This CL ensures we properly points to deps shared with chromium,
e.g. '//third_party/abseil-cpp...' and not '../third_party/abseil-cpp...'
NB: This is only applied to dependencies which were missing,
and doesn't fix existing ones.
Bug: webrtc:10037
Change-Id: If4bbb00df39401c65def9d56e36e5feb5d67b9dd
Reviewed-on: https://webrtc-review.googlesource.com/c/111600
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25762}
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.
Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
This change introduces a clockdrift detector operating on the estimated
delay of the echo path delay estimator. Each time the delay estimate
changes it is compared to previous estimates. If the estimates are
slowly increasing or decreasing, clockdrift is detected.
Four different patterns are considered clockdrift:
- k, k+1, k+2, k+3
- k, k+2, k+1, k+3
- k, k-1, k-2, k-3
- k, k-2, k-1, k-3
A delay estimate history matching the three last elements in one of the
patterns is considered probable clockdrift. Matching all four elements
is considered verified clockdrift.
If the delay is constant for some time after clockdrift is detected the
clockdrift detector will revert to no detected clockdrift.
The level of clockdrift is reported via an UMA histogram.
Bug: webrtc:10014
Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a
Reviewed-on: https://webrtc-review.googlesource.com/c/111381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25758}
This class adds logic for aligning what part of a test video has been
encoded from a reference video. It does that by cropping and zooming in
on a region of the reference video that most closely matches the test
video. A small cropping does not have much impact on human perception,
but it has a big impact on PSNR and SSIM calculations.
For example, if the test video is cropped with one row in the top and
bottom, adjusting for this improves average PSNR from 27.7146 to
29.3357 and average SSIM from 0.934891 to 0.95318 in an example test
video.
TBR=phoglund
Bug: webrtc:9642
Change-Id: I02cfe0e2261fb58df8cdb1e15ba93285e3dc4538
Reviewed-on: https://webrtc-review.googlesource.com/c/99480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25755}
- Rename avg_max_bitrate_kbps to link_capacity_estimate_kbps and change
the type to optional.
- Remove the RateControlRegion enum. The old code seems to have the invariant
that the region is kRcMaxUnknown iff avg_max_bitrate_kbps is uninitialized.
- Change floats to double.
Bug: webrtc:9942
Change-Id: Ic071a11ec4950053ec92beaa06f28f43192521d7
Reviewed-on: https://webrtc-review.googlesource.com/c/111247
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25752}
This test fails on slow runners when TCPChannelClient has not
yet finished communication, but the test thread times out checking
that mock methods are called.
Bug: webrtc:9955
Change-Id: Ia91ada6b01ca1bab48afa57fe76aedd08770a641
Reviewed-on: https://webrtc-review.googlesource.com/c/111383
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25751}
inline InOrder check
remove it from IsRetransmit check as redundant
avoid call to IsRetransmitOfOldPacket when packet arrived in order
take current time once
Remove packet overhead counting as unused
Bug: None
Change-Id: Icd8bf69b5076e4469c349529c9ac79a1b15d9515
Reviewed-on: https://webrtc-review.googlesource.com/c/111746
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25749}
Since the link capacity is designed to be a more stable value, we don't
need the smoothing. This allows us to react faster to changes in link
capacity while still avoiding to react to changes in target bitrate due
to normal control behavior.
Bug: webrtc:9718
Change-Id: I2fbf6bb882f312a7b28ea43d27057886d035ac45
Reviewed-on: https://webrtc-review.googlesource.com/c/111511
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25745}
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.
In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory
Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
This allows sending the full BitrateAllocationUpdate to the encoder.
This will be used in a later CL to use the link capacity field in the
update to control the Opus decoder.
Bug: webrtc:9718
Change-Id: I1c228cc318c7f9f1b0fec232e27732177b80705a
Reviewed-on: https://webrtc-review.googlesource.com/c/111509
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25739}
This way it can be forwarded to lower layers. This makes it easier to
add information without having to change signatures of intermediate
classes. This will be used in a later CL to use the link capacity in the
Opus decoder.
Bug: webrtc:9718
Change-Id: I4a4c9d104fedb0e4a0bb7f14d169475940edbf7e
Reviewed-on: https://webrtc-review.googlesource.com/c/111508
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25738}
This change removes the ability to set CryptoOptions through the PeerConnection
Factory in both Java and IOS. Native will be removed after the Chromium change
lands. The semantics have been changed such that these options should only be
set on individual PeerConnections and not directly on the Factory itself. This
allows for more flexibility in setting CryptoOptions for PeerConnections which
are created as part of a factory.
Bug: webrtc:10020
Change-Id: I9ef3d431e728927b9ced5de6188cedeb2671254b
Reviewed-on: https://webrtc-review.googlesource.com/c/111560
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25736}
JsepTransportController got a bit ugly with one super long method.
Splitting it to two, so that MediaTransport creation is separated.
Bug: webrtc:9719
Change-Id: I0b5aead2f96d79d6fc369a16810be58c8a661e71
Reviewed-on: https://webrtc-review.googlesource.com/c/111288
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25732}
Move HdrMetadata to ColorSpace as part of preparing for joint transmission
of these two objects.
Bug: webrtc:8651
Change-Id: Ie948011a2c0106d5967cb5ef3b9565217e798272
Reviewed-on: https://webrtc-review.googlesource.com/c/111481
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25730}
Avoid that the client code relies on the adaptive digital mode being
enabled by default (error prone).
Bug: webrtc:7494
Change-Id: I765fecf535cf31a2163e10595a42520473c233b6
Reviewed-on: https://webrtc-review.googlesource.com/c/111586
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25728}
This CL changes the behavior for RunFor and RunUntil so they do not
anymore restart the underlying streams every time they are called.
This has a side effect on the semantics of the calls. Previously,
both RunUntil and RunFor would restart the session and run until the
given time had passed. Now RunFor will still run for the provided
duration, however, to make the name of RunUntil more correct, it
will run until the time since start is equal to the max_duration
parameter. An extra overload of RunUntil was added to allow using
this behavior without providing an ending condition.
Bug: webrtc:9510
Change-Id: I9fe56a44116907fba3d102894b5c96af2ba6cffb
Reviewed-on: https://webrtc-review.googlesource.com/c/111502
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25726}
This CL utilizes the input frame rate in the RTCVideoEncoderH264, by setting it into VT Property.
The main purpose is to guide VT encoder to make correct decision of the encoded frame size.
Bug: webrtc:10015
Change-Id: Id5c89f2876539f3181030f49b546326fc40b8ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/111420
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25724}
Test disabled on TSAN due to repeated failures. There are data races
in a low-level syncronization primitive (semaphore). Since
syncronization primitives should handle that, I think TSAN may be
configured incorrectly.
The locking scheme is written entirely in the unit test. This means we
are losing some test coverage of *unit tests*.
TBR=jamiewalch@chromium.org
Bug: webrtc:10019
Change-Id: Ieafa00a5a789acf8d0bacf6ad669c6daca7efa17
Reviewed-on: https://webrtc-review.googlesource.com/c/111585
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25723}
This reverts commit a13be019017449c57f48203d0fb778f34f7553a7.
Reason for revert: The GN definitions cause problems for downstream tooling. They're also generally complicated and reach deep into Chromium's build which is undesirable. Setting `rtc_use_pipewire = true` by default should also be re-evaluated.
Original change's description:
> Default to dlopening the PipeWire.
>
> Reuse the existing infra from Chromium to do that. Additionally the
> target_gen_dir needs to the added to the include directories, otherwise
> the Chromium build will fail as it won't find the generated stubs. Also the
> pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
> require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
> doesn't work with them correctly. With all these changes in place the PipeWire
> support is enabled when compiling on Linux.
>
> Bug: chromium:682122
> Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
> Reviewed-on: https://webrtc-review.googlesource.com/c/111081
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#25720}
TBR=phoglund@webrtc.org,mbonadei@webrtc.org,braveyao@webrtc.org,tomas.popela@gmail.com
Change-Id: Iec20b07cb1cff7d57f8114ac6ec2d0d250e61214
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/111584
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25722}
Reuse the existing infra from Chromium to do that. Additionally the
target_gen_dir needs to the added to the include directories, otherwise
the Chromium build will fail as it won't find the generated stubs. Also the
pw_properties_new() was replaced with pw_properties_new_string() as it doesn't
require a variadic parameter because the //tools/generate_stubs/generate_stubs.py
doesn't work with them correctly. With all these changes in place the PipeWire
support is enabled when compiling on Linux.
Bug: chromium:682122
Change-Id: I3bbc5efaecd9a08e20cbcf998b2cb534224eae7d
Reviewed-on: https://webrtc-review.googlesource.com/c/111081
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Commit-Queue: Tomáš Popela <tomas.popela@gmail.com>
Cr-Commit-Position: refs/heads/master@{#25720}
The target bandwidth is a more stable target rate as it does not follow
the variation in the control signal directly. It's intended to be used to
configure the audio frame length.
Bug: webrtc:9718
Change-Id: Idcc83ba0fef90e0ead2926d18ba6893a2b0f085f
Reviewed-on: https://webrtc-review.googlesource.com/c/107729
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25718}
This increases expected value of maximum buffer level in VP8/9 tests
up to 1 second and thus alignes it with the value that WebRTC uses by
default for these codecs.
Bug: webrtc:10017
Change-Id: I8fd41e8006f11c230d844a053c04656408c2ec97
Reviewed-on: https://webrtc-review.googlesource.com/c/111503
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25716}
