rtc::CopyOnWriteBuffer::SetSize extends buffer with uninitialized memory by design.
It is up to the user of the rtc::CopyOnWriteBuffer to ensure it is initialized.
Bug: chromium:1403397
Change-Id: Ic0111a84bda32379770ddb1c7d24bee10d96b7a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289041
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38959}
preparing to put them at session level when max-bundle is set.
Drive-by: move m= serialization to helper.
BUG=None
Change-Id: I04d918ee8eb70c0cc40baf8ebc12054c6b3a2a15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288820
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38950}
which needs to be taken into account separately if the
primary SSRC has been acknowledged but the RTX SSRC has
not.
If nothing has been acknowledged, mid+rid are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If the primary SSRC has been acknowledged, no extensions are sent on the primary SSRC and mid+rrid are sent on the RTX SSRC.
If both the primary SSRC and the RTX SSRC have been ack'd, no extensions are sent on either primary or RTX SSRC.
BUG=webrtc:13896
Change-Id: Ice251fae23a881ee9c9edc71b5d5c45a32ac76d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288980
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38949}
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet.
The purpose is to be able to remove extension information from webrtc/call/receive_stream.h.
Header extensions are negotiated per mid, not per receive stream.
The goal is to reduce the number of places where packets are parsed and demuxed.
Bug: webrtc:7135, webrtc:14795
Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38944}
UnitBase own function to divide with rounding overflows when dividend is close to the max int64_t
Bug: b/262999013
Change-Id: I5b0c3f4408165a0f03690cab80bd098e506fc984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38943}
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
For each stream in streams, add stream.id to
[[AssociatedMediaStreamIds]] if it's not already there
This applies to addTrack and setStreams and the set of streams in
addTransceiver.
Tests that default to the stream id as sync group add
"-sync" as a postfix
BUG=webrtc:14769
Change-Id: I806d2fd87a98d50e54709755541f3f1efff1d8ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288701
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38942}
This reverts commit 315b95ca11161bdea715d5316f92828edd41f0d5.
Reason for revert: Breaks internal bots.
Original change's description:
> Enforce stream id uniqueness in RtpSender::set_stream_ids
>
> https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
> has a step saying
> For each stream in streams, add stream.id to
> [[AssociatedMediaStreamIds]] if it's not already there
>
> This applies to addTrack and setStreams and the set of streams in
> addTransceiver.
>
> BUG=webrtc:14769
>
> Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38937}
Bug: webrtc:14769
Change-Id: I6fd22ff0550c0894057fb1dc15f1b95819fa6df2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288744
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38940}
The STUN message about mismatched passwords is not interesting.
Only emit it in VERBOSE mode when DCHECK is on.
Bug: webrtc:14578
Change-Id: Ie83080d88be6da24e7f2f79d7eb279087f84c2a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288740
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38938}
https://w3c.github.io/webrtc-pc/#dfn-create-an-rtcrtpsender
has a step saying
For each stream in streams, add stream.id to
[[AssociatedMediaStreamIds]] if it's not already there
This applies to addTrack and setStreams and the set of streams in
addTransceiver.
BUG=webrtc:14769
Change-Id: If6be813396a1987dfe49fd73f976f96c71459eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287864
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38937}
This CL removes a couple more opportunities for client code
to interact directly with the MediaChannel implementation classes.
No-try because of infra failure.
No-Try: true
Bug: webrtc:13931
Change-Id: I658b8b04eff11de7831a1933d16d40fc59c3f0fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38935}
This CL records the time it took to capture a frame.
Bug: chromium:1291247
Change-Id: I31cbb2ca6ae5b9449b8fd154182105a3ce2c851e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288660
Commit-Queue: Salman Malik <salmanmalik@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Salman Malik <salmanmalik@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38933}
- Test behavior with no input volume controller
- Test behavior with startup volume higher than the minimum
input volume
Bug: webrtc:7494
Change-Id: I36d48e2bd277b8a71eb6fbb0272c26c7176b3d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286380
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38932}
See commit
https://webrtc.googlesource.com/src/+/c8a6fb2bb8762de17008dee97c5fb6e762f7e056
where the setup methods for RTCCameraVideoCaptureTests' test cases were
lost. Both "setup" where XCTest instead looks for "setUp", and
"setupWithMockedCaptureSession" which isn't called explicitly anywhere.
This commit splits the old RTCCameraVideoCaptureTests into two;
RTCCameraVideoCaptureTests for tests using "setup", and
RTCCameraVideoCaptureTestsWithMockedCaptureSession for tests using
"setupWithMockedCaptureSession".
Bug: webrtc:8382
Change-Id: I64cefff744e12f62d65e04133512de1e10d17d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288601
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38931}
Make sure that the input volume controller implementations exhibit
the adaptive behavior regardless of the sample rate and the number
of channels. The newly added tests check that:
- a downward adjustment takes place with clipping input
- an upward adjustment takes place with a too low speech level
- a downward adjustment takes place with a too high speech level
Bug: webrtc:14761
Change-Id: I1795e74c5f219e15107e928ebaca2bfa75214526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287301
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38930}
This is a reland of commit d49d49ad89e67d1a3c63fbc638af445af5648875
Fixed seconds to milliseconds conversion in VideoAnalyzer.
Original change's description:
> Report total and squared inter frame delays measured in OnRenderedFrame
>
> After https://webrtc-review.googlesource.com/c/src/+/160042 we ended up with two sets of metrics representing total and total squared inter frame delays: old is measured in OnDecodedFrame and new in OnRenderedFrame. Reporting of old metrics was unshipped in https://webrtc-review.googlesource.com/c/src/+/278100. The metrics are used for calculation of harmonic frame rate and are desired to be measured as close as possible to rendering. This CL removes calculation of inter frame delay metrics from OnDecodedFrame and reports the metrics calculated in OnRenderedFrame to the stats.
>
> Bug: webrtc:11108, b/261512902
> Change-Id: Ia21b321aab3a1ac0b6136dc0df7d95f2f0fd24c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286842
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38909}
Bug: webrtc:11108, webrtc:14792, b/261512902
Change-Id: Ic5d0bc4622ee0cb46b6c225cdddccc217200e794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288641
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38929}
srte@ is the only owner and is not very active....
Bug: none
Change-Id: I4fbedae4fe34765ebf1befbd37dbb98770dce91d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38924}
This was done in crbug.com/webrtc/4559 since "CELT-only mode does not have DTX", but that should not be the case anymore (support was added in Opus v1.2.1).
One exception where DTX does not work is with OPUS_APPLICATION_AUDIO (used with stereo) and low complexity settings. This should not be a common config.
Bug: None
Change-Id: I1476083b836bcabeb73df83d5bf06c3878146d28
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288420
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38923}
This reverts commit d49d49ad89e67d1a3c63fbc638af445af5648875.
Reason for revert:
# Check failed: total_freezes_duration_ms_double <= total_frames_duration_ms_double (196 vs. 0.044783)
https://ci.chromium.org/p/webrtc/builders/perf/Perf%20Mac%20M1%20Arm64%2012
it also breaks the metric 'freeze_duration_ratio':
https://chromeperf.appspot.com/report?sid=6e919d271ff5885c3fa6363dd255b9793d5e79332a9f202b725c33cc7d3da31a
Original change's description:
> Report total and squared inter frame delays measured in OnRenderedFrame
>
> After https://webrtc-review.googlesource.com/c/src/+/160042 we ended up with two sets of metrics representing total and total squared inter frame delays: old is measured in OnDecodedFrame and new in OnRenderedFrame. Reporting of old metrics was unshipped in https://webrtc-review.googlesource.com/c/src/+/278100. The metrics are used for calculation of harmonic frame rate and are desired to be measured as close as possible to rendering. This CL removes calculation of inter frame delay metrics from OnDecodedFrame and reports the metrics calculated in OnRenderedFrame to the stats.
>
> Bug: webrtc:11108, b/261512902
> Change-Id: Ia21b321aab3a1ac0b6136dc0df7d95f2f0fd24c6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286842
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38909}
Bug: webrtc:11108, b/261512902, webrtc:14789
Change-Id: Ie0da33c1071c48c50bff6608830c9e2a5a928fb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288402
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38922}
removes cached pixelbufferpool and instead retrieves current pool from the compressionSession each time (as recommended by apple docs)
Bug: webrtc:14688
Change-Id: I2244e69e7f32b912021db0905b9d5867d0bf6357
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284240
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38920}
This is a reland of commit 6c27d56a2aeb2cff10a216d714552f4970d99d32
avoiding the Fuchsia/C++17 compilation issues.
Original change's description:
> sdp: measure rtp header extension collisions
>
> since extension ids are required to be unique in a BUNDLE group:
> https://www.rfc-editor.org/rfc/rfc8843#name-rtp-header-extensions-consi
>
> Measure how much enforcing this would break in UMA first.
>
> BUG=webrtc:14782
>
> Change-Id: Ieaf7a436feea677032499e11ca14973eebda322e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288362
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38914}
Bug: webrtc:14782
Change-Id: If9449b0381ebe33ba30eff0d733a1f3c94470d5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288383
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38919}
