Jim Gustafson
7c9970cacb
Remove lbred experiment
2024-06-06 15:10:30 -07:00
Jim Gustafson
a170a82bb0
Update to use Opus 1.5
2024-04-05 14:07:50 -07:00
Jim Gustafson
c37ca3fc86
Merge branch m122
2024-02-14 22:44:28 -08:00
Jim Gustafson
6e5158df93
m120 merge fixes
...
- Use worker_thread TaskQueue for channel operations
- Fix use of deprecated DNS resolver
- Restore quantization of audio levels
- Simplify crypto options change
- Move channel blocking operations to pc
- Sync opus for merge
2024-01-24 09:14:46 -08:00
Philipp Hancke
de17252e8e
Reland "Unify access to SDP codec parameters"
...
This is a reland of commit 63d03f586b
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
2024-01-03 12:03:11 +00:00
Mirko Bonadei
6c9c958c69
Revert "Unify access to SDP codec parameters"
...
This reverts commit 63d03f586b
.
Reason for revert: Breaks downstream project (not backwards compatible API change)
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I841735d98533d3b66850b9cfcf7ee0a99ddde078
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41377}
2023-12-13 16:28:44 +00:00
Philipp Hancke
63d03f586b
Unify access to SDP codec parameters
...
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
2023-12-13 14:22:15 +00:00
inaqui-signal
fa4fd71354
Merge branch 'm118'
2023-11-07 15:00:28 -06:00
Jim Gustafson
62d543d814
Add low bitrate redundancy support
2023-10-31 13:14:36 -07:00
Jim Gustafson
7da0a87124
Add more audio control and safe defaults
2023-08-23 10:42:30 -07:00
inaqui-signal
c570368abc
Merge branch 'm116' into 5845
2023-08-09 14:40:20 -05:00
Jim Gustafson
281e582847
Add function to check if packet represents speech
...
The original code assumed that one packet contains one frame, which is not
true anymore since multi-frame packets and DTX are now supported.
Includes an updated reference to signalapp/opus so that DTX frames are not
padded.
2023-05-12 09:01:29 -07:00
Rashad Sookram
147fdb9f46
Merge branch 'm112' into 5615
2023-04-27 12:45:13 -04:00
Jared Siskin
c018bae807
Format /modules
...
git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^modules/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs
Bug: webrtc:15082
Change-Id: I2c3cd28740062794f8c10e39d8406aadb9e9a35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jared Siskin <jtsiskin@meta.com>
Cr-Commit-Position: refs/heads/main@{#39901}
2023-04-20 02:02:45 +00:00
Rashad Sookram
03ddb5df82
Merge branch 'm110' into 5481
2023-02-17 11:35:29 -05:00
Jakob Ivarsson
757da3cf70
Stop setting OPUS_SIGNAL_VOICE when DTX is enabled.
...
This was done in crbug.com/webrtc/4559 since "CELT-only mode does not have DTX", but that should not be the case anymore (support was added in Opus v1.2.1).
One exception where DTX does not work is with OPUS_APPLICATION_AUDIO (used with stereo) and low complexity settings. This should not be a common config.
Bug: None
Change-Id: I1476083b836bcabeb73df83d5bf06c3878146d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288420
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38923}
2022-12-20 11:06:48 +00:00
Per Kjellander
e0b4cab69c
Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead
...
Bug: webrtc:6762
Change-Id: I520188a13ee5f50c441226574ccb3df54f842835
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285300
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38783}
2022-11-30 20:19:36 +00:00
Jakob Ivarsson
918eb19303
Fix crash when Opus maxptime < 20ms.
...
A follow up cl will be created to better handle nullopt frame length range in AudioSendStream.
Note that maxptime is still not used for setting the frame length (only ptime is).
Bug: chromium:1109337
Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38702}
2022-11-22 01:21:24 +00:00
Rashad Sookram
2e9c66e1b1
Finish resolving merge conflicts
2022-11-11 19:10:59 -05:00
Rashad Sookram
bd151651d3
Merge in M108
2022-11-11 17:02:35 -05:00
Artem Titov
e39115a0ca
Migrate audio perf tests on new perf metrics export API
...
Bug: b/246095034
Change-Id: Id659e43c116428cab47d334c93a6036f74dbb8e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276626
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38192}
2022-09-25 18:55:50 +00:00
Rashad Sookram
9ed4194e49
Update to 5005 (M102) ( #86 )
2022-08-24 11:07:33 -04:00
Ali Tofigh
714e3cbb48
Adopt absl::string_view in modules/audio_coding/
...
Bug: webrtc:13579
Change-Id: Ifec66fb6ba9724d18539de7245a358c2d13c7939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268547
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37573}
2022-07-20 13:34:23 +00:00
Oleh Prypin
cc7bd85748
Don't add libopus to public_deps, its headers are only used directly
...
Bug: webrtc:8603
Change-Id: I2ce1f96a80dd23e420b3693b899d2b14382fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266765
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Oleh Prypin <oprypin@google.com>
Cr-Commit-Position: refs/heads/main@{#37363}
2022-06-28 19:13:14 +00:00
Niels Möller
ea1e6f44f8
Delete rtc_base/format_macros.h
...
It defined RTC_PRIuS, which was needed for compatibility with MSVC
prior to version 2015.
Bug: webrtc:6424
Change-Id: I5668d473376201cad3e8da65927c967fc397804b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261314
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36814}
2022-05-09 12:03:21 +00:00
Florent Castelli
c3e6e3a3e8
Remove dependency on rtc_base_approved from most targets
...
Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
Peter Thatcher
4a2e0e5d45
Update to 4896 (M100) ( #72 )
2022-04-15 17:13:23 -06:00
Florent Castelli
4467ad7835
Remove //rtc_base:macromagic from public deps
...
Bug: webrtc:8603
Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-05 12:36:12 +00:00
Byoungchan Lee
604fd2f1ab
Remove RTC_DISALLOW_COPY_AND_ASSIGN from modules/
...
Bug: webrtc:13555, webrtc:13082
Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35771}
2022-01-24 11:50:20 +00:00
Ivo Creusen
deb1b1bc70
Always call IsOk() to ensure audio codec configuration is valid when negotiating.
...
We should avoid creating codecs with invalid parameters, since this can
expose security issues. For many codecs the IsOk() method to check the
codec config is only called in DCHECKs. This CL ensures IsOk() is always
called, also in non-debug builds.
Bug: chromium:1265806
Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35422}
2021-11-26 10:11:21 +00:00
Peter Thatcher
4af16890e5
Fix AudioEncoder
2021-10-29 09:07:06 -06:00
Peter Thatcher
1a0b210a9d
ks
2021-10-29 07:49:26 -06:00
Artem Titov
cfea2182f8
Use backticks not vertical bars to denote variables in comments
...
Bug: webrtc:12338
Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34696}
2021-08-10 10:40:03 +00:00
Artem Titov
d00ce747c7
Use backticks not vertical bars to denote variables in comments for /modules/audio_coding
...
Bug: webrtc:12338
Change-Id: I02613d9fca45d00e2477f334b7a0416e7912e26b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227037
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34621}
2021-08-02 10:45:40 +00:00
Jesús de Vicente Peña
d674ec77af
Not dropping the refresh DTX packets but substituting them by 1 byte packets.
...
Bug: webrtc:12380
Change-Id: I27029c591ac2555d6ae61b706adcf97c9498a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217880
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33983}
2021-05-11 19:47:34 +00:00
Yura Yaroshevich
d46a174f0c
Expose adaptive_ptime from iOS SDK.
...
Bug: None
Change-Id: I48fd0937f51dc972b3eccd66f99ae80378e32fe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214968
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33766}
2021-04-18 21:53:32 +00:00
Peter Thatcher
13cb722ac9
Update to WebRTC 4389 (e7d9f74)
...
Contains changes for M86 (4240), M87 (4280), M88 (4324), and M89 (4389).
2021-04-16 13:26:31 -07:00
Jesús de Vicente Peña
3b9abd8dee
Avoiding the noise pumping during DTX regions by just forwarding the refresh DTX packets that decrease the comfort noise level at the decoder.
...
Bug: webrtc:12380
Change-Id: I60e4684150cb4880224f402a9bf42a72811863b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202920
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33174}
2021-02-05 10:05:25 +00:00
Peter Thatcher
52871c1f98
Avoid CHECK when decreasing the ptime ( #44 )
2021-01-22 11:08:46 -08:00
Peter Thatcher
22490bf0b9
Fix configuration of ptime so it's not overridden in the future ( #43 )
2021-01-22 09:41:43 -08:00
Peter Thatcher
84775e0ab6
Allow control of min and max bitrates ( #42 )
2021-01-21 10:49:55 -08:00
Peter Thatcher
f3ce387c3b
Make audio encoder changes more resilient to BWE changes ( #41 )
2021-01-20 16:58:31 -08:00
Peter Thatcher
1ff88760c8
Add PeerConnection::ConfigureAudioEncoders ( #40 )
...
Allows more dynamic, easy, and precise control of OPUS from PeerConnection.
2021-01-20 11:52:27 -08:00
Minyue Li
c940870b72
Revert "opus: take SILK vad result into account for voice detection"
...
This reverts commit 686a3709ac
.
Reason for revert: crbug.com/1144220
Original change's description:
> opus: take SILK vad result into account for voice detection
>
> BUG=webrtc:11643
>
> Change-Id: Idc3a9b6bb7bd1a33f905843e5d6067ae19d5172c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176508
> Commit-Queue: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31743}
TBR=devicentepena@webrtc.org ,minyue@webrtc.org,fippo@sip-communicator.org
Bug: webrtc:11643
Change-Id: I9c77e4f6e919c4b648a5783edf4188e1f8114602
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191485
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32542}
2020-11-04 07:29:48 +00:00
Jakob Ivarsson
36274f9158
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
...
This is a reland of 1dbe30c7e8
Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}
Bug: webrtc:6762
Change-Id: I6d79894a213fc42d2338409e7513247725881b1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191221
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32534}
2020-11-02 11:05:56 +00:00
Björn Terelius
d546186b89
Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
...
This reverts commit 1dbe30c7e8
.
Reason for revert: Speculative revert due to failing tests.
Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841
>
> Original change's description:
> > Default enable WebRTC-SendSideBwe-WithOverhead.
> >
> > Bug: webrtc:6762
> > Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Ali Tofigh <alito@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32472}
>
> Bug: webrtc:6762
> Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32492}
TBR=stefan@webrtc.org ,jakobi@webrtc.org,alito@webrtc.org
Change-Id: I7e0378788576236059627cf8c3bad58cd70aff7e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32504}
2020-10-27 10:51:46 +00:00
Jakob Ivarsson
1dbe30c7e8
Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
...
This is a reland of 87c1950841
Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}
Bug: webrtc:6762
Change-Id: Icf096a8755d29600a13bd08b1f22f5a79de21e90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190143
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32492}
2020-10-26 12:35:47 +00:00
Jakob Ivarsson
27af3c4c24
Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
...
This reverts commit 87c1950841
.
Reason for revert: breaks downstream tests
Original change's description:
> Default enable WebRTC-SendSideBwe-WithOverhead.
>
> Bug: webrtc:6762
> Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Ali Tofigh <alito@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32472}
TBR=stefan@webrtc.org ,jakobi@webrtc.org,alito@webrtc.org
Change-Id: If59fd41dcd8f6db76ea297c34c25fe19ae2ae973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189973
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32474}
2020-10-22 16:57:18 +00:00
Jakob Ivarsson
87c1950841
Default enable WebRTC-SendSideBwe-WithOverhead.
...
Bug: webrtc:6762
Change-Id: I18ace06a33b3b60d5a19796d4769f70cd977d604
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188801
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32472}
2020-10-22 13:37:18 +00:00
Jeremy Leconte
c8850cbf55
Change gtest name to allow filtering based on the story name.
...
It is meant for Pinpoint to run only the relevant tests when running a bisection.
The Pinpoint side of this change can be found here:
https://crrev.com/c/2404161
Bug: webrtc:11084
Change-Id: I466f39816b83e2f83a3a49845c99605f4d5a857b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183763
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32082}
2020-09-11 14:11:27 +00:00