This is a reland of commit 050ffefd854f8a57071992238723259e9ae0d85a
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
NOTRY=true
Bug: b/322132132
Change-Id: I25dd34b9ba59ea8502e47b4c89cd111430636e08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41736}
This reverts commit 050ffefd854f8a57071992238723259e9ae0d85a.
Reason for revert: Breaks downstream project.
Original change's description:
> Extends WebRTC logs for software encoder fallback
>
> This CL extends logging related to HW->SW fallbacks on the encoder
> side in WebRTC. The goal is to make it easier to track down the
> different steps taken when setting up the video encoder and why/when
> HW encoding fails.
>
> Current logs are added on several lines which makes regexp searching
> difficult. This CL adds all related information on one line instead.
>
> Three new search tags are also added VSE (VideoStreamEncoder), VESFW
> (VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
>
> It has been verified that these added logs also show up in WebRTC
> logs in Meet.
>
> Logs from the GPU process are not included due to the sandboxed
> nature which makes it much more complex to add to the native
> WebRTC log. I think that these simple logs will provide value as is.
>
> Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
>
> Bug: b/322132132
> Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41733}
Bug: b/322132132
Change-Id: I24d0a4e71a43ac192485f1af208563a51d919865
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41735}
This CL extends logging related to HW->SW fallbacks on the encoder
side in WebRTC. The goal is to make it easier to track down the
different steps taken when setting up the video encoder and why/when
HW encoding fails.
Current logs are added on several lines which makes regexp searching
difficult. This CL adds all related information on one line instead.
Three new search tags are also added VSE (VideoStreamEncoder), VESFW
(VideoEncoderSoftwareFallbackWrapper) and SEA (SimulcastEncoderAdapter). The idea is to allow searching for the tags to see correlated logs.
It has been verified that these added logs also show up in WebRTC
logs in Meet.
Logs from the GPU process are not included due to the sandboxed
nature which makes it much more complex to add to the native
WebRTC log. I think that these simple logs will provide value as is.
Example: https://gist.github.com/henrik-and/41946f7f0b10774241bd14d7687f770b
Bug: b/322132132
Change-Id: Iec58c9741a9dd6bab3236a88e9a6e45440f5d980
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41733}
This approach actually wraps the unique identifier generation into the
function that provides the output path for a test.
This way we don't need to add `CreateRandomUuid()` everywhere that we
have `test::OutputPath` and instead just rename to
`test::OutputPathRandomDir`
Bug: webrtc:15833
Change-Id: Ic9b69b5b599727f07b2906569a84a40edeecd1a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41730}
Add a small clean up in LossBasedBandwidthEstimatorV2ReadyForUse since IsReady() includes IsEnabled().
Bug: webrtc:12707
Change-Id: I20dfeb2ab31e7724041f89af9f312211a3ae3d23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339521
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41727}
This reverts commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b.
Reason for revert: Breaks WPTs
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I7b545e91f820c3affc39841c6e93939eac75c363
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41725}
This is a reland of commit 3f3f991c03bb4073a06da37c822daaa9deed9307
Original change's description:
> Let port allocator create ice tie breaker
>
> Moves the responsibility for creating the ICE tie breaker from the JSEP transport controller to the port allocator. This will allow a future change to separate the ICE tie breaker (which is sent over the network and hence known to the peer) from the "port allocator random" (that is used to seed the ICE candidate foundation crc32 checksum) as an implementation detail.
>
> BUG=webrtc:14626
>
> Change-Id: I3a9a0980238d6108b1b154f45de2975b08793b1c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281660
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41707}
Bug: webrtc:14626
Change-Id: Id3c8f257c5611958551bd66d7ce7a885bf8ba2f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41720}
Replace num_proc_channels() with num_output_channels() in
GainController2. The number of channels is only used in
InputVolumeController.
Bug: webrtc:7494
Change-Id: I6b3f66980a518401fefab304e18c9910eee28d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338922
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41717}
This reverts commit 3f3f991c03bb4073a06da37c822daaa9deed9307.
Reason for revert: API breaking change on PortAllocatorSession.
Is it possible to duplicate the ctor of PortAllocatorSession and remove
the deprecated one (the one without ice_tiebreaker) in another CL?
Original change's description:
> Let port allocator create ice tie breaker
>
> Moves the responsibility for creating the ICE tie breaker from the JSEP transport controller to the port allocator. This will allow a future change to separate the ICE tie breaker (which is sent over the network and hence known to the peer) from the "port allocator random" (that is used to seed the ICE candidate foundation crc32 checksum) as an implementation detail.
>
> BUG=webrtc:14626
>
> Change-Id: I3a9a0980238d6108b1b154f45de2975b08793b1c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281660
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41707}
Bug: webrtc:14626
Change-Id: I342c9a96ac1909244aedea6a7779f5682088a5fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339280
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41715}
This RaceChecker is intended to be used on API thread only when we are
not capturing, however, since StartCapture() can be called while already
capturing, we have to avoid using it to guard members that do not meet
this expectations. Use API checker for _captureStarted instead and move
the capture race checker down where we can be sure that capturing is not
happening.
Bug: webrtc:15181
Change-Id: I52f74b893f2c36c3ce0facd053b003fa497101b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338040
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#41714}
Use only one RaceChecker as intended with the original change. This gets
rid of specific RaceChecker for PipeWire members. Make PipeWireSession
guarded by API checker instead, since this member is accessed only in
[Start/Stop]Capture and move the race checker within PipeWire thread
loop lock. Also remove race check from OnStreamStateChanged where we
only modify one property guarded by API mutex.
Partially reverts a9d497b52dc21497fdfd0e8c03ab2f8559e02d15 reviewed
on https://webrtc-review.googlesource.com/c/src/+/326781.
Bug: webrtc:15181
Change-Id: I46449fce86611124688a65d5337771c75853f2ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338021
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#41713}
Ensure VideoSendStreamImpl does not register allocation on stray encoded
image if there is no active encodings.
Bug: chromium:41497180
Change-Id: I32afd7cc71f154dff240934e2be1745d8ead127c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41708}
Moves the responsibility for creating the ICE tie breaker from the JSEP transport controller to the port allocator. This will allow a future change to separate the ICE tie breaker (which is sent over the network and hence known to the peer) from the "port allocator random" (that is used to seed the ICE candidate foundation crc32 checksum) as an implementation detail.
BUG=webrtc:14626
Change-Id: I3a9a0980238d6108b1b154f45de2975b08793b1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41707}
To reduce number calls to the CreateVideoDecoder
Bug: webrtc:15791
Change-Id: I5d6ecc2e5e68165d4e012b3ad7edb6eaa40e1913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41706}
Providing unique identifiers for files and directories created as part
of unit tests.
Bug: webrtc:15833
Change-Id: If2835c362c47a111aa99b0e3c6ad6a33be061978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41704}
Test that BWE proving can be started without sending audio or video.
Bug: webrtc:14928
Change-Id: Ie55cb2de774f0c3b497b2636e7a6f5eac58d36a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337322
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41703}
Rework transport_description_factory to only have non-DTLS mode for
testing, and rewrite tests accordingly.
Bug: webrtc:11066, chromium:804275
Change-Id: Ie7d477c4331c975e4e0a3034fbbb749ed9009446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41697}
by creating an id collision and letting UsedIds resolve it.
Also avoid id=15 which is forbidden by
https://www.rfc-editor.org/rfc/rfc8285#section-4.2
so might cause interop issues in offers to implementations
not supporting two-byte extensions.
BUG=webrtc:15378
Change-Id: I27926f065f8e396257294da7acf2be9802169805
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41696}
As described in the issue. This won't compile under most compilers,
at least that what godbolt indicated.
So instead, using const vector<T> to prevent the compilation errors. Keep in mind this is a bit more restrictive as it also implies constness on the vector itself and not only its members, meaning we cannot push/pop.
Bug: webrtc:15829
Change-Id: I19c1223f0a3c56082da75b39c2afe152ae43a3a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41695}
These function calls break RAII and are redundant since NetEq and voe::AudioLevel have just been constructed. Remove them to fix an issue where NetEq ignores the min/max delay passed to its constructor.
This will not change default behavior since the defaults in NetEq::Config for max_delay_ms and min_delay_ms are both 0.
Bug: webrtc:15835
Change-Id: I27d4745ac85d6c6948796fe1d6286a0b79429474
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338209
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41694}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
