Commit graph

157 commits

Author SHA1 Message Date
Jim Gustafson
49c96f3e79 Merge remote-tracking branch 'google/branch-heads/6478' 2024-06-21 16:31:45 -07:00
Jim Gustafson
c43adafcd5 Merge m123/6312 2024-06-12 22:25:35 -07:00
Keiichi Enomoto
a70274a82f Remove duplicated parentheses from deprecated attribute
These lines cause an error when building a project with libwebrtc as a dependency in Microsoft Visual Studio.

Bug: webrtc:15864
Change-Id: I1abfe257d0ea1c16c4c5b718594e8085036f7763
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342320
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41881}
2024-03-12 10:58:59 +00:00
Jim Gustafson
c37ca3fc86 Merge branch m122 2024-02-14 22:44:28 -08:00
Jakob Ivarsson
9d9b3a3553 Add option for the audio encoder to allocate a bitrate range.
Bug: webrtc:15834
Change-Id: I3223162e747621983dbe2704661bb87f7890b3fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338221
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#41682}
2024-02-07 09:47:16 +00:00
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
Harald Alvestrand
78f905e5cc Move some users to use webrtc::RefCountInterface
Bug: webrtc:15622
Change-Id: I2d4c20c726af1a052e161b7689a73d1e5e3eb191
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325526
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41067}
2023-11-02 14:45:57 +00:00
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
Jared Siskin
bceec84aee Format ^(api|call|common_audio|examples|media|net|p2p|pc)/
half of the remaining folders

git ls-files | grep -e  "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs

Bug: webrtc:15082
Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39977}
2023-05-03 11:09:26 +00:00
Rashad Sookram
147fdb9f46 Merge branch 'm112' into 5615 2023-04-27 12:45:13 -04:00
Rashad Sookram
03ddb5df82 Merge branch 'm110' into 5481 2023-02-17 11:35:29 -05:00
Jakob Ivarsson
6e1ae443ac Don't use low complexity Opus on all ARM devices.
Bug: none
Change-Id: I4be504ffa271e3a5879cec9efe153b1f895a96c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288401
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38921}
2022-12-20 10:30:30 +00:00
Henrik Lundin
8754a3c945 Update some audio modules with new OWNERS
Bug: b/260832909
Change-Id: I3d2ebad978988eabf228475c3fc46708e12cf5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285780
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38788}
2022-12-01 14:55:38 +00:00
Christoffer Jansson
b00f88179e Remove xooglers from WATCHLISTS and OWNERS
Bug: b/260832909
Change-Id: I683c714da35c21c23404d4b1c6500da28d680ed5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285470
Commit-Queue: Christoffer Jansson <jansson@google.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38777}
2022-11-30 15:33:25 +00:00
Alessio Bazzica
17887eb04a Reland "[ACM] iSAC audio codec removed"
This is a reland of commit b46c4bf27b

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ia22c4d7724b6022238235fede93e36e570a49376
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283843
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38665}
2022-11-17 12:52:35 +00:00
Alessio Bazzica
fbeb76ab51 Revert "[ACM] iSAC audio codec removed"
This reverts commit b46c4bf27b.

Reason for revert: breaks a downstream project

Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}

Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: 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@{#38655}
2022-11-16 20:40:52 +00:00
Alessio Bazzica
b46c4bf27b [ACM] iSAC audio codec removed
Note: this CL has to leave behind one part of iSAC, which is its VAD
currently used by AGC1 in APM. The target visibility has been
restricted and the VAD will be removed together with AGC1 when the
time comes.

Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319

Bug: webrtc:14450
Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38652}
2022-11-16 16:42:55 +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
Mirko Bonadei
5c9b7da038 Add missing dependencies.
Bug: b/251890128
Change-Id: Ia9312797a5552ad1ceb4a80968014b849121a1b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278580
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38333}
2022-10-10 10:18:37 +00:00
Mirko Bonadei
9d9c2d5795 Make header files self contained.
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.

Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
2022-10-08 08:38:36 +00:00
Rashad Sookram
9ed4194e49
Update to 5005 (M102) (#86) 2022-08-24 11:07:33 -04:00
Niels Möller
7c8c4db8ea Add rtc::make_ref_counted to api/
This cl adds a forwarding header, a build target, and migrates headers
in api/ to use it.

Moving actual implementation, will follow, in
https://webrtc-review.googlesource.com/c/src/+/265390.

Bug: webrtc:12701
Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37200}
2022-06-13 15:53:27 +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
Florent Castelli
a30aef3dea Move event_tracer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-24 14:47:40 +00:00
Florent Castelli
f9c5984a1d Move buffer out of rtc_base_approved
Bug: webrtc:9838
Change-Id: I14feff7b1f0182d031b6644d281be44122820ce7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259307
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36629}
2022-04-22 21:19:28 +00:00
Peter Thatcher
4a2e0e5d45
Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
Florent Castelli
57aa81bce7 Remove //rtc_base:stringutils from public deps
Bug: webrtc:8603
Change-Id: Ic2dfbe28d310cb4b35983b73e895fc95e8439669
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257913
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@{#36453}
2022-04-05 22:42:19 +00:00
Florent Castelli
e10a9f609a Remove //rtc_base:safe_conversions from public deps
Bug: webrtc:8603
Change-Id: I285ac30975039f8fe9882d1673cc8e4a615c8618
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257912
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@{#36452}
2022-04-05 20:04:59 +00:00
Florent Castelli
f86f6f9afd Remove //rtc_base:refcount from public deps
Bug: webrtc:8603
Change-Id: Ib27a107ae809df739492846175f0e9c4af40d21a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257910
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36447}
2022-04-05 15:32:29 +00:00
Jonas Oreland
e62c2f2c77 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 12/inf
rename WebRtcKeyValueConfig to FieldTrialsView

Bug: webrtc:10335
Change-Id: If725bd498c4c3daf144bee638230fa089fdde833
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256965
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36365}
2022-03-29 10:14:00 +00:00
Jonas Oreland
6e2b9e2210 WebRTC-DeprecateGlobalFieldTrialString/Enabled/ - part 5/inf
Add field trials to audio api.

It is added as a pointer with nullptr as default.
It is not (yet) used anywhere.
Usage of field trials comes in subsequent patches.

Bug: webrtc:10335
Change-Id: Icbe22d95c356a6fefde34590f11ea63f005ab09e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36213}
2022-03-16 09:11:43 +00:00
Byoungchan Lee
c065e739e2 Remove RTC_DISALLOW_COPY_AND_ASSIGN more.
Bug: webrtc:13555, webrtc:13082
Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35749}
2022-01-20 11:00:18 +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
Ivo Creusen
d823259c7f Set the maximum number of audio channels to 24
The number of audio channels can be configured in SDP, and can thus be
set to arbitrary values. However, the audio code has limitations that
prevent a high number of channels from working well in practice.

Bug: chromium:1265806
Change-Id: I6f6c3f68a3791bb189a614eece6bd0ed7874f252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237807
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35359}
2021-11-16 17:01:54 +00:00
Artem Titov
d3251968d1 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.

Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
2021-11-15 21:44:59 +00:00
Peter Thatcher
1a0b210a9d ks 2021-10-29 07:49:26 -06:00
Artem Titov
0e61fdd27c Use backticks not vertical bars to denote variables in comments for /api
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
2021-07-26 18:27:34 +00:00
Mirko Bonadei
190244bb59 Remove all #include <assert.h>/<cassert> and usage in Obj-C code.
This CL completes the removal of assert() and relative headers from
the codebase (excluded
//examples/objc/AppRTCMobile/third_party/SocketRocket which is in a
third_party sub-directory).

Bug: webrtc:6779
Change-Id: I93ed57168d2c0e011626873d66529488c5f484f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225546
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34528}
2021-07-22 14:00:26 +00:00
Mirko Bonadei
25ab3228f3 Replace assert() with RTC_DCHECK().
CL partially auto-generated with:

git grep -l "\bassert(" | grep "\.[c|h]" | \
  xargs sed -i 's/\bassert(/RTC_DCHECK(/g'

And with:

git grep -l "RTC_DCHECK(false)" |  \
  xargs sed -i 's/RTC_DCHECK(false)/RTC_NOTREACHED()/g'

With some manual changes to include "rtc_base/checks.h" where
needed.

A follow-up CL will remove assert() from Obj-C code as well
and remove the #include of <assert.h>.

The choice to replace with RTC_DCHECK is because assert()
is because RTC_DCHECK has similar behavior as assert()
based on NDEBUG.

This CL also contains manual changes to switch from
basic RTC_DCHECK to other (preferred) versions like
RTC_DCHECK_GT (and similar).

Bug: webrtc:6779
Change-Id: I00bed8886e03d685a2f42324e34aef2c9b7a63b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34442}
2021-07-09 07:49:43 +00:00
Tomas Gunnarsson
c1d589146b Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
2021-04-23 12:04:39 +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
Anton Bikineev
a3575cb848 Remove tautological 'unsigned expr < 0' comparisons
This is the result of compiling Chromium with
Wtautological-unsigned-zero-compare. For more details, see:
https://chromium-review.googlesource.com/c/chromium/src/+/2802412

Change-Id: I05cec6ae5738036a56beadeaa1dde5189edf0137
Bug: chromium:1195670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213783
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33689}
2021-04-12 11:40:14 +00:00
Danil Chapovalov
e904161cec Replace RTC_DEPRECATED with ABSL_DEPRECATED
This remove webrtc-specific macro that has no reason to be webrtc specific
ABSL_DEPRECATED takes a message parameter encouraging to write text how class or function is deprecated.

Bug: webrtc:12484
Change-Id: I89f1398f91dacadc37f7db469dcd985e3724e444
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208282
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33314}
2021-02-22 12:53:23 +00:00
Pablo Barrera González
ff0e01f689 Implement audio_interruption metric for kCodecPlc
Audio interruption metric is not implemented for codecs doing their own PLC.

R=ivoc@webrtc.org, jakobi@webrtc.org

Bug: b/177523033 webrtc:12456
Change-Id: I0aca6fa5c0ff617e76ee1e4ed8d95703c7097223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206561
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@google.com>
Cr-Commit-Position: refs/heads/master@{#33229}
2021-02-11 09:37:24 +00:00
Peter Thatcher
84775e0ab6
Allow control of min and max bitrates (#42) 2021-01-21 10:49:55 -08:00