Ensure probing is not instantiated again until after timeout if a probe has been sent to max rate.
Bug: webrtc:42224658
Change-Id: I7d0d2edcfa81b1b454ea5748962af5a2070b347c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363240
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43068}
We currently specify stream parameters to be a range for both framerate
and resolution, where preferred value is specified. The preferred value
doesn't seem to be taken into account and we end up accepting resolution
from 1x1 to MAX_INTxMAX_INT. In case the other side tries to first match
with lower resolution than requested, we will happily match it and start
streaming low quality video. We should instead request the exact stream
parameters as specified by requested capability. This capability always
come from what has been originally reported as supported so it shouldn't
happen we don't find a matching stream. This also applies to requested
video format. We previously requested mjpg for streams with resolution
higher than 640x480, but it doesn't necessarily mean the camera supports
mjpg for the requested resolution. Again, refer to requested capability
in this case as it should indicate what is supported and we know we can
request exactly the same video format. It can happen that framerate is
set to 0 as unspecified. In that case keep using a range as before, but
with more sane values.
Bug: webrtc:42225999
Change-Id: I46d8e83c636e25e12c45a462596fee1d5e59888e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362820
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Cr-Commit-Position: refs/heads/main@{#43067}
This reverts commit 3753c8190e3f0aca6758a5521e33f8b5d4f09ab4.
Reason for revert: Break assembling of hardware encoded h264 P frame on
weak network condition.
Original change's description:
> h264: fix first_packet_in_frame logic for multislice in a single rtp packet
>
> a frame must be (or should be) first when it contains either SPS (but not just PPS),
> is an IDR or is a slice with first_mb_in_slice == 0.
>
> Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit
> into a single RTP packet which can happen with small 320x196 frames
>
> BUG=webrtc:352379280,webrtc:346608838
>
> Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42652}
Bug: webrtc:368335257
Change-Id: I07725c78be628bff71b79b8b9369677e39f5f5ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363080
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43062}
In many cases, the framerate can be specified as list of possible values
and in that case, we would end up with max FPS to be set to 0 as this
case was not handled.
Bug: webrtc:42225999
Change-Id: I036af6db1da3309b1310b754504369e8fe392d09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362961
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@{#43057}
Passing Environment instead of Clock into this class simplifies some plumbing for downstream consumers that need to read field trials within this class.
Bug: webrtc:362762208
Change-Id: Ia501e9f7f1d91a8115a2f71fb005dd35146db172
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362535
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@{#43048}
We need to have imported proto as proto_data_sources in BUILD.gn to
run the action remotely without workaround config in siso.
Bug: b/366137880
Change-Id: I053774f00b761520a8a85154e386da3edb8f39b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362680
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Takuto Ikuta <tikuta@google.com>
Commit-Queue: Takuto Ikuta <tikuta@google.com>
Cr-Commit-Position: refs/heads/main@{#43040}
Use the DTX state from inside the Opus encoder instead of trying to
mimic the logic outside.
Bug: None
Change-Id: I852044fee261a5b7f9255c557a27adfd0b1701bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43034}
They are now passed as part of the Environment
Bug: webrtc:362762208
Change-Id: I02868e9f41533a546f62fe30fdc6f3a7708eb346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362084
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43032}
The class itself and its unit test remains, for now, but will be removed
later.
Bug: webrtc:14867
Change-Id: I36cec8fca7913663f63c53622ed2760e5e048c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362580
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43023}
Checks if the DirtyRegionMode property is present in GraphicsCaptureSession and logs a boolean histogram with the result.
Detecting support for this property means that the WGC API supports
dirty regions and it can be utilized to improve the capture
performance and the existing zero-herz support.
See also https://issues.chromium.org/issues/347991512 for more details
on how to detect support for dirty regions in WGC.
Bug: chromium:40259177
Change-Id: Ia316c4ece54bd93cfef1fa23c199675c64143f76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362240
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43015}
AV1E_SET_MAX_CONSEC_FRAME_DROP_MS_CBR was added in https://aomedia-review.googlesource.com/c/aom/+/192402. It allows to configure max number of consecutive frame drops using time units. Use it instead of AV1E_SET_MAX_CONSEC_FRAME_DROP_CBR.
Bug: webrtc:351644568
Change-Id: I73265d5258d681926eb5b65e32c2a61b26c310ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360842
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Marco Paniconi <marpan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42995}
to avoid relying on the global field trials.
Bug: webrtc:362762208
Change-Id: I94e96f0a3f16cfd64f7deb4deb4aaa924ac1bba8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361865
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42982}
SvcRateAllocator assumed no temporal layering for screencast content and allocated all bitrate to base temporal layer. Now it distributes bitrate to spatial and temporal layers (if configured) no matter of content type.
Bug: webrtc:351644568, b/364190191
Change-Id: I445f0157d2c14cad033648693dc0564ae97023e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362080
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42979}
ChannelReceive is now owning and interfacing with NetEq directly.
A new ResamplerHelper is added to acm_resampler.cc/.h, to do the
audio resampling that was previously done inside AcmReceiver.
AcmReceiver still remains, since it is used in other places for now.
Bug: webrtc:14867
Change-Id: If3eb6415e06b9b5e729d393713f3fccb31b0570f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361820
Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42974}
This makes it simpler to use in more contexts.
Bug: b/364184684
Change-Id: I1b08ebd24e51ba1b3f85261eed503a78cd006fd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361480
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42956}
This allows to utilize libvpx optimizations considerably improving performance.
The change happens inside libvpx_vp9_encoder and is invisible to other parts of webrtc.
This CL includes unit tests, an E2E test already exists: StandardPath/PeerConnectionEncodingsIntegrationParameterizedTest.Simulcast/VP9 in peerconnection_unittests.
Bug: webrtc:347737882
Change-Id: I03bc27c920787a7305a9775e6341e26904592fb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360280
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42931}
We keep information about the PipeWire camera status as a member of the
PipeWire session, but it's never updated and remains in uninitialized
state. Make sure it gets updated once PipeWire is initialized or when it
fails. There is currently no use for this member variable, but there is
a plan to use it so I'm rather keeping it instead of removing it.
Bug: webrtc:42225999
Change-Id: If409761b148be8f0724fd9ab7a1ed4cf0e459503
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360922
Reviewed-by: Andreas Pehrson <apehrson@mozilla.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42926}
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}
