These function were replaced with AbslStringify
Bug: None
Change-Id: Ia34b98ed4e0ed18bb52fe9370cff7a6f70caae6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364621
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43346}
- WebRTC does use the libopus DTX implementation
- The removed detail is anyways irrelevant in a docstring
Bug: webrtc:376493209
Change-Id: I3dfe1521259e596dbfa0db97f91ffb75deeb16b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367200
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43344}
With L4S in WebRTC, only RTP packets are supposed to be send with ECT(1)
Bug: webrtc:42225697
Change-Id: If10bf74a867d3ea04fd1fb931cdc2a6380176270
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43343}
New api ensures field trials are available at construction time of the AudioProcessing object.
This would allow AudioProcessing implementation to use propagated field trials during construction.
Also, short term, it ensures AudioProcessing is constructed after global field trials are set.
Bug: webrtc:369904700
Change-Id: If3d00c8a3a509299cd0915d55f13a9a3ce4a7140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367201
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43340}
and modify DEPS files accordingly.
This is done in support of the decision to encourage AbslStringify.
Bug: None
Change-Id: I26fee77978d1dd21be6d2ef011c4dfd78a7b43e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367204
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43338}
Increased the number of errors the automation is fixing to 150 from
75 in this commit.
Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43337}
- Avoid redundant get() when dereferencing smartpointers
- Use const ref instead of copy for RtpExtension
- Use `.empty()` instead of `.size() == 0`
- Remove some unused using declarations
Bug: None
Change-Id: I0dfdc0dfdf165f153c9ba119c115cd492e9599fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43334}
These functions seem to have been unused except for tests.
It seems to have been added in 2017.
Bug: None
Change-Id: I01983f4b72456b1df27ec2d346014e0de1b5cae7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366943
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43332}
To stress there is no intention to use each instance more than once.
Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
and remove the testing of the nondefault key size from the "server" parameters to speed up tests
BUG=webrtc:375552698
Change-Id: Ibc1bd491300964aa45826b98962ed3e56c6d4974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43321}
It is used to distinguish between audio/video packets and everything else (retransmit/padding/fec), so naming it is_media makes more sense.
This is a follow up to https://webrtc-review.googlesource.com/366644
Bug: b/375148360
Change-Id: Ia53f4d707ceb85f059688d86bc5dcc2d57908d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366424
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43319}
After landing this change, we can change the corresponding usage in
blink to start using presentation_timestamp as well and then delete
the remaining usage of capture_time_identifier.
Bug: webrtc:373365537
Change-Id: I0c4f2b6b3822df42d6e3387df2c243c3684d8a41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#43317}
Now that Chromium and internal_compile_lite have migrated we can delete
the old name.
Bug: webrtc:375048799
Change-Id: I11d79f1d4ef1c0aa132cb50856faf83250e07caf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366600
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43312}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
This function will be used by RTCVideoEncoderFactory for converting
media::SVCScalabilityMode to webrtc::ScalabilityMode.
Bug: webrtc:41480904
Change-Id: I64d7696457705851657050d2ea2b8c1d845286e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366397
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43308}
Removing AudioProcessingBuilder from few layers would simplify replacing with BuiltinAudioProcessingFactory in the upcoming patches.
While doing cleanup also removed extra always empty parameters and run iwyu.
Bug: webrtc:369904700
Change-Id: I54d44993701c30ca8f4cf38e822af08531fba310
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366260
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43306}
This is a reland of commit 82617ac51e7825db53451818f4d1ad52b69761fd
The reason for the revert was a downstream use of
`rtc::VideoSinkWants::requested_resolution`, so in this reland we don't
rename this field, it's fine just to rename the one in
RtpEncodingParameters for now.
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
NOTRY=True
Bug: webrtc:375048799
Change-Id: Ic4ee156c1d50aa36070a8d84059870791dcbbe5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366660
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43304}
When switching between payload types on same ssrc, a HW decoder is only
used the first payload type received, falling back to SW decoding if
payload type is changed.
This change unregister any external decoder previously registered so it
can be re-initialized if received again.
Bug: webrtc:375097852
Change-Id: Ic04951c5676d9a3854eefb2ab8836ef8a2645d78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366580
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43302}
This reverts commit 82617ac51e7825db53451818f4d1ad52b69761fd.
Reason for revert: Break downstream projects
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
Bug: webrtc:375048799
Change-Id: Ie41723a39420e12e7b5b681d3d00ccd14f66b4b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43301}
This is a pure refactor/rename CL without any changes in behavior.
This field is called scaleResolutionDownTo in the spec and JavaScript.
Let's make C++ match to avoid confusion.
In order not to break downstream during the transition a variable with
the old name being a pure reference to the renamed attribute is added.
This means we have to add custom constructors, but we can change this
back to "= default" when the transition is completed, which should only
be a couple of CLs away.
Bug: webrtc:375048799
Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43300}
Prior to this change, we unnecessarily re-compute AverageReportedPacketLossRatio a few times for the same observation.
Bug: webrtc:12707
Change-Id: I5de3d06bf3a32a39638a6f72baf7ec12144c6399
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366583
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43298}
