For an offer in sendrecv direction, if for example it can send H.265
level 5.2 while receiving 6.0, setCodecPreferences on offerer's transceiver will currently remove H.265 from the offer SDP, since currently we do a precise level match on send_recv_codecs with the codecs from setCodecPreferences.
Update the matching logic to ignore the level when matching.
Bug: chromium:41480904
Change-Id: Id0f89cbf117ce62249a99257dcce18b35f407cb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369960
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43477}
Cases above 100 ms have been observed on mac; use 60 seconds as
an offset.
Bug: webrtc:380712819
Change-Id: I52a085cb196472188bb5493276a1b32524717c1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369881
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43473}
Two fixes to deflake,
1. Increase the ramp up time for all layers - short time was flaky for
720p.
2. Wait for both the scalability mode AND implementation name to update.
Sometimes the implementation name would change before the scalability
mode did due to a race, so some OutboundRtpStats would have the wrong
values.
To achieve #2 (and #1 with some debugging) a new utility
WaitForCondition was added in order to apply matchers to a condition.
This is used instead of EXPECT_WAIT_EQ and similar because it gives
clear feedback on failure.
I have made 500 runs without a further flake.
Bug: webrtc:381216372
Change-Id: I0132377774e379857664e9a0c20f432bc9dc9fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369742
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43472}
When an enum has AbslStringify, we log the text coming from stringifying it, not the numeric value.
Drive by changes,
1. Changed the tests to use string matchers rather than
std::string::find.
2. Fixed test includes.
3. Fix spelling.
Bug: webrtc:381502973
Change-Id: I6bab0afda1b20d72c02629e80ff2ac567650183a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369861
Auto-Submit: Evan Shrubsole <eshr@google.com>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43471}
A few tests in dcsctp_socket_test was named DcSctpSocketResendInitTest
instead of DcSctpSocketTest. There is no reason for that.
Bug: None
Change-Id: I845eb0ab6150c4e5d457307e12f056486f86e468
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43470}
The next expected MID to use (which applies to both ordered and
unordered streams, in contrast to SSNs) was properly handed over for
streams this socket sends on, but not for streams this socket receives
on.
Adding handover state first.
Bug: webrtc:41481008
Change-Id: Ib3941f0ee1a34c24605792d9f0b658bb6a9ade4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369821
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43469}
`VideoStreamEncoder` should not recreate the
`FrameInstrumentationGenerator` instace unless the encoder is actually
released. Otherwise it will restart and expect a keyframe the encoder
will likely not produce for a while.
Bug: webrtc:358039777
Change-Id: I111149d5e9b632df9eeb88bbbe8a07969c3e3f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369740
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43468}
Previous computation assumed that local clock is UTC. It isn't.
Adding integration test for abs-capture stats.
Bug: webrtc:380712819
Change-Id: I054d61984cbd017b7ad04ab13e5a687eab89db69
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43465}
~BitstreamReader() DCHECKs that the last read has been verified, so all
paths where we may leave the slice_reader instance's scope early must be
guarded by an Ok().
Bug: None
Change-Id: Ic67f87c04d1f042392c1dd6a066fdccf26e19003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43463}
Return default value T() if no values have been added to the filter.
Together with
https://webrtc-review.googlesource.com/c/src/+/369440, DataRate etc can be used by the filter.
Bug: None
Change-Id: I3d0e1a3e698a91a6197bf434ace2ff8246dc393e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43461}
0 is natural default value for types that can be accumulated
Having default constructor simplify usage of these types in templated code.
Bug: None
Change-Id: If005c69018a2a11011bc789502fdbc600cad3278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43460}
This is a reland of commit 38ddea5ee3320bf3441aeb3654e099b3695c9789
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: I97d1d3d390e6d3de8bf9355b895ec336339d079f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369260
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43454}
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}
For H.265 when scalability mode is not configured for simulcast layers,
the default mode of L1T1 should be assumed instead of L1T3, as that is
the most commonly supported temporal scalability on all devices for
H.265.
Bug: chromium:41480904
Change-Id: Ia9bc91729eb393850dfe5e8fb04280b4f784560d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369080
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43452}
This reverts commit 38ddea5ee3320bf3441aeb3654e099b3695c9789.
Reason for revert: not backwards compatible
Original change's description:
> Add InsertPacket method that takes RtpPacketInfo.
>
> The version which only passes receive_time will be removed (once migrated).
> Keeping the version that only passes header and payload for convenience.
>
> This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
>
> Bug: webrtc:42223109
> Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43445}
Bug: webrtc:42223109
Change-Id: Ie7cf397cfbe5dedca009f16e5e9e3af40adbe99b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369200
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43450}
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}
If max allocated bitrate change, default max limit probe to 2x current
BWE.
Bug: webrtc:369044000, b/370883514
Change-Id: Ibaf79fff94157186002728828d6574bea21afd24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368820
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43448}
The version which only passes receive_time will be removed (once migrated).
Keeping the version that only passes header and payload for convenience.
This will allow us to attach more metadata on the worker thread before InsertPacket, instead of on the playout thread after GetAudio. Eventually, the plan is to split the RTP handling on the worker thread into a separate class.
Bug: webrtc:42223109
Change-Id: I5399b53b9fc5c2f1c996e109054b1b0877ecca05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369000
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43445}
This is a prerequisite for enabling implementation-specific filter
settings for automatic corruption detection.
Bug: webrtc:358039777
Change-Id: I363c592aa35164f690dd4ad1204e90afc0277d8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43443}
This stores usage for all cases, making it easier to discover
abusive usages on unexpected patterns.
Bug: None
Change-Id: I62c9b07498e811ac04c221f57cfbc02312aaaacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43442}
- Add `WebRTC-Audio-OpusDecodeStereoByDefault` field trial
- Behind that field trial, `AudioDecoderOpus::SdpToConfig` uses 2
instead of 1 as default number of channels when the `stereo` codec
param is unspecified
- Instead of wiring up `FieldTrialsView` to `SdpToConfig`, which
requires API changes that break downstream projects, a change in
`AudioDecoderOpus::Config` is made to signal when the number of
channels is forced via SDP config
Bug: webrtc:379996136
Change-Id: If70eb19bc7e3bc74dd0423610cb04ae33ea602fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368860
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43440}
Too big median window will cause errors with large clock drifts, since we'll end up using old values for estimated clock drift.
If the window is too small, the remote clock offset estimation could be noisy or we could even end up using outliers as the offset estimation.
I will not claim that I choose the correct value, and I'm not sure how to measure the quality of the remote clock offset estimations.
Bug: webrtc:379809147
Change-Id: Ib317548d3eec74105d468ef53830e12eb114df7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368580
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43439}
The clock rate is already known by the RTP statistician.
Also included some minor code cleanup.
Bug: b/331602608
Change-Id: I335fa2a1cfd7dcceb286706d295a175a92f6797c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368920
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43436}
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}
In particular, some platforms have a limited pool of frames in the
capturer stack, so we need to avoid stashing raw frames in the frame
instrumentation generator that may be dropped by limiting the size of
the queue and avoid putting anything in there until we know we will
send it to the encoder.
Bug: webrtc:358039777
Change-Id: I054ae53dd5e6ac6a22da39c5049f47788561e77a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368641
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43432}
Also CHECK in OutputPathWithRandomDirectory. This function is used in tests that need a unique folder to avoid interaction with other tests that may run in parallel. Continuing with a non-unique folder if the creation fails, is likely to cause surprising errors later on.
Bug: webrtc:379973428
Change-Id: I6a30ef9034be8132e2362eff5e46e3b99b30acd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43431}
