d14f98f635
Update m1 arm64 audio bitexactness tests hashes
...
Follow up for https://webrtc-review.googlesource.com/c/src/+/232061/5 . Updated mac M1 tests that was missed because they are not part of CQ
Bug: b/199885455
Change-Id: I77618ac2869ba601f322857f4391b63220d20252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232220
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35021}
2021-09-17 08:59:05 +00:00
e742d8c163
Updae bitexactness tests to match new canonical results
...
This CL has to be submitted together with chromium roll
Roll chromium_revision 48501b3f18..a098e1265b (917742:921682)
Change log: 48501b3f18..a098e1265b48501b3f18..a098e1265bbdbd6f899c..ecf6ff89bff90eed6a5e..fabb3638a77de86cbf37..a5b6419452edf77b2d2d..44ea7aba6a1b17fd57e6..e8d5d843cf59835db543..7adcb6a6bfa299c990bc..21d7875bec2@0.10.cr0 ..version:2@1.1.2.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotation: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_check_api: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_core: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_errorprone_error_prone_type_annotations: version:2@2.7.1.cr0..version:2@2.9.0.cr0
* src/third_party/android_deps/libs/com_google_guava_guava: version:2@30.1-jre.cr0..version:2@30.1.1-jre.cr0
* src/third_party/androidx: TnotTDnWGUJDh0mSOMrgnIwzbfWFiwo2NTtr2SlUJ0MC..QRXHawrfyCC8nr_u3VCPoilYsqHO01yv1EiduoQUPcgC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/dddb60eb97..27a3328a37
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d9a9ebbe89..6b072fac88
* src/third_party/depot_tools: 789dfc223b..9c67b230207482c98f15..5b626281f1955c7f837e..159c9ad23e280366ba6a..6602cbb706ece15d049f..3f443830bd15a75b4530..65a1751e5b49ebc996aa..0896c348738420673b4c..de98afcd4abdf3dd3f28..62d7d0c928e3721e5cf9..a27c09f45b48501b3f18..a098e1265b48501b3f18..a098e1265bTBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com ,marpan@webrtc.org , jianj@chromium.org ,
Bug: b/199885455
Change-Id: I64329c57949e9233f5cf65167f012db0637fe4fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#35008}
2021-09-15 18:38:41 +00:00
3abd10889f
Fix define if chain in audio decoder unittest
...
Follow up https://webrtc-review.googlesource.com/c/src/+/228247 . Turned out "#elif defined(WEBRTC_MAC) && defined(WEBRTC_ARCH_ARM64) // M1 Mac" branch was unreachable
Bug: webrtc:13053
Change-Id: Icf1aa5147347a1fad0dce8cca893bb3c598f658e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228381
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34699}
2021-08-10 13:00:33 +00:00
8c654aa059
Update bit exactness tests to match changes
...
Follow up for https://webrtc-review.googlesource.com/c/src/+/227773 , updating M1 checksums that were not updated in the previous CL.
Example M1 failed run: https://ci.chromium.org/ui/p/webrtc/builders/ci/MacARM64%20M1%20Release/401/overview
Bug: webrtc:13053
Change-Id: I111d1d3c4bf5828ee499f20799b527ca916d77e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228247
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Andrey Logvin <landrey@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34697}
2021-08-10 12:19:13 +00:00
ee96675eda
Reland "Roll chromium_revision de0d050e64..42d795c24f (908789:908899)"
...
This is a reland of e369928e040623340e04390ce933e5dc491af22f
Original change's description:
> Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
>
> Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
> Commit-Queue: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34658}
Bug: None
Change-Id: Ibc843ef6e4e50d9d62b6b3550d5cde6eaebc02e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227773
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34662}
2021-08-06 13:51:07 +00:00
d937f50e20
Revert "Roll chromium_revision de0d050e64..42d795c24f (908789:908899)"
...
This reverts commit e369928e040623340e04390ce933e5dc491af22f.
Reason for revert: Breaks downstream project
Original change's description:
> Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
>
> Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
> Commit-Queue: Artem Titov <titovartem@webrtc.org >
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
> Cr-Commit-Position: refs/heads/master@{#34658}
TBR=mbonadei@webrtc.org ,ivoc@webrtc.org ,titovartem@webrtc.org ,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Id01cdb6a6344d7d08ee38fb152cb209a4705aa39
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227772
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34659}
2021-08-06 11:07:58 +00:00
e369928e04
Roll chromium_revision de0d050e64..42d795c24f (908789:908899)
...
Change log: de0d050e64..42d795c24fde0d050e64..42d795c24ff14f1b7600..6551b66fbfe360729c13..496f4dc82b2965e1969a..fda9d9017836299f559a..cb835b4820e99cff4446..aec4ec11c20a4dd4181a..cc487710bb00e6f338d0..7d0822e5b15219d6859a..667c51bbcade0d050e64..42d795c24fde0d050e64..42d795c24fhttps://webrtc-review.googlesource.com/c/src/+/227770
Commit-Queue: Artem Titov <titovartem@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34658}
2021-08-06 10:51:58 +00:00
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
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
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
46d002cb36
Add M1 Mac expected results for AudioDecoderIsacFixTest
...
Bug: webrtc:12882
Change-Id: I56c1fcdd85fab88924b9a9f53a1a20485633f840
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223660
Commit-Queue: Christoffer Jansson <jansson@google.com >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#34389}
2021-06-30 07:03:52 +00:00
7208457e80
Same length for all ARM64 platforms
...
Update more audio checksums for M1
Bug: webrtc:12882
Change-Id: I527a43a01afe2b2e4af137852174159bf3111652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224081
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Commit-Queue: Christoffer Jansson <jansson@google.com >
Cr-Commit-Position: refs/heads/master@{#34372}
2021-06-28 11:18:40 +00:00
213dc2cfc5
Temporarily disable Opus decode test.
...
Bug: webrtc:12518, webrtc:12543
Change-Id: I5481ee96fe2a3f9fd549e17cd9424441223a8b63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211245
Reviewed-by: Ivo Creusen <ivoc@webrtc.org >
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#33417}
2021-03-10 12:47:18 +00:00
36274f9158
Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
...
This is a reland of 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6
Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> 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
d546186b89
Revert "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""
...
This reverts commit 1dbe30c7e895c7eb4da51c968a7a8897f25ad7e6.
Reason for revert: Speculative revert due to failing tests.
Original change's description:
> Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
>
> This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
>
> 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
1dbe30c7e8
Reland "Default enable WebRTC-SendSideBwe-WithOverhead."
...
This is a reland of 87c1950841c3f5e465e1663cc922717ce191e192
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
27af3c4c24
Revert "Default enable WebRTC-SendSideBwe-WithOverhead."
...
This reverts commit 87c1950841c3f5e465e1663cc922717ce191e192.
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
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
30a3e78794
iSAC encoder: Make it possible to change target bitrate at any time
...
Not just at construction time.
Bug: webrtc:11704
Change-Id: I952c7dbe20774cc976065c7d2f992a80074ebf63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177663
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#31550}
2020-06-22 14:59:22 +00:00
16ddae924e
Update Opus tests for Opus 1.3
...
This updates various bitexactness tests and other tests that no longer
pass.
Bug: webrtc:11325
Change-Id: Ifa3e4b42e303f5573e028dfdf8a108a76f6318ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168952
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30688}
2020-03-05 08:53:37 +00:00
b28e57e725
NetEQ audio decoder unit test: use ParsePayload
...
AudioDecoder::Decode() is obsolete. This CL replaces it with
ParsePayload() in the audio decoder NetEQ unit tests.
Bug: webrtc:10098
Change-Id: I602b0330adbe1d0921b0c4524aa7305b500f2ebf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168486
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30511}
2020-02-13 09:05:55 +00:00
c31a4ec66a
Disable opus tests to allow upgrade to opus 1.3
...
The upgrade to opus 1.3 is easier to carry out while the opus
bitexactness tests are temporarily disabled.
Bug: webrtc:11325
Change-Id: I96eecdbc93a01da88b92ae7f6473034c9795f3a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167726
Commit-Queue: Ivo Creusen <ivoc@webrtc.org >
Reviewed-by: Sam Zackrisson <saza@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#30425}
2020-01-30 14:57:15 +00:00
48b32b748e
Delete support for enabling adaptive isac mode
...
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.
Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-20 10:41:09 +00:00
608e6ba394
Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object
...
This CL will make AudioDecoderIsacT symmetrical to AudioEncoderIsacT.
Bug: webrtc:10826
Change-Id: I78d1cf7bc2245bf4a282aabd81c8ece6ca23f285
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146683
Commit-Queue: Jiawei Ou <ouj@fb.com >
Reviewed-by: Tommi <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28847}
2019-08-14 00:40:19 +00:00
7eb0a5e210
AudioDecoderOpus: Add support for 16 kHz output sample rate
...
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Minyue Li <minyue@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#28104}
2019-05-29 12:42:38 +00:00
10542f21c8
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
...
Mechanically generated by running this command:
tools_webrtc/do-renames.sh update all-renames.txt && git cl format
Then manually updating:
tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
682aac5103
Enable clang::find_bad_constructs for audio_coding (part 1/2).
...
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6a7d4964723a5e195189aac30a83d9e924e61dd7
Reviewed-on: https://webrtc-review.googlesource.com/89743
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#24053}
2018-07-20 13:07:47 +00:00
665174fdbb
Reformat the WebRTC code base
...
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
b602123a5a
Replace rtc::Optional with absl::optional in modules/audio_coding
...
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_coding'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org >
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 12:46:20 +00:00
12ab00b4d8
Optional: Use nullopt and implicit construction in /modules/audio_coding
...
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
R=kwiberg@webrtc.org
Bug: None
Change-Id: I055411a3e521964c81100869a197dd92f5608f1b
Reviewed-on: https://webrtc-review.googlesource.com/23619
Commit-Queue: Oskar Sundbom <ossu@webrtc.org >
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org >
Reviewed-by: Elad Alon <eladalon@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20728}
2017-11-17 11:58:37 +00:00
31fbb5425e
NetEq: Drop unnecessary dependency on the audio decoder implementations
...
BUG=webrtc:8396
Change-Id: I7524dae93b43b656a13fdd535e48373bc29b405e
Reviewed-on: https://webrtc-review.googlesource.com/10804
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org >
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#20310}
2017-10-16 12:57:47 +00:00
92ea95e34a
Fixing WebRTC after moving from src/webrtc to src/
...
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf
Moving src/webrtc into src/.
...
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org >
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
