ModuleRtpRtcpImpl and ModuleRtpRtcpImpl2 share certain components, RtcpReceiver in particular.
To always have Environment in RtcpReceiver both legacy and new module need to propagate it.
No-Iwyu: suggests too many changes, better address them separately.
Bug: webrtc:362762208
Change-Id: I2c885f57e24f135229fb7cd9781126d663017b3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361142
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42908}
Delegate control over number of times to encoder using AV1E_SET_AUTO_TILES that was added in https://aomedia-review.googlesource.com/c/aom/+/191102.
Bug: webrtc:351644568
Change-Id: I87ed11734e907c7f6c6508ac7389c84ececf5b21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361140
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42903}
And checks similar fields in Configuration struct are not set.
Migrate rtp_rtcp to use new constructor.
Bug: webrtc:362762208
Change-Id: I2385439c169a7432d174c72ca57ecb0ca639d864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361100
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42896}
Based on the results of the experiment (b/335129329).
Bug: webrtc:15827, b/320629637, b/335129329, chromium:329396373
Change-Id: I1599f4c1be79ee3385aac1ff345168982c8278f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42895}
We can avoid using the global now that field trials from Environment are
used in NetEq. This allows running multiple instances in parallel with
different settings.
Bug: webrtc:42220378
Change-Id: Icff8539e3ae9b61c86bb393d9a313e786e032b93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359720
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42894}
This is needed in order to create corruptions (by altering the filter loop params) to test the corruption detection algorithm.
Bug: webrtc:358039777
Change-Id: Ib26e9c0187b79c13b9862898625742def4091b91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42890}
This replaces the payload type tracking in AcmReceiver with the one in
NetEq and should be a noop.
Bug: None
Change-Id: Iaf124b5e56a646f994b5c2af65d349ede550b7fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360840
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42875}
This fixes an edge case where the frame size changes for a DTX packet.
We should avoid having the frame size larger than the timestamp gap.
Bug: None
Change-Id: I0a384cfb06f5aebc1654c1e3d127541fd24e05c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360722
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42873}
This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.
Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
same as they attached to other packets.
Otherwise there is risk that ssrc will be acked after few initial pure padding packets are sent, before remote endpoint seen any mid or rid attached.
Bug: b/361257385
Change-Id: I695b379221debe2518ad33d13d65620877f0b2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42851}
This change refactors existing self-assignments within if clauses across
the WebRTC codebase.
*Why:*
- Bug Prevention: Assignments within conditionals are frequently
unintended errors, often mistaken for equality checks.
- Clearer Code: Separating assignments from conditionals improves code
readability and reduces the risk of misinterpretation.
Change-Id: I199dc26a35ceca109a2ac569b446811314dfdf0b
Bug: chromium:361594695
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360460
Reviewed-by: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42850}
The current solution does not work for GFD since GFD is only parsed from the first packet of the frame. As a result, to access the generic information, we have to check every packet when traversing the packet buffer to find the first packet of frame. This fix is necessary to ensure temporal scaling works correctly with GFD.
Bug: webrtc:42225186
Change-Id: Iadda4ec690deab62c32eb6101583e6a6d75cfeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344840
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42836}
by encryption a packet with sequence number 65535 followed
by a packet with sequence number 1. The second packet is encrypted
with a SRTP ROC of 1 as described in
https://datatracker.ietf.org/doc/html/rfc3711#section-3.3.1
The packets are (received and) decrypted in a different order,
the packet with sequence number 1 (and ROC=1) is decrypted first.
Since the ROC is maintained locally the decrypting session assumes
it to be 0.
Why is that a problem? The RFC recommends estimating the ROC with +-1 which, as demonstrated by the test, libSRTP does not.
But this is a rare problem that requires a random in a high range combined with packet loss/reordering which turns into no-a-problem if you choose carefully as done by packet_sequencer.cc which restricts the initial sequence number in the range 0..32767 which means you do not run into this issue in production.
See also Q6 in libsrtp's historical documentation at
https://srtp.sourceforge.net/historical/faq.html
BUG=webrtc:353565743
Change-Id: I9bd72b198c946937aeb25c229005a0c682447f53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42798}
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}
Current version of the dav1d decoder does not propagate any QP value to the Decoded callback. This CL updates this such that the base QP gets propagated from the frame header.
Bug: None
Change-Id: Ib7624b7e27d2c973f1821df5688cbb444e4847a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359740
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42790}
This can happen when VP8 simulcast is negotiated while two-byte header
extensions are not negotiated via extmap-allow-mixed. For VP8 the
DD extension would be 23 bytes long which exceeds the maximum size
of 15 bytes for a one-byte header extension.
To test, revert
f04b52b4a7
and test using VP8.
Note that this works for VP9, AV1, H264 out of the box.
BUG=webrtc:40191093
Change-Id: I2f5d04d8b58b71d32547b06fab6b9a9006df9f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359623
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42786}
Filter out devices that do not support any format supported by WebRTC.
This will for example be IR cameras that show as duplicated in the list
of cameras, but support only GRAY8 format and for that reason do not
work at all.
Bug: webrtc:42225999
Change-Id: Ic2905bc66b55c3f48b49ff4097167f10d17ad656
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358864
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42785}
Finalize change started in https://webrtc-review.googlesource.com/c/src/+/359243
Remove fallback to old interface and unneeded clock member in the config struct.
Bug: None
Change-Id: I4c2b65a09dd1c8a0d44ee76320b095516e2c61fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359561
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42782}
e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
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@{#42773}
If all packets are dropped for a period of time, an observation window will have the same length as the period when packets are dropped.
If later, no packets are lost, there is no point in loss based bwe backing down.
Therefore, ignore the observation with most loss and least loss when calculating an instant upper bound.
Bug: webrtc:42222865
Change-Id: I1d0125d6c76e68018b2aec1ecaa9b65729963136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/356380
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@google.com>
Cr-Commit-Position: refs/heads/main@{#42772}
to make it available for creating AudioDecoders
Bug: webrtc:356878416
Change-Id: Ibd24a55df70985dfe02d924da037618f13661032
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359241
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42771}
To propagate field trials in addition to clock
Bug: webrtc:356878416
Change-Id: Idefc4848ec4af30c8aed0f93b7fadfc3181bddb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358980
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42761}
To make it available for FEC to use field trials in follow ups
Bug: webrtc:355577231
Change-Id: I4a6260a38e50a70dae27db28401b08bf0160aaec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358680
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42740}
To propagate field trials into the NetEq and further towards Audio Decoders
Bug: webrtc:356878416
Change-Id: Ia7cf18451aef70441ca958bf652f492138c6051a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358620
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42739}
To make it available for FEC to use field trials in follow ups
Bug: webrtc:355577231
Change-Id: Ie0b7761915696e6ee7453df3d0531b0f7ad30ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358240
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42732}
Passing Environment would allow to propage field trials with it further to NetEq and AudioDecoders
Bug: webrtc:356878416
Change-Id: Ic68420df3b157ed341146207a2c45cb49e59a931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358501
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42728}
Similar to PipeWire implementation of desktop capture, we have to avoid
CFI check for calls of dlopened PipeWire library. This avoid crashing
PipeWire camera backend when "is_official_build=true" option is used as
this turns on "is_cfi=true" enabling control flow integrity.
Bug: chromium:354776214
Change-Id: I7a9fc1c2d77c4ee0e8fe0586369b7246e0bb9180
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358103
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42706}
