This is a port of crrev.com/c/2936677.
Previously we only checked avx2 support and then use avx2/fma
intrinsics in SincResampler(crrev.com/c/2654647),this CL also
checks the fma support and avoids using avx2 code if fma is not
supported.
Bug: chromium:1410691
Change-Id: Ibf7c0a1bead87ebe5d3978cfd20cc23525169f40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291702
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39238}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This is a prerequisite step to break apart AudioCodingModule and AcmReceiver.
Bug: webrtc:14867
Change-Id: Iba589c7a31b6346ff4acb727793d84077162c8c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291534
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39235}
This ensures that adding features by SDP munging gets a review
by people who understand how this works in the community.
Bug: none
Change-Id: I36feb0e3c7896d4f7bec81078109d7914c349a0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291339
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39229}
For frames captured and sent to the callback immediately, we
are not sending the capturer ID as we used to do in base capturer
pipewire. Adding the capturer id as well as the frame capture time
so as to keep the sent frame to be in sync with the
non-immediate-frame-send implementation.
Bug: chromium:1291247
Change-Id: I02693907928b9e770ea56f89b46c37f17f4bc4a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291680
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39228}
Today, behaviour is decided based on if transport sequence number v2 is
in the SDP answer. But it might be better to decide based on received
packets since it is valid to negotiate both extensions.
Another bonus With this solution is that Call does not need to know
about receive header exensions.
This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337
Bug: webrtc:7135
Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39226}
Remove integration with socket server of the current thread
Network thread that uses PhysicalSocketServer shouldn't be allowed to do blocking calls
Other threads that use NullSocketServer do not need to process any messages while blocking
Bug: webrtc:14856
Change-Id: I56865b86e0992e60376ecefe163ff6b23911edca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291527
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39225}
which potentially allows switching to that pattern in the future.
Video FEC mechanisms (ulpfec, flexfec-03, RED) that currently
do not have any feedback parameters but will still be considered "common" and feedback may be sent for them.
For audio this causes rtcp-feedback to be sent for G711 and G722 if negotiated.
BUG=webrtc:14802
Change-Id: I54852d39e176f918d4b36462526ceb40617b8fbe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290702
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39224}
Earlier, we were relying on CRD to periodically do frame captures.
This is however not needed since Wayland captures are event based
and once the compositor has send the event/frame to us, we can just
handover the frame to a registered callback. This ensures that we
have a single frame clock that (i.e. one present in the compositor).
Without this change, there is a chance that CRD capture clock is run
out of sync with the compositor's clock and cause unnecessary frame
delays.
See the following ideal scenario, for example, where '+' indicates
when a frame is sent by the compositor and when CRD tries to capture
it. This assumes that the same clock cycle for both CRD and the
compositor, each cycle length is enclosed within | .... |).
Compositor Frame Ready | +... | | +... |
CRD Frame Capture | .+.. | | .+.. |
In this case, when both the clocks are in sync, CRD is able to
capture frame right after it is generated by the compositor. But if
they are completely out of sync, then CRD can always see a stale
frame (delayed by one cycle and it can cause users to feel stutter).
Compositor Frame Ready | .+.. | | .+.. |
CRD Frame Capture | +... | | +... |
This stutter can become worse if the compositor is delayed in
generating the frames for some reason (e.g. load on the system).
Bug: chromium:1291247
Change-Id: I7c10c724fbbd87dc523d474e7ece8e8aa146fd7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291080
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39218}
This change adds support for renegotiating the frame rate
via pipewire.
Bug: chromium:1291247
Change-Id: Iacd4a3c924900839a8db75a50b448df6c48c83ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291460
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39216}
Allow CSRCs to be modified per-frame in an Encoded Insertable Streams
transform, to support a web API which allows per-frame CSRC
modifications to signal when a JS application has changed the source
of the video which is written into an encoded frame.
Initially only for Video, with Audio support likely to follow later.
Bug: webrtc:14709
Change-Id: Ib34f35faa9cee56216b30eaae42d7e65c78bb9f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291324
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tove Petersson <tovep@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39214}
This removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused.
Some assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused.
Bug: webrtc:12397
Change-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39213}
This ensure BWE works as intended with transport sequence numbers on
audio.
Tested with webrtc_perf_tests --gtest_filter=CallPerfTest.Min_Bitrate_VideoAndAudio
and --gtest_filter=Rampup*
Bug: webrtc:14854, webrtc:7135, b/266786240
Change-Id: I3b7a743149c22035e582a2157b5f0a93747857cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291523
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39208}
This is to clear any remaining buffers and other state such as the next packet timestamp.
Bug: webrtc:12397
Change-Id: I2ef9a6f7254d82a69a2896ec8aa619ced2694fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291327
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39206}
This CL adds an unittest that a ChannelReceive can be constructed
and destroyed without crashing. It is a basis for further testing.
Lack of unit test was discovered while pursuing bug mentioned below.
Bug: webrtc:13931
Change-Id: Iddb110f2df25e3806c74a5d00bbfab6d6d8e267f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291338
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39200}
Instead, ensure extensions are registered so that both transport and send streams are aware.
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I7710113893e2c5e23c1365de6aa3b761e3408308
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291333
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39193}
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
