- Add `WebRTC-Audio-OpusDecodeStereoByDefault` field trial
- Behind that field trial, `AudioDecoderOpus::SdpToConfig` uses 2
instead of 1 as default number of channels when the `stereo` codec
param is unspecified
- Instead of wiring up `FieldTrialsView` to `SdpToConfig`, which
requires API changes that break downstream projects, a change in
`AudioDecoderOpus::Config` is made to signal when the number of
channels is forced via SDP config
Bug: webrtc:379996136
Change-Id: If70eb19bc7e3bc74dd0423610cb04ae33ea602fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368860
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43440}
Mark old overload deprecated.
This allows to migrate both calls through AudioDecoderFactory and direct calls to AudioDecpderOpus trait.
Bug: webrtc:356878416
Change-Id: I1502aee5b18aac43a8258e77b770c8e73a056f92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359741
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42793}
Plan is to pass field trials using Environment when creating individual AudioDecoder rather than providing single set of field trials for the factory.
Current implementation is not used, and doesn't pass field trials when actually creating an AudioDecoder
Bug: webrtc:356878416
Change-Id: I0f79f09f7a6aa63e20fbdd783e90e8d026158330
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359221
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42762}
Field trials are passed during AudioEncoder construction through Environment parameter
All known users were migrated to the same named function without parameters.
Bug: webrtc:343086059
Change-Id: I79e2edae22ab43f98a386430da82b41d1c71e426
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358061
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42693}
Make AudioEncoderFactory::Create pure virtual.
To finalize migrating AudioEncoderFactory to new interface for creating AudioEncoder and thus guarantee AudioEncoders always have an Environment at construction.
Bug: webrtc:343086059
Change-Id: I1d607082437c15201c8a75dd7a3925fe0f75b70f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355800
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42686}
The template never pass that parameter to create an audio encoder, and thus is de-facto always unused.
Plan is to propagate field trials differently, during construction of the encoder rather than during construction of the encoder factory.
Bug: webrtc:343086059
Change-Id: I96d49ef7f7ff37524a1eb64f7a2d73881c07e00a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354360
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42527}
First batch of applying iwyu to the repo.
Done with:
> ./tools_webrtc/iwyu/apply-iwyu api
> git add api/[a-s]*
> python3 gn_autodeps.py ~/local/webrtc/src out/Default
Last step is a custom script I wrote to automatically apply new required
dependencies for target in gn, which saved tons of time manually going
over the files and fixing.
If this is something that interest others, I can submit it as well.
Bug: webrtc:42226242
Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42512}
Update AudioEncoderFactoryTemplate implementation to expand unit tests to the new api.
Bug: webrtc:343086059
Change-Id: Ib63640de38aa15cc36067d5a3d1de2bf42cec313
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42500}
and move usages to webrtc::RefCountInterface
This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.
Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
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}
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}
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}
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}
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}
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}
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}
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}
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}
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}
