and ensure there is only one, similar to what is done with RTX.
This avoids exposing a payload type there.
See also
https://github.com/w3c/webrtc-pc/issues/2696
BUG=webrtc:42221750,webrtc:360058654
Change-Id: Id7c2ddeaf47a3169db9be43c9c5b8e59346f1d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376760
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43929}
Follow-up to
https://webrtc-review.googlesource.com/c/src/+/375847
moving to a big outer loop over formats instead of calling the inner loop with single-element arrays.
BUG=webrtc:360058654
Change-Id: I7d263c1014d80f2312bf93595ee8e8ef9c4e7953
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376081
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43896}
This is a reland of commit 1ad3e14e9981772554a848c5034c7c555680aef7
The original CL removes all sending streams since all codec types has
been attempted for encoding and we have no other codecs to fallback to.
However some downstream applications will still attempt to set the codec
preferences even all send streams have been removed.
As a result, we follow previous logic to keep the send streams, to avoid
the regression.
Original change's description:
> Follow codec preference order for sending codec fallback.
>
> When encoder selector is not enabled, currently we always fallback to
> VP8 no matter how the codec preference is setup. Update to follow codec
> preference order for the fallback.
>
> Bug: chromium:378566918
> Change-Id: Ia3fbfc9d407683ef7b3d6246af7e9ec58535dc89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370707
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43566}
Bug: chromium:378566918, b/384725754
Change-Id: Ifd48b30b80ae51c3ede9391ed62e8ce408864aa0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374852
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43874}
This CL fixes two issues with the old way targetBitrate was reported:
1. The target is per encoder, i.e. per SSRC, but the old way to report
it was per sender and was approximately the sum of all encodings'
targetBitrate in most cases.
2. The old value did not come directly from the VideoBitrateAllocation
and tended to be greater than the sum of all targets (don't know
why).
We know the old value was wrong and the new value correct because
the actual bytes produced by the encoder closely matches the configured
target, which wasn't always the case with the old metric implementation.
Tested with unit tests and manually in Chrome by going to
https://henbos.github.io/codec-quality/src/index.html and ensuring
target ~= actual bytes produced. It also matches the debug logging of
video_stream_encoder.cc.
Bug: webrtc:42225524, chromium:392424845
Change-Id: I7a6f69e053ebc3fd972c2c4b7712750e721c0acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376460
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43854}
and make it less opus-specific.
BUG=None
Change-Id: I6fe2975ba6e45a3758fedc5b950de90e8d9df362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375436
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43846}
Adds test coverage for the following mitigations (need both to pass):
1. https://webrtc-review.googlesource.com/c/src/+/375847
2. https://webrtc-review.googlesource.com/c/src/+/376022
Like the comment says, this is neither mandated by the spec or
guaranteed, but when the number of codecs is quite small like it is in
this test it will be true for all RTX codecs.
Bug: webrtc:360058654
Change-Id: Ib73cea59d06a62390dd039eb2dc04677d6178460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375865
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43841}
Changes the order of payload type assignment to keep the rtx_pt =
primary_pt+1 pattern (even if not required by the specification)
On https://webrtc.github.io/samples/src/content/peerconnection/pc1/ this
changes PTs as follows:
M132: m=video 9 UDP/TLS/RTP/SAVPF 96 97 102 103
104 105 106 107 108 109 127 125 39 40 45 46 98 99 100 101 112 113 114
(121 more lines) mid=1
M134: m=video 9 UDP/TLS/RTP/SAVPF 96 109 99 118 100 119 101 120 103 121
104 122 37 123 40 127 97 114 98 115 107 44 108 (121 more lines) mid=1
M134 with this patch: m=video 9 UDP/TLS/RTP/SAVPF 96 97 107 108 109 114
115 116 117 118 119 120 37 121 40 124 98 99 100 101 127 42 43 (121 more
lines) mid=1
Note that this pushes red and ulpfec into lower range but those codecs
are not widely used and it is possible to get them back into the upper
range e.g. by using setCodecPreferences to disable H264 (where ulpfec is
not supported)
BUG=chromium:391132280
Change-Id: I892f00a2f276728d16c37e8ba5f76d01f6322a7d
No-Iwyu: single include missing, keep it more merge-able
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375847
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43833}
Prior to this CL, IsSameRtpCodecIgnoringLevel() only ignored level IDs
if the codec was H265, incorrectly considering, for example, different
levels of H264 Baseline as not equal.
- This CL fixes that problem by using IsSameCodecSpecific() which is
already used in other places, reducing the risk of different
comparisons using different comparison rules.
This also fixes https://crbug.com/webrtc/391340599 where
setParameters() would throw if unrecognized SDP FMTP parameters were
added to a codec as part of SDP negotiation via SDP munging.
This CL makes the following WPT tests pass:
- external/wpt/webrtc/protocol/h264-unidirectional-codec-offer.https.html
- fast/peerconnection/RTCRtpSender-setParameters.html
Bug: chromium:381407888, webrtc:391340599
Change-Id: I5991403b56c86ba97e670996c6687f6315dde304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43797}
This is a reland of commit bcb19c00ba8ab1788ba3c08f28ee1b23e0cc77b9
Original change's description:
> Allow sending to separate payload types for each simulcast index.
>
> This change is for mixed-codec simulcast.
>
> By obtaining the payload type via RtpConfig::GetStreamConfig(),
> the correct payload type can be retrieved regardless of whether
> RtpConfig::stream_configs is initialized or not.
>
> Bug: webrtc:362277533
> Change-Id: I6b2a1ae66356b20a832565ce6729c3ce9e73a161
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364760
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43197}
Bug: webrtc:362277533
Change-Id: Ia82c3390cceb9f68315c2fd9ba5114693669af32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374780
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43787}
In order to reduce the size and scope of a follow-up CL, this CL makes
some cleaning up and improvements to existing tests and adds some minor
test utility methods that will be used in the follow-up.
No change in behavior, this CL...
- Makes use of NiceMock in RtpTransceiver tests to avoid wall of text
spam for various "uninteresting" method calls in all tests in this
file.
- Refactors creating senders, receivers and transceivers to allow the
follow-up CL to create such objects for kind "video" as well.
- Exposes cricket::FakeVideoEngine* to RtpTranscieverTest and allows
adding unidirectional video codecs in the fake engine, to be used by
the follow-up CL's tests.
- Allows creating fake video engine codecs from SdpVideoFormat in the
fake decoder factory (already possible in the fake encoder factory).
Bug: chromium:381407888
Change-Id: Ie07eff79d832dd21800b95fd584891ebf4520798
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43776}
In order to align with this PR[1], setParameters() should not throw if
the H265 level ID we're trying to send does not match what was
negotiated. This was believed to be fixed by [2] but we were still
throwing due to a check on a different layer (media_engine.cc).
In order to reproduce the issue despite WebRTC lacking SW
encoder/decoder for H265, peer_connection_encodings_integrationtest.cc
gets a new test with real stack but fake encoder/decoder factory. This
allows negotiating H265 and doing SetParameters() even though the codec
is not processing any frames.
- Basic test coverage is added for singlecast and simulcast H265.
- Test coverage for the bug being fixed added.
- In Chrome the equivalent WPTs exists for when real HW is available
here[3]. Those tests PASS with this CL (currently FAIL).
[1] https://github.com/w3c/webrtc-pc/pull/3023
[2] https://webrtc-review.googlesource.com/c/src/+/368781
[3] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/h265-level-id.https.html
Bug: chromium:381407888
Change-Id: I3619a124586b8b26d3695cfad8890cf40bd475db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43759}
Before this CL VP8 and AV1 used the same max QP=56. Tests show that at this QP AV1 delivers a worse PSNR than VP8. We want AV1 min quality to be not worse than VP8. This CL reduces the default max QP for AV1 to 52. With this value libaom AV1 encoder delivers PSNR close to libvpx VP8 at QP 56.
Bug: webrtc:351644568, b/369540380
Change-Id: I2e27ddab562f9c9710b11dc09076b03d7b308bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374041
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43751}
by storing
[[LastCreatedOffer]] / [[LastCreatedAnswer]]
which are similar to the W3C equivalent but as
description objects instead of serialized SDP strings.
While rejecting all SDP munging is not feasible, this lets us
measure and reject certain modifications gradually.
Chromium metrics CL:
https://chromium-review.googlesource.com/c/chromium/src/+/6089633
This is measured at three points during the lifetime of a peerconnection:
* for the first SLD call
* when the connection is first established
* when the connection was established and is being closed
Note that the "first" SDP munging detected is returned which may hide that something uses more than one modification.
BUG=chromium:40567530
Change-Id: I964e3ee6e75f73b777d90556fac8691a6f3dc27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43741}
With this CL, the decision if an RTP packet should be sent as ect(1) is made in RtpControllerSend depending on if RFC 8888 has been negotiated and if CCFB is received with ECN enabled.
Since webrtc does not yet adapt to ECN feedback, packets are sent as ECT(1) until the first feedback is received.
Change-Id: Iddf63849328afbe54a7c8f921f2e8db134aeff6a
Bug: webrtc:42225697
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367388
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43609}
This was exercised by a test, but multi-codec RED is not currently
generated by WebRTC.
RED spec allows it, so failing in comparator seems wrong.
This was one of the cases where the referenced bug was triggered,
but not the only one.
Bug: webrtc:384756621
Change-Id: I28c101aa34a62083b72b5f7fc12d25fc637db209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43607}
When incoming codec_settings_list size is more than
the internal RTP source indentifiers, then it would
cause an invalid memory acccess.
The fix is to operate the stream config update only
when these sizes are match.
Bug: chromium:378724147
Change-Id: I2195df82e0e05619cd2a9bc2d4cb5e8f3efa1446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43603}
This reverts commit 1ad3e14e9981772554a848c5034c7c555680aef7.
Reason for revert: Breaks downstream project. We are investigating into a potential problem when running on mobile platforms. We will get back with info or reland.
Original change's description:
> Follow codec preference order for sending codec fallback.
>
> When encoder selector is not enabled, currently we always fallback to
> VP8 no matter how the codec preference is setup. Update to follow codec
> preference order for the fallback.
>
> Bug: chromium:378566918
> Change-Id: Ia3fbfc9d407683ef7b3d6246af7e9ec58535dc89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370707
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43566}
Bug: chromium:378566918
Change-Id: I09086b5ad100a8f66e87df167e903d0b5fe5b589
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372080
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43600}
This allows us to verify consistency of codec lists in more places.
Bug: webrtc:360058654
Change-Id: Ibd0d10579c4b8058031db0df458e8fc9e2181152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371921
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43594}
Lists of codecs have a lot of cross references (RTX/APT and the like).
We should introduce functionality to verify that those linkages are correct
before modifying the handling of these.
This CL introduces the CodecList class, which can be extended to do
that verification. It is used by pc/media_session.cc, but inter-module
APIs are not changed in this version (they will be later).
Bug: webrtc:360058654
Change-Id: Ifd6313d0289cfa090e51ac28bc775265d18fe6f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43582}
When encoder selector is not enabled, currently we always fallback to
VP8 no matter how the codec preference is setup. Update to follow codec
preference order for the fallback.
Bug: chromium:378566918
Change-Id: Ia3fbfc9d407683ef7b3d6246af7e9ec58535dc89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370707
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43566}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.
The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.
This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...
b) if a dtls restart happens, delete and restart socket.
Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch
Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
removing the webrtc need for having sources in it.
BUG=webrtc:42226155
Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43540}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
The video engine MapCodecs function returned an empty list of
codecs when errors occured, which caused crashes downstream.
This created issues with diagnosing errors caused by PT redesign.
Bug: webrtc:360058654
Change-Id: I0b5bdc9f95814ac4cfb99f749075990c3077e7a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370420
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43499}
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43490}
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
For-loop has a 'continue' statement that skips increment of the index.
Added such an increment before 'continue' for the index to keep up with
the for-loop.
I guess current implementation will bug on codecs sequence like:
'red, unknown, opus'
since the subsequent for-loop (the 'red_codec' one) will not be able to
find 'opus'.
Seems like adding second increment statement is the easiest way to fix it.
Bug: None
Change-Id: Iab9cc66cf569458af9fd9ba5b938d83186c78c73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369700
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43488}
also fix a few TODOs
Bug: None
Change-Id: I2d287ed1a58f71ef32d5dc5624879ae8c63348b5
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370122
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43485}
These methods previously had a default implementation that triggered
a crash. All implementations must now return a valid object, which
simplifies the code that calls them.
Bug: webrtc:13931
Change-Id: I877fbc929b58c6b83767c6ac5a81c8aa942e3fef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369021
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43453}
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).
Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
According to latest requirement, when the level reported by
RtpSender.getCapabilities() for H.265 is different from that was
negotiated, we should not throw when setParameters() is called with
level-id set to that reported by RtpSender.getCapabilities().
Underlingly negotiated codec level should remain unchanged.
Bug: chromium:41480904
Change-Id: I28bbdb5f0a0ab0d98315f56c80004601afc91a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43434}
This is preparatory to ensuring that transport-cc gets turned off when
RFC8888 ccfb is negotiated.
Bug: webrtc:378698658
Change-Id: Ie76677bd6aa046701562bbd93d8489858488f863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368543
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43426}
On a sendrecv m-line, the offered level-id represents the maximum that
can be both sent and received; on a sendonly m-line, the offered
level-id represents the maximum that can be sent; on a recvonly m-line,
the offered level-id represents the maximum that can be received.
Also according to RFC 7798 section 5, the highest level indicated by the
answer is either equal to or lower than that in the offer
Bug: chromium:41480904
Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43425}
This is a reland of commit 775639e930f14a619974944594b40c633cc574a3
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43412}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.
This should eliminate the need for the "force" flag in the field trial.
Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
This reverts commit 775639e930f14a619974944594b40c633cc574a3.
Reason for revert: Breaks internal tests.
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
No-Try: true
Change-Id: I5485b1abfd5f586ec187cc57817940aa2efd72af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368200
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43396}
H.265 does not have software fallback, and it may have issue supporting
more than 1 temporal layers on some devices. Set default to L1T1 when
scalability is not configured, or if a scalability mode is reported as
not supported by encoder.
Bug: chromium:41480904
Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43389}
The std::lcm and std::gcd functions are part of the C++ standard
library. The existing functions are marked as deprecated rather than
deleted in the case of possible third party uses.
#rtc_cleanup
Bug: webrtc:377205743
Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43368}
Inactive encoders are included in the string when they are paused due to
bitrate allocation being 0 for that simulcast layer.
#rtc_ktlo
Bug: webrtc:376804631
Change-Id: I4234b452b60fee58981907380df41962fda5bf40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43367}
