This is a minor change for places where we use
AudioFrame::kMaxDataSizeSamples sized intermediary buffers. The change
uses `std::array<>` instead of C style arrays which allows for use
of utility templates that incorporate type based buffer size checking.
Also adding `ClearSamples()` method, which complements CopySamples.
Bug: chromium:335805780
Change-Id: I813feb32937e020ceb9ca4b00632dc90907c93fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351681
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42400}
Both CaptureFrame() and ProcessBuffer() hold a lock over the frame queue
and it happens that one waits for the other, causing unnecessary delays
since we already work with a queue having two frames, but this way we
don't really need a queue. Instead, keep reference to the last processed
buffer, which we will always use in CaptureFrame() and update it every
time at the end of ProcessBuffer(). This avoid unnecessary waiting for
the lock over the queue to be released.
Bug: chromium:333945842
Change-Id: I4afeb1daacd342e92578a50ac6e1c89a691bb8f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350042
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#42394}
Split out time_util.h and cc from target rtp_rtcp to its own target.
This is to avoid possible circular dependencies and not having all targets using them to depend on the full RtpRcp module.
Bug: webrtc:343076000
Change-Id: I7b3c84456b17f1920f71afdd5a644d27e28caed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352480
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42392}
This depenency is not needed and may lead to a circular dependency. The cl removes old unused functionaliy to log BWE related statistics using compile time flags.
Bug: webrtc:42225697
Change-Id: I6cc01b367c0c48ab30f34c12a10afc58d1e7822f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352142
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42386}
From the new header file:
* MonoView<>: A single channel contiguous buffer of samples.
* InterleavedView<>: Channel samples are interleaved (side-by-side) in
the buffer. A single channel InterleavedView<> is the same thing as a
MonoView<>
* DeinterleavedView<>: Each channel's samples are contiguous within the
buffer. Channels can be enumerated and accessing the
individual channel data is done via MonoView<>.
There are also a few utility functions that offer a unified way to check
the properties regardless of what view type is in use.
Bug: chromium:335805780
Change-Id: I28196f8f4ded4fadc72ee32b62af304c62f4fc47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349300
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42377}
This is done to better reflect the responsibility of the class.
The implementation implement a new interface FeedbackGeneratorInterface. The purpose of the interface is to allow a new implementation that supports RFC 8888.
Bug: webrtc:42225697
Change-Id: Id087dd7422abbcd6016693c076a65f4c4efd5712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42366}
The test tests that a route change does not cause BWE do drop unless the adapter is changed.
Bug: webrtc:42221538
Change-Id: I49be55172aff285c55d2564ec3389f3fc7c01d62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350820
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42347}
So that this class can use propagated field trials instead of the global
Bug: webrtc:42220378
Change-Id: Ic1dba0c4967735606904329f7e9e6c09f186b809
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42326}
This makes the downcasts currently used in eg
modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
safer.
Bug: webrtc:339815768
Change-Id: Ie6c7ab84666d399dbca4c95846b850aac5327f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350361
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42325}
Repeated initial probes are sent every second until
ProbeController::OnMaxAllocatedBitrate is invoked (Media is beeing sent) or 5s has passed.
Each probe has a duration of 100ms, sent in packet bursts every 20ms.
ProbeController::waiting_for_initial_probe_result_ is no longer needed
and is removed.
Setting field trial for duration between probe packets bursts are moved
from BitrateProber to ProbeController.
Bug: webrtc:14928
Change-Id: I3170533f679fc2cc2aa5402e248fa1f6996d3ccd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350640
Reviewed-by: Diep Bui <diepbp@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42323}
- Assume a non-zero probability of starting in transparent state
(transparent mode can be reached sooner).
- Relax the requirements for when the filter is considered converged
(reduces the risk of incorrectly entering transparent mode in the
presence of near-end noise).
Bug: b/340578713
Change-Id: I6be9b5b74457066f9900c8020c0ebf19623a70df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350602
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42318}
This makes Environment and thus field trials a required parameter, thus
RemoteBitrateEstimators member no longer need to fallback to the global field trial string.
Bug: webrtc:42220378
Change-Id: Ieb6ff442d5fde5fa9715573c758a7e078f0ceea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349922
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42314}
This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4
Original change's description:
> Split digest methods from ssl target into digest target
>
> 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}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
Environment guarantees field trials are provided, thus GoogCcNetworkController doesn't need to fallback to the global field trials.
Bug: webrtc:42220378
Change-Id: Iff8e00504b43b074dc41b5ac9908fd0e2be18959
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350540
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42300}
The 'apmtest' folder contains code that is not part of any build graph
and has not been updated since 2017 since the code migrated locations.
At a glance, it does not seem to be testing anything specific to the
audio-processing module either.
This implicitly resolves the usage of the deprecated ALooper_pollAll API
by removing the code entirely.
Bug: webrtc:42225691
Change-Id: I79e14140ee40c567e1d07431f874d5f48e39d384
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350270
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42299}
To allow various VideoBitrateAllocators to use propagated rather than global field trials
This relands the
https://webrtc-review.googlesource.com/c/src/+/349920
where patchset#1 is identical to the original change,
patchset#2 undoes (postpones) the expectation downstream propagates the Environment too.
Bug: webrtc:42220378
Change-Id: I4a9a32bb0926a875d37f3ba19dd5309e97546553
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350364
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42298}
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.
Bug: webrtc:338232699
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-Commit-Position: refs/heads/main@{#42292}
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}
This reverts commit e1607ed3a619ae30cf8564ce401df5e03dd7bf4b.
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 4344eb713bb9a6d04d922d00fb492dfb31c9111f.
>
> 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 4344eb713bb9a6d04d922d00fb492dfb31c9111f.
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}
