Use SPA_CHUNK_FLAG_CORRUPTED and SPA_META_HEADER_FLAG_CORRUPTED flags to
determine corrupted buffers or corrupted buffer data. We used to only
rely on compositors setting chunk->size, but this doesn't make sense for
dmabufs where they have to make up arbitrary values. It also looks this
is not reliable and can cause glitches as we end up processing corrupted buffers.
(cherry picked from commit cfbd6b0884)
Bug: chromium:341928670
Change-Id: Ida0c6a5e7a37e19598c6d5884726200f81b94962
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349881
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Original-Commit-Position: refs/heads/main@{#42292}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351563
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/branch-heads/6478@{#1}
Cr-Branched-From: 16fb7903e546051483720548168cd40cded7a040-refs/heads/main@{#42290}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
Bug: webrtc:42220378
Change-Id: I52816628169a54b18a4405d84fee69b101f92f72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42288}
And thus in those helpers query RateControlSettings field trials via propagated FieldTrialView instead of the via global field trial string
Bug: webrtc:42220378
Change-Id: I84f4bf42037d864519c4d2031d25cf909fd5010f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350305
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42286}
This ensure the absolute capture timestamp from the first audio sample
encoded in the payload is used for the corresponding rtp header.
Bug: webrtc:42226041
Change-Id: Ib8f2e3a5df5c82c5806171bd5b36a26d92fbea72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349265
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42281}
Replace factory that takes optional FieldTrialView with a constructor that takes non-optional reference to the same interface - all callers already guarantee it is not nullptr
Replace several local IsEnabled/IsDisabled helpers with the same helpers in FieldTrialView
In CongestionWindowPushbackController tests pass field trials bypassing global field trial string
Bug: webrtc:42220378
Change-Id: Ic49ad78919d834a5e3b9b69545d3b39088023a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349900
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42270}
This would allow network controllers, GoogCcNetworkController in particular to have access to Environment at construction and thus it can rely on propagated field trials and won't need to fallback to the global field trial string
Bug: webrtc:42220378
Change-Id: I36099522e3866a89a8c8d6303da03f7d5b1cad8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350201
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42260}
This adds a new mode to the ProbeController that sends probes every
second the first 5seconds. Intented usage is before normal media has
started flowing. If enabled, the max allocated bitrat is ignored during
these initial probes.
The purpose is to have a more accurate estimate at the beginning of a
call.
The cl also removes ProbeController::SetFirstProbeToMaxBitrate.
Bug: webrtc:14928
Change-Id: I04feefb2f1b82ff38b35ee2be810f9119c53536a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349924
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42252}
This is a cleanup in VP9 encoder wrapper. The removed code paths were only used in tests. In prod layers are configured explicitly via VideoCodec::spatialLayers[].
Bug: webrtc:42225151
Change-Id: I1de90039488b36e3c88e788c78e675bf2ee68f9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349222
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42250}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
Environment includes propagated field trials that can be later passed to
RemoteBitrateEstimators member, and would allow not to rely on the global field trial string
Bug: webrtc:42220378
Change-Id: Icf75a433c20352b2c22829c2148c92f69a2517aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349645
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42242}
This was a kill-switch for frame dropping in AV1 encoder. The frame dropping was enabled in June 2023. Since we have not heard about about any issues related to the frame dropping, we can remove the field trial.
Bug: webrtc:42225542
Change-Id: I4b2f1d5ff61e4ae3a4a7fc6711bb83f7d522fc6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349921
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42241}
It is not necessary for the caller to use it and the mute info can be
found on AudioFrame.muted().
Bug: None
Change-Id: I458f1f2e8489c1d8f8a9078b21f889b2540bdab9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349940
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42236}
which needs to be added to the remote codecs a=fmtp:
This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.
This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.
BUG=webrtc:10107
Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
This reverts commit e1607ed3a6.
Reason for revert: downstream project adjusted
Original change's description:
> Revert "h264: bail out early when failing to parse SPS/PPS ids"
>
> This reverts commit 4344eb713b.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > h264: bail out early when failing to parse SPS/PPS ids
> >
> > This currently gets caught later in the process by the H264 SPS/PPS
> > tracker but can be rejected explicitly here. The network observable
> > behavior should be similar and request a key frame after a 200ms delay, at least for entities that send such bad bitstreams
> >
> > BUG=webrtc:337076010
> >
> > Change-Id: I239c64efa7db631460ef9e9986d283335303df5f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349060
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Philipp Hancke <phancke@meta.com>
> > Cr-Commit-Position: refs/heads/main@{#42211}
>
> Bug: webrtc:337076010
> Change-Id: I15b815c69f1d25e41fb222d46359655242589fba
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349661
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42217}
Bug: webrtc:337076010
Change-Id: Ibe5a960b9b5fdf9a35e5dfffb47b78ade36b0cec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349700
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42223}
There is no TRACE_EVENT_ASYNC_STEP in the perfetto legacy API.
The corresponding legacy API that matches best is
TRACE_EVENT_ASYNC_STEP_INTO.
Bug: b/42226290
Change-Id: I6725973895878e34d96b6cd3314ab8de402a911b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42219}
This reverts commit 4344eb713b.
Reason for revert: Breaks downstream project.
Original change's description:
> h264: bail out early when failing to parse SPS/PPS ids
>
> This currently gets caught later in the process by the H264 SPS/PPS
> tracker but can be rejected explicitly here. The network observable
> behavior should be similar and request a key frame after a 200ms delay, at least for entities that send such bad bitstreams
>
> BUG=webrtc:337076010
>
> Change-Id: I239c64efa7db631460ef9e9986d283335303df5f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349060
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42211}
Bug: webrtc:337076010
Change-Id: I15b815c69f1d25e41fb222d46359655242589fba
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349661
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42217}
Support for parsing the packet is gated behind field trial
WebRTC-RFC8888CongestionControlFeedback/Enabled/.
Bug: webrtc:15368
Change-Id: Ib4478e821fe5a43510af5131543e7861cf54d901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348664
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42215}
This currently gets caught later in the process by the H264 SPS/PPS
tracker but can be rejected explicitly here. The network observable
behavior should be similar and request a key frame after a 200ms delay, at least for entities that send such bad bitstreams
BUG=webrtc:337076010
Change-Id: I239c64efa7db631460ef9e9986d283335303df5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349060
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42211}
If pacing rate, (current loss based bwe * pacing factor) is larger than the current upper link capacity estimate, reduce pacing factor to max of current bwe and upper link capacity.
Bug: webrtc:42220543
Change-Id: I5246da1f38530f8d411e7314adaa8651fc848f48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349601
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42210}
This is to avoid the case where the initial probe fail and the BWE is not updated, which can lead to a long period of low bandwidth.
Bug: webrtc:14928
Change-Id: Ie8f84270507b59995d57e4ab6e2a984570191529
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349580
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42208}
Implements a two-pass approach to packetization which creates
packets of an even size similar to RtpPacketizer::SplitAboutEqually.
This improves the bandwidth estimation.
The algorithm does a first pass with the existing packetizer, then
iterates through the resulting packet sizes and sums up the bytes left unused in each packet.
It then calculates a new maximum packet length as
configured_max_packet_len - ((unused_bytes - packets + 1) / packets)
adjusts for the overhead and re-runs the packetization algorithm.
For example, a list of OBUs with sizes
{1206, 1476, 1431}
currently gets packetized greedily as payload sizes
{1200, 1200, 1200, 523}
With this change, it gets packetized as
{1032, 1032, 1032, 1028}
This change is guarded by the field trial
WebRTC-Video-AV1EvenPayloadSizes
which is acting as a rollout flag.
BUG=webrtc:15927
Co-authored-by: Shyam Sadhwani <shyamsadhwani@meta.com>
Change-Id: I4f0b3c27de6f06104908dd769c4dd1f34115712c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348100
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42203}
Start introducing ArrayView to AudioFrame and code that flows down
from there. In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
audio buffer. When AudioFrame is not initialized however, data_view()
will return a nullptr whereas the current data() method never returns
nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
the samples per channel and number of channels that's required for
accurately reserving the returned mutable ArrayView.
A notable behavior change is that if the requested number of channels
is larger than supported or the calculated buffer size is too large,
the function will trigger a check.
* Add TODOs for following work.
Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
Instead of from the global field trial string.
Bug: webrtc:42220378
Change-Id: Iddb41429e388792de02f702b4caa35689c57d9e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347720
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42201}
This is the first new style trace event so this CL adds and updates
WebRTCs Perfetto configuration.
* Changes all #includes to target "third_party/perfetto". Added this
to DEPS.
* Expose the Perfetto public config in the "tracing" group using
an all_dependent_configs statement. This means the config is included
in all parts that include the "//:tracing" group. However, direct
perfetto includes are banned per DEPS.
* In order to expose Perfetto types (ie Flow/TerminatingFlow) the
perfetto headers are a dependancy on all targets. This should not
affect binary size as these are not used when perfetto is not enabled
and will not be linked.
Bug: b/42226290
Change-Id: I5711d81dae95ee907cbcd41bf1ee9b55d2ec595c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349161
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42197}
Environment provides non-null interface for FieldTrialsView and thus VideoCodingModule no longer need to rely on FieldTrialBasedConfig class to provide field_trials when not passed at construction.
Bug: webrtc:10335
Change-Id: Iedfb29e8b29056618a85f2e7a1528da29e3be5c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347701
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42163}
These are required by the Perfetto API and the missing argument prevents
the use of Perfetto.
Bug: webrtc:15917
Change-Id: Ie40c0344dc9d8cd40f7c751b133d150b975a33c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347702
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#42147}
This cl adds an implementation of the RTCP feedback packet as specified in https://www.rfc-editor.org/rfc/rfc8888.html
Bug: webrtc:15368
Change-Id: I0b9a7fb15512ff9f9e721efd8e03ebe981a8d9bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347901
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42140}
