As this is handled higher up the pipeline in a single
place for all encoders/decoders
Bug: webrtc:10460
Change-Id: I95b0a69aecaf07283c8776ac0d7e85d097e3576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139882
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28172}
Using this flag, an encoder may inform the RTP sender module that
the packet is not elligible for retransmission. Specifically, it
may not be retransmitted in response to a NACK message,
nor because of early loss detection (see CL #135881).
Bug: webrtc:10702
Change-Id: Ib6a9cc361cf10ea7214cf672e05940c27899a6be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140105
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28169}
Some of the TODOs associated with webrtc:10336 which are
currently in the codebase have recently been resolved,
but not all relevant TODOs have been removed.
TBR=kwiberg@webrtc.org
Bug: webrtc:10336
Change-Id: Iff1d0fc94dee5bf49226f6ea3d9127fea77e9d68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139902
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28167}
Replaced by separate methods
SendPictureLossIndication and SendFullIntraRequest.
The split SetKeyFrameRequestMethod/RequestKeyFrame implicitly
requires that the two methods are called on the same thread, to avoid a
data race. After downstream code is updated, both deprecated
methods and the member |ModuleRtpRtcpImpl::key_frame_req_method_| can
be deleted.
Bug: None
Change-Id: I454f6d16b667f2306cba0dec467ddc183ad449c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140043
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28163}
It was introduced, together with the PacketLossStats class, in cl
https://codereview.webrtc.org/1198853004 (#9568). It is unused in webrtc,
but there's downstream usage of the PacketLossStats class, which
should perhaps be moved or deleted in a later cl.
Bug: None
Change-Id: I17a3d5c8748f2cc9809c438630cbe8ab680466c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140042
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28153}
Insert first packet before calling to decode.
Bug: webrtc:10690
Change-Id: I721b7af0506f0dbaf4fa2ed6a9ba6a87250d08f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139103
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28149}
Loss notifications may either be sent immediately, or wait until another
RTCP feedback message is sent.
Bug: webrtc:10336
Change-Id: I40601d9fa1dec6c17b2ce905cb0c8cd2dcff7893
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28142}
This is an usual error while using neteq_quality_test. This tool
does not support wav files as input. Adding a validation.
Bug: webrtc:10690
Change-Id: I18ed308d2f688106728df5df25e0a58c7170f411
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139104
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28141}
The test RunPlayoutAndRecordingInFullDuplex makes the speakers play the
signal it simultaneously records from the microphone, which can cause
full howling.
The test itself measures buffer usage and does not depend on what signal
is played through the speakers. This change mutes the speakers to prevent
howling when running modules_unittests.
Bug: webrtc:10704
Change-Id: I6176adb2fb5ce0e4d86f6f447476be8a88c2f2cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139889
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28140}
Currently, if LNTF and NACK messages are both created, they will
be sent out in separate RTCP messages. This is wasteful.
This CL is the first of in a series of CLs that will ensure that
these feedback messages can be buffered together, without introducing
more of a delay than the CPU time required to process both messages.
Bug: webrtc:10336
Change-Id: I950324112ee346695a12a17d025483ea5e99c732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139112
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28136}
Instead of setting a runtime, allow neteq_quality_test to
consume a complete file using --runtime_ms -1
Bug: webrtc:10690
Change-Id: I90d35cf31996d9336fef817b9332a2cd1d04e77e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139101
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28134}
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: If73bf7ff9c1c0d22e0d1caa245128612850f8e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138268
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28104}
Currently, the packet in the history that most closely matches the bit
budget between two PacedSender::Process() calls is selected to be
retransmitted. This usually means that the smallest packet in the
history is selected over and over.
With this new field trial, we ignore the size constraint (since you're
sending padding, you obviously have some bandwidth to spare) and
instead prefer packets that have the fewest transmission times first,
and after that we prefer new packets over older ones. This way, in
case of available bandwidth but small loss, these padding packets have
a greater chance of actually being useful to the receiver.
Bug: webrtc:8975
Change-Id: I15a69231f44bfbefcb9ab38dd7886b966e3af6fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135954
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28084}
The probe configuration is currently a single field trial. To allow
multiple experiments with non-overlapping subsets of these keys I've
added a few extra keys that override different subsets of the config.
Bug: webrtc:10394
Change-Id: I54ffd1105129794fcdae4cce314910acaa4074af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138274
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28076}
The imports of Packet, ReceivedPacket from ForwardErrorCorrection::
collides with other usages of the names introduced in a followup CL.
Bug: webrtc:9883
Change-Id: Ib042c9091ad8e350cbdf01b837af09c820dbff33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138279
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28075}
In addition to the 48 kHz that we've always used.
Bug: webrtc:10631
Change-Id: I5e4f6600e39a463d20d3988db098c7e38281f4a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138264
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28074}
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats
In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
audio streams, but the code is not wired up to make use of these stats.
In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.
[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067
Bug: webrtc:10455
Change-Id: Id8940090cc9c121e8f06ccdb068a22ce24c07092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138066
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28072}
This is part of implementing RTCRemoteInboundRtpStreamStats. The CL was
split up into smaller pieces for reviewability. Spec:
https://w3c.github.io/webrtc-stats/#dom-rtcremoteinboundrtpstreamstats
In [1], ReportBlockData was implemented and tested.
This CL adds the plumbing to make it available in MediaSenderInfo for
video streams, but the code is not wired up to make use of these stats.
In follow-up CL [2], ReportBlockData will be used to implement
RTCRemoteInboundRtpStreamStats. The follow-up CL will add integration
tests exercising the full code path.
[1] https://webrtc-review.googlesource.com/c/src/+/136584
[2] https://webrtc-review.googlesource.com/c/src/+/138067
Bug: webrtc:10456
Change-Id: Icd20452cb4b4908203b28ae9d9f52c25693cf91d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138065
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28071}
Because of a low bitrate target, base layer has drops much more frequently
than other layers. But it reduces overall framerate, especially then
input framerate is low (5 fps).
This CL allows pre-layer drops and disables droppoing on higher spatial
layers for screenshare, solving the issue.
Additional care have to be taken then new spatial layers are enabled
dynamically to not create non-compatible with RTP references.
Bug: webrtc:10257
Change-Id: Ie056484c99a3f35ff4405ef71337dc2d034db8bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138262
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28063}
It seems unnecessary to lock it if not actually reinitializing.
Bug: webrtc:10205
Change-Id: Ib3292e1d640a92a7df77400aebe9583cf877f824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/115460
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28060}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
This CL makes it more flexible and easier to include/exclude H264 code
when using other build systems because it delegates the decision to
remove the code to the preprocessor instead of GN.
This CL should be a noop, and for WebRTC/Chromium the GN param
`rtc_use_h264` will still be the only thing to change in order to
include/exclude H264.
Moving code that requires ffmpeg or h264 out of the #ifdef/#endif
part should break the build since dependencies are only added if
`rtc_use_h264=true`.
Bug: webrtc:9213
Change-Id: Ibc04edc2f6b9e51489ffe638d5be4b32959cdca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137430
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28055}
Reland with fixes.
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: Id40a780d461e6b51cb44d275b8aa5d7b348d3586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138215
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28054}
This reverts commit eb1754c5750dfcad23ac62b47aa3aa2176ae7be2.
Reason for revert: breaks downstream projects
Original change's description:
> VP9 screenshare: Don't base layers frame-rate on input frame-rate
>
> If input framerate is a little unstable, using it to cap layers will
> make output framerate even smaller for longer periods of time.
>
> Also, fix screenshare_loopback test for low-fps vp9 testing.
>
> Bug: webrtc:10257
> Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28050}
TBR=ilnik@webrtc.org,ssilkin@webrtc.org
Change-Id: I82bfbac58249cfe0da5ff565aa97a4745fd078ff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10257
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138213
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28051}
If input framerate is a little unstable, using it to cap layers will
make output framerate even smaller for longer periods of time.
Also, fix screenshare_loopback test for low-fps vp9 testing.
Bug: webrtc:10257
Change-Id: I64aa087e859ab4ab8e484c9ab7f5ac0fb18bd37d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138204
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28050}
Change the way the forget factor converge to the steady state so that we don't overemphasize the first packets received.
The logic is controlled by the delay histogram field trial which has an added parameter to control if emphasis should be even (c=1, default) or put on later packets (c>1) until we reach our steady state forget factor.
Bug: webrtc:10411
Change-Id: Ia5d46c22d1a4a66994652f71c8cde664362bfacb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137050
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28039}
WebRTC combines VP9 SVC spatial layer frames into superframe and passes
it to a decoder. The chromium HW VP9 decoder (wrapper) needs to know
location of each spatial layer frame in the frame buffer. To provide
decoder with such information this CL:
- Adds Set/SpatialLayerFrameSize methods to EncodedImage.
- Sets size of each spatial layer frame on superframe at assembly stage.
Bug: webrtc:10495
Change-Id: I68c3c0d668c67dfa1740e004059d860dd98f67f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28032}
This reverts commit a8ae407a480a2a9982eecf9e3a9b10da5373cd9a.
Reason for revert: This CL incorrectly affects non-experiment branch. A new CL affecting only the experiment will be uploaded.
Original change's description:
> Add ability to cap the video jitter estimate to a max value.
>
> Bug: webrtc:10572
> Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27744}
TBR=stefan@webrtc.org,mhoro@webrtc.org
Bug: webrtc:10572
Change-Id: I4af334168ca70ecfae7fd18fc7c852819a98d866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138063
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28025}
