Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
PacketBuffer takes RtpVideoHeader struct as an input that is complicated
and hard to fuzz. Current PacketBuffer doesn't fuzz it and thus has very
low coverage.
RtpVideoFrameAssembler uses PacketBuffer underneath and takes as input
almost raw rtp packet and thus easier to fuzz and better match production input
Bug: webrtc:7408
Change-Id: I00394c35e002a667760eed477f11ac7898f7eacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290574
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39013}
making it clear what unit is being used.
BUG=webrtc:13756
Change-Id: I6354d35a8e02bb93a905ccf32cb0b294b4813e41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289460
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39008}
This is a reland of commit 97ba853295578975a04fc504315cccd465f9f0bd
This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502.
Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}
Bug: webrtc:14802
Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38997}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
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}
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}
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}
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 helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
Follow-up of https://webrtc-review.googlesource.com/c/src/+/287221,
instead of asking GTest for the test suite and the test name, let's
propagate the test case passed by the user of PCLF.
Bug: b/237982523, webrtc:14757
Change-Id: Ia2a6ed4781f8c53c25b0006b8c7483e08ecead26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287126
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38867}
This metadata key is temporary, as explained in bugs.webrtc.org/14757,
this information will be at some point directly accessible via the
webrtc.test_metrics.Metric.test_case field.
Bug: b/237982523, webrtc:14757
Change-Id: Ie77875a33db5961f8a5572bd1b7066ad8ba17291
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287221
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38858}
This MidCallProbingRampupTriggeredByUpdatedBitrateConstraints blocks https://webrtc-review.googlesource.com/c/src/+/285740 submitting. I was able to complete the test locally, but cannot manage to do so remotely.
Bug: none
Change-Id: I75979af25552b4a31487a26e40857a713299e0eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287022
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Diep Bui <diepbp@google.com>
Cr-Commit-Position: refs/heads/main@{#38848}
Purposes of this refactoring:
1. Add functionality for reading a specified frame.
2. Change resolution and frame rate on per-frame basis.
Both features are needed for https://webrtc-review.googlesource.com/c/src/+/283525
Bug: b/261160916
Change-Id: I6d60e62dbc3913c43b5c1b491690f5cb4a8632dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285483
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38829}
Default send transport wide sequence numbers on audio
Use 32kbit/s audio.
Pace in bursts 40ms, See chromium:1354491
Bug: none
Change-Id: I40b1305ce71478749723a53f6cc84669ddf930e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285883
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38827}
Differently from the ChromePerfDashboardMetricsExporter, this new flag
doesn't default to storing the output file to NSDocumentDirectory (and
with a default name, for example perftest-output.pb) but instead
just stores the file at the location specified by --webrtc_test_metrics_output_path.
Bug: b/237982523
Change-Id: Ibb504fdbc94ca5179f4b3da5b06d8cea82140140
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286280
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38807}
This reverts commit 75170be4acc90fece7c65f1a5b9bef03a5cc3880.
Reason for revert: Perf regression not affecting open source.
Original change's description:
> Revert "Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream"
>
> This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc.
>
> Reason for revert: Tentative revert due to possible perf regression. b/260123362
>
> Original change's description:
> > Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream
> >
> > VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
> > Therefore this cl:
> > - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
> > - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
> > - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
> >
> > Bug: none
> > Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38698}
>
> Bug: none
> Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38725}
Bug: b/260400659
Change-Id: Ie8e545edcad85284a7d612183a8e4201672d0b5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285900
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38794}
macOS has deprecated OpenGL as of macOS 10.14. Chromium is moving to
using Metal more and more, but we're going to be forced to keep using
OpenGL, so explicitly silence the OpenGL deprecation warnings.
Bug: chromium:1393687
Change-Id: I668e8d9bf57669f715f341f940ea12f3293faa9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Avi Drissman <avi@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38771}
By declaring and defining the flag in a separate and reusable library
it can be used by other main() implementations as well.
Bug: b/237982523
Change-Id: Ia5445ee6e85bc1d536bee2ddd842439f8832116b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285480
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38765}
* Add '--quick' argument to 'low_bandwidth_audio_test' even though it doesn't look like it makes much timing difference.
* Add sharding for 'svc_tests' and 'video_engine_tests'.
Change-Id: I6e3357954d18ad03ea9f62912dd77e0e1a74b97d
Bug: webrtc:14713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285100
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38748}
This reverts commit c48a2653466f926ecc1d259a16ede304333fc14b.
Reason for revert: logcat issue should be fixed with https://crrev.com/c/4055461.
Original change's description:
> Do not log on stderr on Android tests.
>
> On Pixel 2, this causes an increase in flakiness. This needs to be
> reenabled once the root cause is fixed.
>
> Bug: chromium:1384172, b/259113795
> Change-Id: Ie94d3e2daad3a2de5af673c763362ea1b42fde7d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283522
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38623}
Bug: chromium:1384172, b/259113795
Change-Id: Iadd7c484f4e73deea952df7980acc0164c96a592
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38731}
This reverts commit d8c4de71722c9de38f942932be21d4015f32a3bc.
Reason for revert: Tentative revert due to possible perf regression. b/260123362
Original change's description:
> Remame VideoSendStream::UpdateActiveSimulcastLayers to StartPerRtpStream
>
> VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
> Therefore this cl:
> - Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
> - Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
> - RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
>
> Bug: none
> Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38698}
Bug: none
Change-Id: I4f0d27679e51361b9ec54d2ae8e4d972527875d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284940
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38725}
The expected behavior is to have something similar than python:
https://docs.python.org/dev/library/argparse.html#dest:
"Any internal - characters will be converted to _ characters to make sure the string is a valid attribute name".
This allows to catch chromium arguments like 'isolated-script-test-output' that previously needed some preprocessing done for example in flags_compatibility.py.
This CL also fixes a fuchsia specific issue where the test runner needs a 'isolated-script-test-output' argument but then pass the argument to WebRTC that expects a 'isolated_script_test_output' argument. Thus calling flags_compatibility before the test_runner fails and there is not much room to change the argument in between the test runner and the test.
Change-Id: I48a591743fa50484a0ec584a3f9e97d9e0fd25ef
Bug: webrtc:14694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284541
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38707}
VideoSendStreamImpl::Start and VideoSendStream::Start are not used by PeerConnections, only StartPerRtpStream.
Therefore this cl:
- Change implementation of VideoSendStream::Start to use VideoSendStream::StartPerRtpStream. VideoSendstream::Start is kept for convenience.
- Remove VideoSendStreamImpl::Start() since it was only used by tests that use call and is confusing.
- RtpVideoSender::SetActive is removed/changed to RtpVideoSender::Stop(). For normal operations RtpVideoSender::SetActiveModules is used.
Bug: none
Change-Id: I43b153250b07c02fe63c84e3c4cec18d4ec0d47a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283660
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38698}
