Per the docs, the caller is responsible for freeing the memory.
Bug: chromium:1441804
Change-Id: I9aaae493a1a86d8ab4f03930715a643a3c9fb61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304061
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39983}
ReportBlockData class is better documented and has wider usage.
Bug: webrtc:13757
Change-Id: Ie5f2275f2f0236267172e6dd1ce5c2dfb2193ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304101
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39980}
These accessors would allow to deprecated report_block() accessor and
then would allow to remove redundant RTCPReportBlock and ReportBlock types converging on single
ReportBlockData type to pass that information across WebRTC components
helpers like fraction_lost() and jitter() would also allow to unify conversion of the rtp specific format into more common way of represent such information
Bug: None
Change-Id: I3c97f96affcf83b529095899bd63af007f8b4014
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303880
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39975}
When testing manually with gstreamer and v4l2loopback, the incoming
buffer is often larger than the expected size. This change allows
such frames, while still logging the error.
Bug: webrtc:14830
Change-Id: I399aa55af6437d75b50830166a667547f6d144d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291530
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39972}
Improves the capture latency by reducing the amount of
copying needed from the frame. We keep track of the
damaged region of previous frame and union it with
the damaged region of this frame and only copy this
union of the frame over. X11 capturer already has
such synchronization in place.
The change is beneficial especially when there are
small changes on the screen (e.g. clock ticking).
For a 4k screen with 128 cores, I observed the
capture latencies drop from 5 - 8 ms to 0 ms when the
system is left idle. This is in line with the X11
capturer.
Bug: chromium:1291247
Change-Id: Iffb441f9e1902d2658031f5f35b5372ee8e94073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299720
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39968}
This allows encoded frames to be written to any encoded insertable
streams writer without needing to somehow set valid RTP sequence
numbers. Assumes streams are using the Dependency Descriptor header ext.
A short term fix while we discuss whether we can remove the sequence
number check in RtpFrameReferenceFinder::ManageFrame.
Bug: chromium:1439799
Change-Id: I3c1d83793cd8b6cae2a8ad2129b3b6daab1d11c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302301
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39966}
Instead of crashing with a CHECK fail when an insertable stream of a
Video RTPSender is given a frame from an RTPReceiver's insertable
stream, construct a reasonable analogous sender frame and pass it
through to be decoded.
A small step towards removing the split we have between Sender and
Receiver implementations of TransformableFrameInterface which just
confuses users of the API.
Counterpart to https://webrtc-review.googlesource.com/c/src/+/301181 in
the opposite direction.
Bug: chromium:1250638
Change-Id: If66da7d553f14979ff1c5b4e00bff715f58cfce0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303480
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Palak Agarwal <agpalak@google.com>
Cr-Commit-Position: refs/heads/main@{#39963}
Can be used to calculate the average delayed packet outage duration and
number of packet loss events by subtracting from concealment events.
Only used in simulations currently.
Bug: None
Change-Id: I03740a2bcb781af09e28a4d13d9e41c0f84bc506
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303600
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39957}
This should not be relevant anymore and is causing some issues due to
SetMinimumDelay events early in the log.
Bug: None
Change-Id: Ib7e3c624608c9bceaed31bd6669db59887d24659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303580
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39956}
This makes the device light turn off when stopped.
Bug: webrtc:15109
Change-Id: I1deecbc2463e2e316e01ff1f061ab6b0313c1aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302200
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39953}
This is in preparation of using the state that SourceTracker manages
for more things than only getContributingSources. Audio levels reported
via getStats(), aren't consistent with levels reported via getCS.
Since more operations will be derived from the ST owned data, moving
the management of it away from the audio thread, reduces the potential
of contention.
Bug: webrtc:14029, webrtc:7517, webrtc:15119
Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39943}
This makes it a bit more clear that values of member variables of
SourceFrame are never directly changed and that doing so is not an
intentional part of the design. Also made use of `SourceFrame` vs
`const SourceFrame` more consistent since the audio frame of a
`const SourceFrame` was being modified in some places.
Accessing the embedded AudioFrame can be done via the const
audio_frame() accessor or via the mutable_audio_frame() accessor when
modifying the frame is needed. This helps with clarifying later on
when downstream code paths such as ones that access the `packet_infos_`
data, can know that it won't be modified for the rest of the frame's
lifetime (thus avoiding having to make copies).
Bug: none
Change-Id: I175cec8fcdb85063239a5f9c299b7878c639f00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302383
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39926}
This reverts commit 2080dacfb7946daf79ecd3f69efbd0c9e08b9be2.
Reason for revert: This CL is causing a lot of flakiness on iOS bots
https://ci.chromium.org/p/webrtc/builders/ci/iOS%20Debug%20%28simulator%29
Original change's description:
> For AV1, disable error resilience on upper temporal layers. Error resilience is no longer required for upper temporal layers. Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
>
> Bug: webrtc:15106
> Change-Id: Id92d51defbd26c1a77e3c9fe19607e9db4a3e7c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302001
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39900}
Bug: webrtc:15106
Change-Id: I24515280113ed6681c9766026ec24d689035c031
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301983
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39903}
Instead of crashing with a CHECK fail when an insertable stream of a
Video RTPReceiver is given a frame from an RTPSender's insertable
stream, construct a reasonable analogous receive frame and pass it
through to be decoded.
A small step towards removing the split we have between Sender and
Receiver implementations of TransformableFrameInterface which just
confuses users of the API.
Bug: chromium:1250638
Change-Id: I02e0f1d9d35c16dc12718927c5200ff7cf4407e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301181
Reviewed-by: Palak Agarwal <agpalak@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39888}
This helper class is no longer used.
Bug: webrtc:14502
Change-Id: I7940de762ebb9a7c6d04927603f249f5b0061051
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39880}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ic3ac439b3dd3492e6c9c85869efa80a6708658ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301521
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39876}
native_test_jni_onload depends on base_jni which depends on modules/audio_processing:api. This requires to include audio_device_java in pure video targets like video_codec_perf_tests.
Bug: webrtc:14852
Change-Id: I5e7b102fd730801562695bf3f4d5170ec8e59b58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301363
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39873}
Ensure probes are not created until after the transport becomes writable even if the network route change.
DTLS negotiation must complete before there is a point in sending probes.
Bug: webrtc:14928
Change-Id: Ib3cb93aef9f38e306b094dd700ed595cf9eb3f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301362
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39870}
This is what Firefox implementation relies on and I can see that also
the V4L2 implementation is doing the same.
Bug: webrtc:15087
Change-Id: I641062ba879b6ef83e31af79ecc9d06fdae54adb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301320
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39869}
The jitterbuffer would call Flush which takes a mutex from
InsertPacket, which is already under the same mutex. Fix
this by introducing an internal flush method that assumes
a locked state.
The change also adds more thread annotations in case more
problems were present. No more problems were detected.
Fixed: b/277930190
Change-Id: If85609f27f8187ade9370847fecc2bc83d940dd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301340
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39868}
* Removes a ~60Hz thread-wakeup signal caused by the FrameArrived event
* Initial power measurements shows a reduced power consumption
* No increase in time to first captured packet found
Bug: chromium:1428592
Change-Id: Ia23b5db0c87e70e5b0d6919394494a24d8944493
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301200
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39861}
This CL removes the usage of the Magnifier screen capture API on
Windows. The idea is to remove the actual source in a second step
once this change lands.
Bug: chromium:1428341
Change-Id: Id2cb25632c7edbea2cf527959b14b27ee00b0e56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301164
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39856}
These tests often fail in 'ExtrapolateLocalTime' because the result gives a negative Timestamp.
Here is the stack from https://chromium-swarm.appspot.com/task?id=6173230e67897b10:
PC: @ 0x7f03afdb8e87 (unknown) raise
...
@ 0x55f4a360ba71 352 webrtc::Timestamp::operator+()
@ 0x55f4a47ecaf3 160 webrtc::TimestampExtrapolator::ExtrapolateLocalTime()
Low-Coverage-Reason: coverage isn't that low.
Change-Id: If3e7cbf31d6c4800727b24352ed2c6edc425fc73
Bug: webrtc:15022
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39853}
More or less bit-exact, only difference is that we don't seek in the
input file before returning silence for DTX packets.
Bug: webrtc:13322
Change-Id: I147b70d4a0f2c78719c9673b55df6617e064bd61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301104
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39851}
This is step to allow migration of Test ADM to the AudioDeviceModuleImpl
as a base class to include AudioDeviceBuffer into SUT.
Also it will allow to remove WaitForRecordingEnd() method from Test
ADM
Bug: b/272350185, webrtc:15081
Change-Id: If2aa43ec0c31f6ad9aab8aa3e36cabc4a7a73c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300862
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39849}
Since PacketRouter now is only used on the worker thread, there is no need for a lock.
Bug: webrtc:14502
Change-Id: I65778f68b7e211d7bc7388a4615888a49ceb5f59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300964
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39848}
Set of codecs for testing is hardcoded to AV1, VP8, VP9, H264, H265. Some codecs may not be available due to lack of support on the platform or due to some issue in our code which would be a regression. Reporting zero metrics for failed tests would allow the perf tool to detect such a regression.
This also enables codec tests by default. The tests should not run on bots since video_codec_perf_tests binary is not included in any test suits yet.
Bug: webrtc:14852
Change-Id: I967160069055036f93e595d328c4d5f1ca483be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300868
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39840}
The idea is to land this in Canary and ask for feedback from users
who can reproduce the issue, solve the issue and then revert this CL.
Example: https://paste.googleplex.com/6080504230051840
Bug: chromium:1421656
Change-Id: Ic214dc341a322470970abeca1794493f45b93843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301080
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39834}
