Call operators do not improve code clarity, and usage was moderate.
Bug: None
Change-Id: I8d86bd7d435ce88e99f4abee8ab95a336d47dc22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292960
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39294}
Step 1 of combining the sender and receiver types
Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used
by the transformable receiver frame, I couldn't work out a better way
of managing the dependencies, and everything else seemed to work fine.
Bug: chromium:1412687
Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39255}
This is used when an unsignaled stream with a known payload type is received and later a RTX packet is received.
Bug: webrtc:14817
Change-Id: I29f43281cec17553e1ec2483e21b8847714d2931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291328
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39243}
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.
Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
Before this, an empty list of CSRCs was always provided up to encoded
insertable streams transforms for remote video tracks, regardless of
the actual CSRCs on received frames. Audio already works correctly.
Bug: chromium:1411614
Change-Id: I51ab4dc5e67a1a35893fefff16c1f057e9047e6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291539
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39240}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
This reverts commit 3e61f881cd2ba9040a07371e0ba6dda902aa60ae.
Reason for revert: Issue fixed in https://webrtc-review.googlesource.com/c/src/+/291104
Original change's description:
> Revert "Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp"
>
> This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
>
> Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
>
>
> Original change's description:
> > Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
> >
> > PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> > Therefore DirectTransport is provided with the extension mapping.
> >
> > CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
> >
> >
> > Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> > Bug: webrtc:7135, webrtc:14795
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> > Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> > Commit-Queue: Per Kjellander <perkj@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39137}
>
> Bug: webrtc:7135, webrtc:14795, webrtc:14833
> Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Auto-Submit: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39146}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: I3fb0210d7a33c600ead5719ce2acb8cc68ec20bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291222
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39157}
This reverts commit 3b96f2c770df7691df90c2cc1be40509a76ae425.
Reason for revert: Seems to cause test failures and perf regressions in tests: webrtc:14833, and CallPerfTest.Min_Bitrate_VideoAndAudio
Original change's description:
> Change CallTests to use new PacketReceiver::DeliverRtp and PacketReceiver::DeliverRtcp
>
> PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
> Therefore DirectTransport is provided with the extension mapping.
>
> CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
>
>
> Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
> Bug: webrtc:7135, webrtc:14795
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39137}
Bug: webrtc:7135, webrtc:14795, webrtc:14833
Change-Id: Ib6180a47cf7611ed2bc648acc3b9e5cfeec4d9cf
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291220
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39146}
PacketReceiver::DeliverRtp requires delivered packets to have extensions already mapped.
Therefore DirectTransport is provided with the extension mapping.
CallTests and tests derived from CallTest create transports in different ways, this cl change CallTest to create tests in only one way to simplify how extensions are provided to the transport but at the same time still allows different network behaviour.
Change-Id: Ie8b3ad947c170be61e62c02dadf4adedbb3841f1
Bug: webrtc:7135, webrtc:14795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39137}
RTPVideoHeader is changed to non-const to allow modifying it. We want
to do this when implementing setMetadata() in JavaScript or when
refactoring clone() as "construct + set bytes + setMetadata".
Unblocks
https://chromium-review.googlesource.com/c/chromium/src/+/4164979.
Bug: webrtc:14709
Change-Id: I6089df9c03e9aa33feeb0830dd240dd456cb565e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39113}
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}
When disabling a spatial layer, reconfiguration of the encoder is not
necessary (bitrate will never be assigned to the inactive layer anway).
This CL however makes sure we reconfigure the encoder when a spatial
layer is activated. Some encoder implementations may encoder the wrong
number of spatial layers if the active layers have not beens set
correctly.
Bug: webrtc:14809, b/261097903
Change-Id: I8d34aaec95eb50a9717c06ea38f25088e5a96429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290560
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38999}
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}
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}
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}
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}
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}
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}
This CL reverts change that was made in https://webrtc-review.googlesource.com/c/src/+/174220 where capturing of frame rendered time was moved into VideoFrameMetaData and construction of VideoFrameMetaData was placed before renderer->OnFrame(). That change made video freeze metrics (freeze and pause durations, harmonic frame rate) more off from the actual user experience.
Bug: webrtc:11489, b/261512902
Change-Id: Ic92a0cc1bb6d7b3ee1023804a73aa282848f8c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38883}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
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}
In preparation of adding RTPVideoHeader::SetFromMetadata() method, the
VideoFrameMetadata construct-from-RTPVideoHeader is replaced by
RTPVideoHeader::GetAsMetadata(). This serves two purposes:
1. Having "GetAs" and "SetFrom" in the same file reduces the risk of
these two methods getting out of sync as we expand its usage.
2. This is necessary to avoid a circular dependency that would
otherwise be introduced by RTPVideoHeader::SetFromMetadata().
Bug: webrtc:14709
Change-Id: I127b3d15f9a8c6af210449a5a50d414c9ba79930
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285080
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38735}
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}
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}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
