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}
To allow delete old signature of the AudioEncoderOpus::MakeAudioEncoder function and thus guarantee Opus AudioEncoder always has an Environment
Bug: webrtc:343086059
Change-Id: Ib660678aeb5a549dddd1dffa3d8c28b2ec6b9d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356981
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42690}
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}
Low-Coverage-Reason: EXPERIMENTAL_CODE Code is behind field trial that will only be used for testing.
Bug: webrtc:13322
Change-Id: Ie306be808381b3a20b4e0d58349927bf3524018a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335840
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41608}
- 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
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}
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.
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}
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}
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}
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}
