Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
This is a step towards getting rid of reconfiguration via tearing down
and reconstructing receive streams when parts of the configuration
change at runtime.
Bug: webrtc:11993
Change-Id: I337e523f17805b75826ddbd75bd3d0eb6e910bd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269250
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37680}
clangd ignores ASSERT_EXCLUSIVE_LOCK macro attached to an inline function in header, thus IDEs relying on clangd issue false positive warnings about members acceesses without the check of the current sequence.
Attaching assert attribute to an inlined lambda function seems to solve that issue
Bug: None
Change-Id: I6199fee26061aa4223f2e3ea7b7b14bb5820c0bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270480
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37678}
This renames the tests to also capture the expected outcome of the test
along with some minor code cleanups. Some tests have also been added or
extended to tests more invariants.
Bug: None
Change-Id: I0bc733026118eb90646929b164bfc148665556a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267169
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37673}
The goal of this CL is to create a new LogSink::OnLogMessage API which
propagates the source location of the log to the log sinks.
Bug: b/238157120
Change-Id: I5a12bf80fd9c5569ed7aa1ef9185eee58830b19f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269249
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37672}
...for payload type changes and avoid recreating the main video stream.
Bug: webrtc:11993
Change-Id: I03e44ff25d88caeb082a2e44b2e802d3b9392feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269244
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37666}
This CL propose a new API for video dumps in PCLF also removing
differences between p2p and multipeer usage of API.
Bug: b/240540206
Change-Id: Id4d32cc98250500949b3f9e2cf2e86c4bdce7efb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37665}
If a "normal" software buffer frame is dropped during paused state, we
store it as a pending frame and try encoding it after the pause state is
lifted. However, native frames are dropped entirely since keeping e.g.
texture handles for long time periods can lead to side effects.
Work around this by requesting a refresh frame after unpausing if the
dropped frame flag is set.
Bug: webrtc:14276
Change-Id: I9edd1e99454e082bcfe29f3d9041026dd8a390d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270220
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37660}
Some of the timestamps input into UpdateCurrentDelay are not truncated
to milliseconds and thus a small negative delay can result. This means
the delay will not update when it should have.
Bug: webrtc:14168
Change-Id: I5293339b6a39201c680854e9596b717025ee8dc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266370
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37657}
Replace mock implmentation with manual noop implementaion.
libvpx interface is called a lot, and mock implementation of it adds
noticable overhead.
Bug: chromium:1281020
Change-Id: I7fe5cbfd08d5056a14d75e009acff368700c26a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269214
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37656}
This replaces the former WriteFatalLogAndAbort function with the two new
WriteFatalLog functions that're already submitted as overrides in
Chromium.
The default implementations of these are not defined under
WEBRTC_CHROMIUM_BUILD.
Bug: chromium:1216177
Change-Id: I207e1f96f14094d742a51849f4fa6b4f1022333e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Boström <pbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37652}
This adds a file,line version of this function (not yet committed) as
Chromium logging uses LogMessage(file, line, severity) and needs this
information to give better logs.
The two versions of this method will be implemented in webrtc_overrides/
and then committed to Chromium. At this point checks.cc will move its
anonymous-namespace version of this function (and be renamed) to
match this definition, but only define it when not building with
Chromium.
At this point WriteFatalLog will be using LogMessage(LOG_FATAL) to crash
in Chromium allowing us to upload better crash dumps and stacks to crash
reporting.
Bug: chromium:1216177
Change-Id: I3fd6a84cdfbb2552a5e628d46257bd7a00c9e6dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269288
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Peter Boström <pbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37646}
This reduces the number of times we recreate video receive streams
and prepares for not having to do that for flexfec streams and avoid
having to recreate a video receive stream when flexfec config changes.
Bug: webrtc:11993
Change-Id: I11134b6a72eb162623ebbc12521d409da8616107
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261941
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37641}
This removes the unused field trials
`WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` and
`WebRTC-SimulcastScreenshareUpswitchHysteresisPercent` as well as the
`video_hysteresis` and `screenshare_hysteresis` parameters in
`WebRTC-VideoRateControl`.
The hysteresis parameters in `WebRTC-StableTargetRate` are currently
left, their future is unclear...
Bug: webrtc:9734
Change-Id: I9e6bbe4b630a0501d365bf69e87e65164c500122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269207
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37635}
From when callTest's send_transport_ is deleted and until the test is
completely ended, there is a possibility that the background task
webrtc::ModuleRtpRtcpImpl2::MaybeSendRtcpAtOrAfterTimestamp
will call send_transport_ which has already been deleted.
Fix this by deleting rtp_rtcp_ before send_transport_ is deleted.
Bug: webrtc:14202
Change-Id: Ief96c4712875beb55ef232a8bce990d1e9e9efe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266300
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37633}
