Compiling webrtc with `-Werror=unused-parameters` is failling duo to
those parameters.
Also, it shouldn't harm us to put those in comment for code readability as
well.
Bug: webrtc:370878648
Change-Id: I0ab2eafd26e46312e4595f302b92006c9e23d5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364340
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43157}
Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
This is a reland of commit 4334cdfc5c0619a5f06125ea1f039cb123ccf21e
Original change's description:
> Reland "Return audio stats regarless if we have a codec."
>
> This is a reland of commit 7fff587a096c6ef40f5601f47ef50b221b3a4abf
>
> Original change's description:
> > Return audio stats regarless if we have a codec.
> >
> > Bug: b/331602608
> > Change-Id: I2d12a3ed83645fe1e7cbd8950fd86d5ba2d7c94d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361743
> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> > Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42964}
>
> Bug: b/331602608
> Change-Id: I95c89e7059005bc8dd8569ef41bfe9e863b4082f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361762
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42969}
Bug: b/331602608
Change-Id: I743f0d623230bf871de262792981de35c156ba3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364461
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43142}
This has been deprecated since November 2022.
Bug: None
Change-Id: Ia547489b1f703d0744ab7ffc096eeadbb937974a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364381
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43140}
A slight behavior change is that we only increment total samples received when GetAudio is successful.
Bug: webrtc:370424996
Change-Id: I8607418c179ca3bc22963b98792a9e8b9af2d451
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364220
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43139}
When AdaptFrameResolution() applies the requested resolution as a
restriction (max width and max height) it does so on the "input" size
rather than on the "output" size. While this results in the correct
output size anyway, it also produces cropping which results in the image
looking zoomed in (see https://crbug.com/webrtc/369865055 for repro).
To fix this issue the restrict logic is moved and applied on the
"output" instead. The logic is updated to take alignment into account
since the resulting size is the final output.
Bug: webrtc:369865055
Change-Id: I2d5476929432c45173a57c0f4964ab9a38518189
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364163
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43138}
This reverts commit 37458ce40a1741f9d5358d49fe49beed20140502.
Reason for revert: Will be wired up as an experiment instead.
Original change's description:
> Per defaul probe max to 2x current BWE if max total allocated bitrate change
>
> This aligns to probe limits in ALR for example.
>
> Bug: webrtc:369044000, b/369021234
> Change-Id: I3823b308cf97a8b7060b35b2ac38864e75d6f983
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363301
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43074}
Bug: webrtc:369044000, b/369021234
Change-Id: I22b457254c9c21d2d951af2bda01a349ef83b3c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364242
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ranveer Aggarwal <ranvr@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43130}
which has been deprecated since 2022 as shown by
git grep -n "\[\[deprecated" | while IFS=: read i j k; do git blame -L $j,$j $i -n -f | cat; done
BUG=webrtc:42224819
Change-Id: If7c5cc97aabfb43693ea3b07d45c3aa5ecc7236a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364181
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43129}
A number of unit tests assume that payload types will be assigned
without generating an offer. These are flushed out by running tests
with the --force_fieldtrials=WebRTC-PayloadTypesInTransport argument.
Bug: webrtc:360058654
Change-Id: I17cd5bfa275904a9630068190b1cd246e9ce8741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362500
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43127}
To force SvcRateAllocator use propagated rather than global field trials
Bug: webrtc:42220378
Change-Id: I0ca3186ee2428aafe3d7f093603b708e03ada121
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362722
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43123}
These have been deprecated since 2022.
Bug: None
Change-Id: I8340750f67e57c37601754345c679062c3c23436
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364283
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43122}
This reverts commit f8b3dab7c6320a9890f0b003b43d7099e2e00a5b.
Reason for revert: The fix landed in libaom (https://aomedia-review.googlesource.com/c/aom/+/193761) and it is now available in WebRTC (import CL: https://webrtc-review.googlesource.com/c/src/+/364126).
Original change's description:
> Disable LibaomAv1Encoder tests to unblock Chromium roll
>
> The tests exercise the new encoder API that is not used in prod yet.
>
> Bug: webrtc:369633254
> Change-Id: Iee6bc16ebd471f4accdd9531cdb404f159557f51
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363820
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43083}
Bug: webrtc:369633254
Change-Id: Ia02db32f7f09e3abc3d0a46605feeabd82673f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364281
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43120}
These functions have been deprecated since October 2022.
Bug: None
Change-Id: I74f51c9d0e8ee340a2043bf43f7a1b0d8b79726e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43118}
This reverts commit bdc669347c70160cd648f5cab7a417227d41d82a.
Reason for revert: AUDs will be taken into account now.
video_replay with the provided out.pcap and these options:
--codec H264 --input_file out.pcap --media_payload_type 102 --ssrc 40000
plays smoothly.
Original change's description:
> Revert "h264: fix first_packet_in_frame logic for multislice in a single rtp packet"
>
> This reverts commit 3753c8190e3f0aca6758a5521e33f8b5d4f09ab4.
>
> Reason for revert: Break assembling of hardware encoded h264 P frame on
> weak network condition.
>
> Original change's description:
> > h264: fix first_packet_in_frame logic for multislice in a single rtp packet
> >
> > a frame must be (or should be) first when it contains either SPS (but not just PPS),
> > is an IDR or is a slice with first_mb_in_slice == 0.
> >
> > Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
> > into a single RTP packet which can happen with small 320x196 frames
> >
> > BUG=webrtc:352379280,webrtc:346608838
> >
> > Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42652}
>
> Bug: webrtc:368335257
> Change-Id: I07725c78be628bff71b79b8b9369677e39f5f5ac
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363080
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43062}
Bug: webrtc:368335257
Change-Id: Idfae2efc1ebd7b97a2f7ebbd9d1e8c7bf6fcc348
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363842
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43113}
since 1024 is already deprecated by OpenSSL and causes "too small key"
issues on systems enforcing a minimum size. Similar issue here:
https://github.com/nodejs/node/pull/44498
The minimum key size is not yet changed from 1024, this will require more effort for deprecation.
BUG=webrtc:364338811
Change-Id: Id4b24a2c289ec5e3f112288d32b8ac697ba1cfed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361128
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43110}
