"
>
> This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3.
>
> Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java
>
> Original change's description:
> > Enables PeerConnectionFactory using external fec controller
> >
> > Bug: webrtc:8799
> > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> > Reviewed-on: https://webrtc-review.googlesource.com/43961
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22038}
>
> TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
>
> Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8799
> Reviewed-on: https://webrtc-review.googlesource.com/54080
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22040}

TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org

Bug: webrtc:8799
Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8
Reviewed-on: https://webrtc-review.googlesource.com/55400
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:41:55 +00:00
9d138fc7ce Drop dependency of common_video on api:libjingle_peerconnection_api.
Deleting the apparently unused include of api/rtp_headers from
common/video/include/video_frame.h broke the PayloadRouter and
VideoSendStream code under video/. Missing declaration of the
RtpPayloadState struct declared in api/rtp_headers.h. Moving the
declaration of that struct to payload_router.h (outside of the api),
since it's used only internally in video/, and that seemed to be a
more logical place for it.

Bug: webrtc:7504
Change-Id: Ibed8233dfeea8bdf144db5422cdf897da824d6ee
Reviewed-on: https://webrtc-review.googlesource.com/53701
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22080}
2018-02-19 13:20:24 +00:00
de25a9a4be Delete dummy target peerconnection_and_implicit_call_api.
Depends on chromium cl
https://chromium-review.googlesource.com/c/chromium/src/+/921642

Bug: webrtc:7504, webrtc:8667
Change-Id: I3bbe13863d5828a216dc6ffd3024d31930cacf79
Reviewed-on: https://webrtc-review.googlesource.com/53863
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22075}
2018-02-19 10:56:52 +00:00
2ae140ae7e BUILD.gn file for api/audio.
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.

Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-19 10:38:29 +00:00
8b815cddca Add OnTrack PeerConnection callback for Unified Plan
This adds a callback corresponding to the ontrack event as defined
in the WebRTC specification.

Bug: webrtc:7600
Change-Id: Ied8c55e11dcea864428fb194623c1595c21657c7
Reviewed-on: https://webrtc-review.googlesource.com/52660
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22066}
2018-02-17 01:03:59 +00:00
57858b3be0 Reland "Update RTCStatsCollector to work with RtpTransceivers"
Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
> 
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}

Bug: webrtc:8764
Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3
Reviewed-on: https://webrtc-review.googlesource.com/54160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22064}
2018-02-17 00:01:39 +00:00
c453c16dbe Add stubs for AEC3 factory.
This CL creates empty placeholders for EchoCanceller3Factory. This
allows for moving the factory of AEC3 as soon as downstream has been
updated to include echo_canceller3_factory.h.

Bug: webrtc:8844
Change-Id: I77c53d8257291f189c637e1c9ed76c4e74be1858
Reviewed-on: https://webrtc-review.googlesource.com/53862
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22050}
2018-02-16 12:02:48 +00:00