When we offset the measured inter-arrival time due to packet loss, it will sometimes be less than zero. This is the correct value to use when calculating the relative packet arrival delay.
Bug: webrtc:10333
Change-Id: I14a68563a379fa0b9444684304362503a6f1bfca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127547
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27164}
Current implementation of loss based controller has a sensitive filter.
This CL increases the moderate loss rate to ensure robustness to small
changes in network behavior.
Bug: webrtc:10365
Change-Id: I0dcb5ba45904d8dda4c78b39bd13619523bc90ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127901
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27160}
Add a feature (gated by field trial) that stores
packets with unknown ssrc in a circular buffer
and replays them once a receive stream with matching
ssrc is created.
This improves situation where media is incoming
but signaling or SetFrameDecryptor is slow.
BUG=webrtc:10405
Change-Id: I7c7b2f4bd96c942c09e96db0cdae4ce5efef2541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127543
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27159}
The stacktrace unit test was flaking on arm32; my theory is that this
happened when the thread whose stack we were dumping was doing a
system call inside `params->deadlock_start_event.Set();` in
ThreadFunction(). (This would be a problem because, according to the
comment at the bottom of the file, "stack traces originating from
kernel space do not include user space stack traces for ARM32.")
Attempt to solve this problem by spinning on an atomic flag instead,
since this involve no system calls. And add a short sleep to the main
thread, to give the other thread time to get from the barrier to the
thing it's actually supposed to deadlock on.
Bug: webrtc:10420
Change-Id: I4c6392157c8a06c64cb11146ffe9368e5bade6fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128340
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27158}
Reduce resolution of smoke test in PC E2E test framework to reduce load
on bots, cause this test isn't part of performance test binary.
Bug: webrtc:10138
Change-Id: I2c3758583c03e75be17bfef799a31f63357834c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128380
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27157}
This change integrates fuzzing support for RtpDumps in WebRTC. This allows
LibFuzzer to directly fuzz the RTP code path from packet arrival all the way
to actual decoding and rendering. It does this by replaying each RTP packet
in the RTPDump which can be mutated directly by the fuzzer.
For fuzzing support the RtpFileReader needs to support reading from a
buffer instead of an file. The test class requires FILE* for all its
parsing operations and is deeply coupled this way. I chose to solve this
problem at an OS level by using the tmpfile() option and copying the buffer
to the tmpfile(). fmemopen() is no available on most platforms so couldn't
be used as a generic solution. The additional copy isn't ideal but won't
be a bottleneck for the fuzzing.
In the future I plan for the fuzzers to read from a configuration file. But
given the current packaging strategy for fuzzers in WebRTC this isn't easy.
Bug: webrtc:9860
Change-Id: I2560120e82663f9e9fb5b9640e6a6d16f9c1a360
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126682
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27151}
It's used for driving the old jitter buffer, which is used only when
vcm::VideoReceiver is used via the legacy VideoCodingModule api.
Bug: webrtc:7408
Change-Id: I179d5b26e112d9f94615d2e1b410b51a657aa05b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127294
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27147}
This reverts commit a594ef089370b8073ca9dc5a6b6bf4be9a58a313.
Reason for revert: This log is triggered more than 10,000 times per run, spamming the log output to the extent that tests start failing with EXCESSIVE_OUTPUT.
The tests are chromium.webrtc.fyi tests:
* WebRtcStressResolutionSwitchBrowserTest.MANUAL_SurvivesPeerConnectionResolutionSwitching
* WebRtcStressPauseBrowserTest.MANUAL_SurvivesPeerConnectionVideoPausePlaying
on linux, win, and mac.
Example run: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/2556
Original change's description:
> Log an error if the RTT is negative
>
> Bug: webrtc:10407
> Change-Id: I5479cb2b7163c6e9e58854f4ffa7976b3d606da5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127568
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27122}
TBR=srte@webrtc.org,eshr@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10407
Change-Id: Ida2572b722b92bae4893d4567597dd21d1df54b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128120
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27144}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I1b46653b91bce012afabfa0f2d249718e6de2df8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127626
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27139}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: If995d9d9d21534d3c66a1e7c1fc1c62569766f40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127627
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27138}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I41947a24764840ad14b2bcccd99d3212d79c1485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127628
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27137}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I2ea59dc66230182bee6ae7a0925aed0fe9ef823c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127643
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27133}
This CL removes parameters for AEC3 which are no longer used. To reflect
that change, one of the parameters also is renamed
Bug: chromium:941949,webrtc:8671
Change-Id: I26609b396fa14ecb7523eebfe531a1338718103b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127780
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27128}
The video decoder thread is the pilot user.
For now this is an Android-only feature, since that's the only
platform we can print stack traces on.
Bug: webrtc:9987
Change-Id: Ie638c619673b5f159d91a32683fd787baf46479a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126222
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27127}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I08ff36bd689fa7c3716c8e7017bd571cc9f09f35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27125}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: Iedb3be944828a1caba55bbbd4dc0b56c55bbb7d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127624
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27123}
If the repeated task stopped itself, the return value does not matter,
therefore we don't have to check it.
Bug: webrtc:10365
Change-Id: I4a2c7a40a287dd0a8099628e228e9c319409576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127545
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27121}
This change simply calls through all code paths in the SSLCertificate interface
after passing in an untrusted PEM string. Corpus will follow in another CL.
Bug: webrtc:10395
Change-Id: I001642fa89a84ce01505780f5e76f01a0e46a785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127640
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27118}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I20190c01559ff315422be1b3f980853cbc5afbcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127625
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27117}
Adding an example of a request to send simulcast (from the PC).
Adding an example of a request to receive simulcast (from the SFU).
Bug: webrtc:10409
Change-Id: I13b689621e2f89f8e00b7ee8bc542157ccebb873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127621
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27116}
rtp_header_parser currently has 0% fuzzing coverage. To improve this I have
added a basic fuzzer which fuzzes all of the available paths.
Bug: webrtc:10395
Change-Id: I30324b2bfa7629b0110527258b33b7e048e89fcf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27115}
