Whenever encoding info change, this ToString() method is called for some
LS_INFO logging inside video_stream_encoder.cc. Apparently the char
buffer used for constructing this string is not large enough because I
can get WebRTC to crash in a demo page that gets and sets a lot of
parameters.
By changing to rtc::StringBuilder, we don't have to make assumptions
about how long the string can get at runtime.
Bug: None
Change-Id: I32695523282143a301c0e13e06082d55bd2796b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375520
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43805}
At Meta, we have formatted the repo according to WebRTC .clang-format
file. Currently, those changes are stored as patch and we'd like to
apply them to the base WebRTC release instead.
I will be submitting CLs per folder. The plan is to format all h|cc|mm|m
files, while exlcuding Matlab files from the formatter as clang
misinterprets them as ObjC.
Formatting done via:
git ls-files | grep -E '^api\/.*\.(h|cc)' | xargs clang-format -i
No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: I4d7470104983d5d32612f9347301354265fb34c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373520
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43671}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
The std::lcm and std::gcd functions are part of the C++ standard
library. The existing functions are marked as deprecated rather than
deleted in the case of possible third party uses.
#rtc_cleanup
Bug: webrtc:377205743
Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43368}
The const-ref result of .str() must be copied into the returned
value, whereas the result of .Release() can be moved.
Bug: webrtc:374845009
Change-Id: I3abc98be30ce9947127c7664f5ffa6846b772ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43288}
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.
NOTE: This time I made sure to iterate over the C files in the
audio_processing folder and compile them using gcc.
On the original CL that was reverted - that failed with the same error
Danil mentioned. This time it seems fine.
I'll make sure to run the same script on the rest of my CLs for sanity
Bug: webrtc:370878648
Change-Id: I83cea3a08777e21d26a95bcad503a2d1b74566eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364537
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43249}
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}
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}
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}
Level asymmetry is implicitly enabled for HEVC. When comparing two
codec params to see if they match, we only compare profile & tier,
similar as H.264.
Bug: chromium:41480904
Change-Id: I9e9debdf1b34f33986da9344b9fee14071b1ed60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43069}
This will help to reduce redundant ScalabilityMode to temporal layer
count mapping in blink.
Bug: chromium:40763991
Change-Id: Ida3e6abb91383e27465eb1b697ad9431935cf9ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362486
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43031}
Now some HW encoders support simulcast. If parameters are not suitable for
single encoder simulcast, the error code should be forwarded back to
SimulcastEncoderAdapter instead of trying software fallback.
Bug: webrtc:347737882
Change-Id: Id02ff1afc012cd46761d9530b1ce368d5dc480bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42972}
For better consistency with the rest codebase (it is min_/max_ for all params in video_encoder.h; only qp is for some reason prefixed with minimum_).
Also fixed constant names in libaom AV1 encoder wrapper (moved min from suffix to prefix, minimum -> min_).
Bug: chromium:328598314
Change-Id: I6d8521a3abff3a0595a5241c02ef4746eb4694df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356600
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42604}
The minimum QP field will be used to signal what the QP value will be
once the encoder reach its target video quality. This will be used
in the generalized QP convergence detection.
Bug: chromium:328598314
Change-Id: I82299cd921e3c091e651218d1e3f337875176567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Cr-Commit-Position: refs/heads/main@{#42559}
So that this class can use propagated field trials instead of the global
Bug: webrtc:42220378
Change-Id: Ic1dba0c4967735606904329f7e9e6c09f186b809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42326}
Also initial implementation wrapping the libaom AV1 encoder.
Note that for now this is intended for prototype purposes.
Bug: none
Change-Id: Iac42ca4aecb6a204601c9f00bfb300e3eda3c4f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306181
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42108}
This expose a new GetSupportedH265Level API for WebRTC external
factories to calculate H.265 levels to be use for SDP negotation.
Bug: webrtc:13485
Change-Id: Ib420da2b9b1b7af00129294be5b3efec172e8faf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/345544
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42079}
The predefined SdpVideoFormats were not used everywhere,
which caused a discrepancy between send/receive capabilities
for AV1. This CL solves the immediate problems by making sure
send/receive capabilities for AV1 are reported the same way.
Fixed: chromium:331565934
Change-Id: I073091b7b5f987c7f434c17276fd84047ec723c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344681
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41991}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
