BaseTime represents fixed point in time with unknown epoch and thus
make sense to convert to Timestamp type, however Timestamp should always
be positive. however legacy tests expect GetBaseTimeUs to return negative time sometimes.
Bug: webrtc:13757
Change-Id: I3f780a7775fdd1e271402c59384c1298db76f75a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264549
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37076}
When target_os is set to "fuchsia":
BUILD: suppress Wundef flag
DEPS: download the Fuchsia SDK
audio_encoding: add header include
video_capture: video_capture_factory is not yet implemented for Fuchsia
so we add a null capture factory when building for Fuchsia.
Bug: webrtc:14061
Change-Id: Id6ca7418859c85293a0a5e2a8427807ee039db2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262200
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37030}
Ensure each test create its own PacerController.
Move (most) operations on the pacer controller to the actual test. (the
rest should be moved too eventually....)
Use only one test fixture.
Bug: none
Change-Id: I0b8eee9d2c2f91f7102858a1a544e45e8b0b7b5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264120
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37017}
It replaces the relative arrival delay tracker which is equivalent.
This results in a slight bit-exactness change but nothing that should affect quality.
Bug: webrtc:13322
Change-Id: I6ed5d6fdfa724859122928a8838acce27ac2e5d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263380
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37004}
Puts the whole block in contiguous memory and reduce pointer look-up.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: I264aaf764bf53a29f23249105f704b2fdbd7e51c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263203
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36983}
This CL allows the users to propose custom resolution to server
for the captured pipewire streams.
Bug: chromium:1291247
Change-Id: Iaae2c73df1a5f5ebac651ce7d087af4c273113c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263360
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36979}
There are two cases that can be confusing for applications developers
which may result in the playout delay not being set as intended.
First, it is not well defined which min playout delay should be used
when multiple are set. This changes adds a warning to alert application
developers that they are setting multiple playout delays.
Second, if the playout delay header extension is used, developers must
be careful that the max playout delay is always larger than the min
playout delay, otherwise the behaviour is undefined. This change logs an
error when this case is detected.
Bug: None
Change-Id: I8477d48ef64636da080792362fa898e42f038bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263202
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36977}
Putting these classes in a sub folder increases
structure and clarifies that they are used as
helper classes. Affected classes in this change:
* CodecTimer
* InterFrameDelay
* RttFilter
VCMTiming will be moved in a separate CL.
Additional changes:
* Remove VCM prefix from class names.
* Introduce granular BUILD.gn targets.
* Update some includes.
Bug: webrtc:14111
Change-Id: Ia75128aa955a819033b97d4784cb61904de7230b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262960
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36975}
This helper class currently lives in `modules/video_coding`,
but it's only users are in `video/`. Thus, it makes sense to
move the class to `video/`.
Bug: webrtc:14116
Change-Id: I0d3f8961bc8f5fe80f3100dbbd309b206020e6d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262963
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36973}
This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.
It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
Step one in making it a separate type, that will be done as a
followup, after downstream code is updated to use the new name.
Bug: webrtc:11607
Change-Id: I6fa664a0729b1cfd71b7f02b6441880beee0e741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262806
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36946}
AcmReceiver basically only does resampling, which is not something we need to test for bit-exactness.
NetEq bit-exactness is already tested with the same rtp input file as these tests.
Bug: None
Change-Id: Ibb3936c86098e0eea944860d33e2c13bf046e40b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262816
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36944}
Structs with user-declared constructors are not aggregates and cannot
be initialized with designated initializers. Remove declarations that
don't actually affect anything.
Bug: chromium:1284275
Change-Id: Ib45ea334d7be28bfa7bbce132985612f0e6ecd10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262820
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36916}
Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36911}
RTCPReceiver::main_ssrc_ and local_media_ssrc() represent the same
value but could get out of sync when `set_media_ssrc()` was called.
Instead of using main_ssrc_, just use the local_media_ssrc() accessor.
Bug: webrtc:11993
Change-Id: I2b034287e6b6025d9b0d2affa391a168896a614b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262663
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36905}
A packet arrival history is used to store the timing of incoming packets and tracks the earliest and latest packets by taking the difference between rtp timestamp and arrival time. The history is windowed to 2 seconds by default. The packet arrival history will replace the relative arrival delay tracker in a follow up cl.
The playout delay is estimated by taking the difference between the current playout timestamp and the earliest packet arrival in the history. This method works better when DTX is used compared to the buffer level filter that it replaces.
The threshold for acceleration is changed to be the maximum of the target delay and the maximum packet arrival delay in the history. This prevents any acceleration immediately after an underrun and gives some time to adapt the target delay to new network conditions.
The logic when to decode the next packet after a packet loss is also changed to do concealment for the full loss duration unless the delay is too high.
The new mode is default disabled and can be enabled using a field trial.
Bug: webrtc:13322,webrtc:13966
Change-Id: Idfa0020584591261475b9ca350cc7c6531de9911
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259820
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36899}
VP9 automaticResizeOn is disabled if more than one spatial layer is configured via scalability mode.
Bug: webrtc:13960
Change-Id: I7c6351bca6d2f32bcc7391894e8dcc9e74ca2050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261315
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36897}
