which has been enabled by default since M84. This was still available
under an enterprise policy which is gone since M121:
https://chromiumdash.appspot.com/commit/39d28bb7657b482f1fdcab81ca88371d8914809b
BUG=webrtc:10261,chromium:1132854
Change-Id: Icd534342b60799b7862bc3e7edda6825de7ae976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41145}
* Pass codec factories to the video codec tester instead of creating and wrapping codecs into a tester-specific wrappers in video_codec_test.cc. The motivation for this change is to simplify the tests by moving complexity to the tester.
* Merge codec stats and analysis into the tester and move the tester. The merge fixes circular deps issues. Modularization is not strictly needed for testing framework like the video codec tester. It is still possible to unit test underlaying modules with rather small overhead.
* Move the video codec tester from api/ to test/. test/ is accessible from outside of WebRTC which enables reusing the tester in downstream projects.
Test output ~matches before and after this refactoring. There is a small difference that is caused by changes in qpMax: 63 -> 56 (kDefaultVideoMaxQpVpx). 56 is what WebRTC uses by default for VPx/AV1 encoders.
Bug: webrtc:14852
Change-Id: I762707b7144fcff870119ad741ebe7091ea109ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327260
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41144}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
While this is a fairly big CL, it's fairly straightforward. It replaces
uses of TimeMs with webrtc::Timestamp, and additionally does some
cleanup of DurationMs to webrtc::TimeDelta that are now easier to do.
Bug: webrtc:15593
Change-Id: Id0e3edcba0533e0e8df3358b1778b6995c344243
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326560
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41138}
This reverts commit 8039cdbe48f8c8bb91fa1761f807005a7b497196.
Reason for revert: remove functionality after measurement complete
Original change's description:
> Measure wall clock time of capture and encode processing.
>
> (NOTE: This and dependent CLs will be reverted in a few days after
> data collection from the field is complete.)
>
> This change introduces a new task queue concept, Voucher. They
> are associated with a currently running task tree. Whenever
> tasks are posted, the current voucher is inherited and set as
> current in the new task.
>
> The voucher exists for as long as there are direct and indirect
> tasks running that descend from the task where the voucher was
> created.
>
> Vouchers aggregate application-specific attachments, which perform
> logic unrelated to Voucher progression. This particular change adds
> an attachment that measures time from capture to all encode operations
> complete, and places it into the WebRTC.Video.CaptureToSendTimeMs UMA.
>
> An accompanying Chrome change crrev.com/c/4992282 ensures survival of
> vouchers across certain Mojo IPC.
>
> Bug: chromium:1498378
> Change-Id: I2a27800a4e5504f219d8b9d33c56a48904cf6dde
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325400
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41061}
Bug: chromium:1498378
Change-Id: I9503575fbc52f1946ca26fc3c17b623ea75cd3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327023
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#41135}
These functions had no callers and no tests.
Under YAGNI principles, they need to be deleted.
Bug: None
Change-Id: I8b5d74678b804ef2be70409d05a5237f1637eaea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327024
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41134}
This reverts commit 03bc3a0fa67e274efb4518da005f4c5b77c607e9.
Reason for revert: measurement complete
Original change's description:
> VideoStreamEncoder: exclude screencast from capture time measurement.
>
> This CL avoids measurement for screencast encoding work. The reason is
> screencast can cling on to and re-encode old video frames for which
> webrtc::VideoFrame::reference_time() is unchanged.
>
> Bug: chromium:1498378
> Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41097}
Bug: chromium:1498378
Change-Id: I42c1a86123eb1d6c7ad7c8981769f5560884a2f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327025
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41133}
This callback is identical to TimeMillis, but returns a
webrtc::Timestamp instead of a TimeMs.
When all callers have migrated to Now() (and all dcsctp code),
TimeMillis will be removed.
Bug: webrtc:15593
Change-Id: I608387607537f29989736af5bf98c7f184f52ebc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326500
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41127}
- Stable delay mode: this results in a very large reduction in the amount of time stretching and fewer underruns.
- More closely align PLC and CNG logic.
- Stop playing comfort noise after a timeout when no packets are received.
Several tests needed to be updated to match the new behavior.
Note that I should also refactor GetDecision to be easier to test in the future (remove internal state).
Bug: webrtc:13322
Change-Id: I1724a74b3b583d05a4bb8feb4f9a8c4a8b2b7c5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326780
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41125}
This is a reland of commit 3ea9fc4cd8135555360aafbfe788571d9e2f23f9
Original change's description:
> Make frame transformer MimeType pure virtual again
>
> after both audio and video have been implemented.
>
> BUG=webrtc:15579
>
> Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Tony Herre <herre@google.com>
> Cr-Commit-Position: refs/heads/main@{#41114}
BUG=webrtc:15579
Change-Id: Ia020149cba3045022b539f290565d6c1d0e813ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326880
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41121}
std::is_pod is deprecated since C++20. Replace with std::trivial and
std::is_standard_layout. Avoids a lot of warnings.
Bug: chromium:957519
Change-Id: Idb4bde7401c14c0896a84c357ec668b9916f613e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325484
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41117}
after both audio and video have been implemented.
BUG=webrtc:15579
Change-Id: Ib52e8f67292259cbf7497a884672de72f3003282
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326162
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#41114}
Deprecate old version but keep it for the time being.
Bug: webrtc:15368
Change-Id: Icbd2078a00d877ff948a2441c2027a12c85d4f2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326104
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41112}
This uses libSRTPs srtp_remove_stream()
https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597
method to remove SSRCs from the libSRTP session when they are removed
from the RTP demuxer. This works even when the stream was added
automatically via the ssrc_any_inbound mechanism.
Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed.
Guarded by WebRTC-SrtpRemoveReceiveStream field trial.
BUG=webrtc:15604
Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41105}
With this, the code base should be mostly converted from using
DurationMs to rtc::TimeDelta, and the work can continue to replace
TimeMs with rtc::Timestamp.
Bug: webrtc:15593
Change-Id: I083fee6eccb173efc0232bb8d46e2554a5fbee5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326161
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41101}
Before this change, the FCA did not not update its cadence when
max_fps was changed and zero-hertz was already enabled.
See https://paste.googleplex.com/6300124249587712 for more details.
Bug: chromium:1400204
Change-Id: I95d80bdfa85ecac8681784b2b29e98d1a587ba53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326105
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41100}
This CL avoids measurement for screencast encoding work. The reason is
screencast can cling on to and re-encode old video frames for which
webrtc::VideoFrame::reference_time() is unchanged.
Bug: chromium:1498378
Change-Id: I5bf79d29ef7f57ddff2622cbb6c3436480bd16ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41097}
