Bug: webrtc:15719
Change-Id: I7daf8ee5b90fbe9f1246f1d99211ffa0d8a19f73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330780
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41503}
rtc::TaskQueue is a wrapper of TaskQueueBase providing no extra functionality in this case
Bug: webrtc:14169
Change-Id: I5eb27a5dbb16f6097a9c71c2633c807808e50c05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333800
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41501}
Ensure top_active_prio_level_ is set to -1 in MaybeUpdateTopPrioLevel if
last packet is purged.
Bug: webrtc:15740
Change-Id: I81df9ee084de89f79b8ab79db8ce52fe1e20738a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333883
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41498}
This field trial was added 5 years ago in
https://webrtc-review.googlesource.com/c/src/+/111883
probably as a safe guard, but looks never used.
Bug: webrtc:11503
Change-Id: Ia9544b652b25fad4c614d66fe020f3d994c96505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333380
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41490}
This is a reland of commit 63d03f586bb668f72113b61030ec0930aa192010
Original change's description:
> Unify access to SDP codec parameters
>
> which come from the a=fmtp:<pt> lines in the SDP and were used as either
> std::map<std::string, std:string>
> with three aliases,
> cricket::CodecParameterMap
> SdpAudioFormat::Parameters
> SdpVideoFormat::Parameters
>
> Use webrtc::CodecParameterMap in all places.
>
> BUG=None
>
> Change-Id: If47692bde7347834c349c6539b43309d8770e67b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41375}
Bug: None
Change-Id: I5f8f45688df232eb37b12fa3e56a893a1c754e17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41467}
Adds separate priorities for audio and video retranmission.
Done by adding an original type to RtpPacketToSend.
Add possiblity to set TTL for audio nack, video nack and video packet separately.
Oldest packet for these types are dropped when a new packet of that type is pushed to the pacer, or when the pacer switch current priority type to that priority.
Effect is that:
-pacer queue does not grow unlimited for these types if a TTL has been set.
-an old packet is not sent.
Bug: webrtc:15740
Change-Id: I38718bc570aebca54eacbded69824905f3694f41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331823
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41414}
Fix the unintended disabling of RTP retransmissions for cloned encoded
frames, caused by passing an infinite "expected_retransmission_time".
Instead use a constant 10ms for now. For frames encoded locally, this is
set from an estimate of the RTT, but we currently don't have access to
that here (TODO added to pipe it through)
If an integration is cloning and then sending frames it received, it's
almost certainly resending received media to other peers on a local
network, so 10ms is a fair upperbound.
Tested locally with Chrome on Mac, configuring packet drops & observing
on chrome://webrtc-internals that retransmission packets are now sent.
Bug: chromium:1512631
Change-Id: I2483415dc7e0079f8a7b66f6607f4907698514c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41405}
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
After https://webrtc-review.googlesource.com/c/src/+/329141, best candidate can still be less than acked rate if not_increase_if_inherent_loss_less_than_average_loss, or the selected candidate is 95% of current estimate. This cl/ is ensure the previous cl works as intended. And add unit test.
Bug: webrtc:12707
Change-Id: Ie5683ca8ea51f6d80c4c59cbf08c22e8b24c0cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329441
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41298}
Move logging of decode failure from VCMGenericDecoder to VideoReceiveStream2 where remote SSRC is always known. Log frame details such as size and resolution which help to identify this frame in bitstream dump.
Bug: b/309132190
Change-Id: Ibe50799e448ffdc19f9857cc1625cfde0d7aa7a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328821
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41276}
Move some logic from PacketBuffer to NetEqImpl.
Bug: webrtc:13322
Change-Id: I88b1e55c0cd69700730d9ed41be04fcf1effa03f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328861
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41270}
These flags were never experimented or launched.
Bug: webrtc:12707
Change-Id: Iefedeade52fdcf7f978894c4bf837261810f41bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329080
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41265}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
This means that RtpPacketHistory::PaddingMode::kRecentLargePacket is
used per default.
Bug: webrtc:15201, b/284281602
Change-Id: If8feb66105a9b1e13ae4cb28a44a74c8839b72e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327602
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41215}
There are two threads involved here, the thread that calls the API
functions and the pipwire main loop. Using one race checker for both is
wrong and triggers aborts.
Use a different race checker for all variables that are used by the
pipewire main loop or guarded against concurrent access with the
thread_loop_lock.
In one case, two RTC_CHECK_RUNS_SERIALIZED() checks are needed, so
enhance the macro to generate unique variable names.
Bug: webrtc:15181
Change-Id: Ib41514eb7aa98fe85d830461aa0c71e42ba821bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326781
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41198}
This ensure upper link capacity estimate upper limit an increase in
delay based estimate, but the delay based estimate is not decreased if
link capacity estimate decrease.
Bug: webrtc:10498, b/300868877
Change-Id: I87e76e2a869e6f721cc8fe9d422e0194371d4e45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327801
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41196}
This is a reland of commit 496893e89e5bc8139e50befcb1a26eadbd829b0d
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Iccb9af8bf6a6c37704bc58b6e57238b55761b079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327781
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41194}
Add a StartShortCircuiting() callback to allow clients which have
configured Encoded Transforms when creating a PeerConnection to have
all frames skip the transform. This offers a zero cost path for streams
which don't need transforms.
This is preferable to uninstalling/not installing the transform to allow
implementing the behaviour in
https://w3c.github.io/webrtc-encoded-transform/#stream-creation -
giving web apps a chance to configure transforms within a short window
(before the next JS event loop run, so usually sub-millisecond) after stream creation, without any untransformed frames passing.
Usage in Chromium: crrev.com/c/5040731
Bug: chromium:1502781
Change-Id: I803477db1df51e80bdedf6c84d2d3695b088de83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41184}
This reverts commit 496893e89e5bc8139e50befcb1a26eadbd829b0d.
Reason for revert: Breaks https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/16103/overview
Original change's description:
> Added an encode/decode test parameterizable via command line
>
> This enables testing different settings without updating code and rebuilding the test binary. Example of command:
>
> video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
>
> Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
>
> Bug: webrtc:14852
> Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41179}
Bug: webrtc:14852
Change-Id: Ifdce738058c3e3fc7c76886939add2813405cae7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327722
Owners-Override: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41183}
This enables testing different settings without updating code and rebuilding the test binary. Example of command:
video_codec_perf_tests --gtest_also_run_disabled_tests --gtest_filter=*EncodeDecode --encoder=libaom-av1 --decoder=dav1d --scalability_mode=L1T3 --bitrate_kbps=100,200,300 --framerate_fps=30 --write_csv
Also added writing per-frame stats to a CSV. It is more convenient to work with CSV than to parse metrics proto.
Bug: webrtc:14852
Change-Id: I1b3970f7ffa88c016133197aff585de5bc4e35c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327600
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41179}
The updated Flexfec RFC states that a kbit of "0" means this is the last block of the mask, whereas in the 03 draft, "0" meant there's another block.
Reversing the logic in the updated RFC parser to fix.
Bug: webrtc:15002
Change-Id: I40e4c950b09ddf2db9da6c01908737282161bf1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327580
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41174}
Make sure the notifier is reset when tearing down the camera portal and also when we already called it. Destruction of camera portal will be mostly invoked by an object holding it and serving as an implementation of the notifier interface and in such case we have to make sure it will
not get called at this moment.
Bug: webrtc:15407
Change-Id: If0c1fb1493d64d5e1f0228ed71813abbb9280083
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315420
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#41167}
Moving the header file and definitions for PipeWireSession to the source
file allows DeviceInfoPipeWire to be reimplemented or used in wrappers
without the consumer needing to add PipeWire includes and definitions.
Bug: webrtc:15654
Change-Id: I895059d50bdf9e6ed152eca729c618261701457a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327381
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#41163}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
- Stable delay mode: this results in a very large reduction in the amount of time stretching and fewer underruns.
- More closely align PLC and CNG logic.
- Stop playing comfort noise after a timeout when no packets are received.
Several tests needed to be updated to match the new behavior.
Note that I should also refactor GetDecision to be easier to test in the future (remove internal state).
Bug: webrtc:13322
Change-Id: I1724a74b3b583d05a4bb8feb4f9a8c4a8b2b7c5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326780
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41125}
std::is_pod is deprecated since C++20. Replace with std::trivial and
std::is_standard_layout. Avoids a lot of warnings.
Bug: chromium:957519
Change-Id: Idb4bde7401c14c0896a84c357ec668b9916f613e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325484
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41117}
It did not result in big quality improvements.
Bug: webrtc:12201
Change-Id: I9728469a388ee179d6069af8521bfc5571870bd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325533
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41087}
This CL is a follow-up of work done in
https://webrtc-review.googlesource.com/c/src/+/323882 where the goal
was to reduce the amount of FrameDropped error logs in
WebRTC.DesktopCapture.Win.WgcCaptureSessionGetFrameResult.
The previous work avoids FrameDropped logs for a minimized window
being captured with WGC but we still se a large amount of these error
(or rather warning) logs. See [1] which comes from Canary.
This CL does two different things to improve the situation:
1) It adds kFramePoolEmpty to the existing
GetFrameResult::kFrameDropped enum to point out that the warning
comes from the frame pool not being able to return a valid new frame.
It also makes it more clear that it does not cause an outer/final
error as WgcCapturerResult::kFrameDropped. We still keep the inner
GetFrameResult::kFrameDropped but it is only produced when the frame
pool returns NULL and our external queue is empty. Hence, a real
frame-drop error. Note that, it is still easy to provoke
kFramePoolEmpty simply by asking for a high resolution at a high rate.
The example in [2] comes from a 4K screen @30fps. Hence, we have not
fixed anything yet.
2) It also increases the size of the internal frame pool from 1 to 2.
This does lead to an almost zero rate of kFramePoolEmpt
warnings at the expense of a slightly reduced max capture rate. BUT,
with 1 as size, we can "see" a higher max capture rate but it is not
a true rate since it comes with a high rate of kFramePoolEmpty
errors. Hence, we "emulate" a high capture rate by simply feeding
copies of the last frame that we had stored in the external queue.
Using 2 leads to a more "true" rate of what we actually can capture
in terms of *new* frames and also a substantially lower rate of
kFramePoolEmpty.
In addition, with 1 as size, if we ask at a too high rate and provide
a copy of the last frame, our CPU adaptation will not reduce its rate
since we think that things are OK when it is actually not.
Also, the samples in [3] and [4] both use 2 as numberOfBuffers
as well.
Let me also mention that with this small change, I a have not been
able to provoke any kFramePoolEmpty error messages.
Finally, geDisplayMedia can be called called with constraints where
min and max framerate is defined. The mechanism which maintains the
min rate is implemented via the RequestRefreshFrame API and it can
be sent to the source (DesktopCaptureDevice) back to back with a
previous timer interrupt for a capture request. Without this change,
these RRFs were also a source of a large amount of
kFramePoolEmpty error logs. With 2 buffers instead; this is no
longer the case.
[1] https://screenshot.googleplex.com/7sfv6HdGXLwyxdj
[2] https://paste.googleplex.com/4795680001359872
[3] https://github.com/robmikh/Win32CaptureSample/blob/master/Win32CaptureSample/SimpleCapture.cpp
[4] https://learn.microsoft.com/en-us/windows/uwp/audio-video-camera/screen-capture#add-the-screen-capture-capability
Bug: chromium:1314868
Change-Id: I73b823b31a993fd2cd6e007b212826dfe1a80012
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325521
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41079}
