Yury Yarashevich
ea7f3d7230
Update iOS H264 profile+level table.
...
Added H264 profile level information for new devices.
Use machine name to form table to simplify later updates.
Implemented workaround for unknown devices.
Previous update was done as part of:
https://webrtc-review.googlesource.com/c/src/+/256976
Device machine names obtained from:
https://gist.github.com/adamawolf/3048717
Machine name to device model matching was done with:
https://everymac.com/ultimate-mac-lookup/
Bug: webrtc:15094
Change-Id: I85b7faa51b9f239d0b7783b9926449e02f5482d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303760
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39962}
2023-04-28 08:45:25 +00:00
Linus Nilsson
df4bc33e11
Allow EglBase instances to share EGLConnection.
...
This enables clients of EglBase to keep using it but
share underlying EGLContext with other clients.
go/meet-android-eglcontext-reduction
Bug: b/225229697
Change-Id: I42719f25be7db169c39878b57a5f1487e3c1894e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301941
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Linus Nilsson <lnilsson@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39961}
2023-04-27 19:59:05 +00:00
Danil Chapovalov
9ecc76e15b
Use Timestamp type in RtpState struct
...
Bug: webrtc:13757
Change-Id: I7f8fc1a9c4cbf464b3969c4754ce5aa9c5b5f076
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39960}
2023-04-27 11:24:38 +00:00
Philipp Hancke
f78d1f211a
stats: Implement receive RTX stats
...
* retransmittedBytesReceived
* retransmittedPacketsReceived
added to the specification in
https://github.com/w3c/webrtc-stats/pull/735
BUG=webrtc:15096
Change-Id: I6770e5d8d09ac1c2693c918fd943b0ab257ec7ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295260
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39959}
2023-04-27 09:53:00 +00:00
Philipp Hancke
2b72d84733
stats: fix type of inbound-rtp frames_received
...
which gets assigned from a uint32_t VideoReceiverInfo::frames_received so should remain an unsigned type
BUG=None
Change-Id: I1db6a3f96c4ff49eee72dcce54eb6fff346c128c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302342
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39958}
2023-04-26 15:57:46 +00:00
Jakob Ivarsson
2bd878180a
Add delayed packet outage event metric.
...
Can be used to calculate the average delayed packet outage duration and
number of packet loss events by subtracting from concealment events.
Only used in simulations currently.
Bug: None
Change-Id: I03740a2bcb781af09e28a4d13d9e41c0f84bc506
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303600
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39957}
2023-04-26 13:40:17 +00:00
Jakob Ivarsson
ecdedac3da
Remove NetEq simulation step size restriction.
...
This should not be relevant anymore and is causing some issues due to
SetMinimumDelay events early in the log.
Bug: None
Change-Id: Ib7e3c624608c9bceaed31bd6669db59887d24659
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303580
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39956}
2023-04-26 13:07:12 +00:00
Per K
d1771e925d
Enable SSL logging per default
...
Done in order to simplify connection debuging.
Example log:
openssl_adapter.cc:829): connect_loop TLS client read_server_hello
(openssl_adapter.cc:829): connect_loop TLS client read_server_certificate
(openssl_adapter.cc:829): connect_loop TLS client read_certificate_status
(openssl_adapter.cc:829): connect_loop TLS client verify_server_certificate
(openssl_stream_adapter.cc:1128): Accepted peer certificate.
(openssl_adapter.cc:829): connect_loop TLS client read_server_key_exchange
(openssl_adapter.cc:829): connect_loop TLS client read_certificate_request
(openssl_adapter.cc:829): connect_loop TLS client read_server_hello_done
(openssl_adapter.cc:829): connect_loop TLS client send_client_certificate
(openssl_adapter.cc:829): connect_loop TLS client send_client_key_exchange
(openssl_adapter.cc:829): connect_loop TLS client send_client_certificate_verify
(openssl_adapter.cc:829): connect_loop TLS client send_client_finished
(openssl_adapter.cc:829): connect_loop TLS client finish_flight
(openssl_adapter.cc:829): connect_loop TLS client read_session_ticket
(openssl_adapter.cc:829): connect_exit TLS client read_session_ticket
(openssl_adapter.cc:829): accept_loop TLS server verify_client_certificate
(openssl_stream_adapter.cc:1128): Accepted peer certificate.
(openssl_adapter.cc:829): accept_loop TLS server read_client_key_exchange
(peer_connection.cc:1952): Changing IceConnectionState 0 => 1
(openssl_adapter.cc:829): accept_loop TLS server read_client_certificate_verify
(peer_connection.cc:1971): Changing standardized IceConnectionState 0 => 1
(peer_connection.cc:1971): Changing standardized IceConnectionState 0 => 1
(peer_connection.cc:1971): Changing standardized IceConnectionState 1 => 2
(peer_connection.cc:1971): Changing standardized IceConnectionState 1 => 2
(openssl_adapter.cc:829): accept_loop TLS server read_change_cipher_spec
(openssl_adapter.cc:829): accept_loop TLS server process_change_cipher_spec
(openssl_adapter.cc:829): accept_loop TLS server read_next_proto
(openssl_adapter.cc:829): accept_loop TLS server read_channel_id
(openssl_adapter.cc:829): accept_loop TLS server read_client_finished
(openssl_adapter.cc:829): accept_loop TLS server send_server_finished
(openssl_adapter.cc:829): accept_loop TLS server finish_server_handshake
(openssl_adapter.cc:829): accept_loop TLS server done
(openssl_adapter.cc:829): handshake_done TLS server done
(openssl_adapter.cc:829): accept_exit TLS server done
(dtls_transport.cc:688): DtlsTransport[0|1|__]: DTLS handshake complete.
Bug: b/275671043
Change-Id: Ib8d394aa74c5665c489b485bb44152aff67d3b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302300
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39955}
2023-04-26 13:01:13 +00:00
Artem Titov
17d7eb4d52
Do not compile some test targets with chromium
...
Move copy_to_file_audio_capturer, copy_to_file_audio_capturer_unittest
and test_common under "!build_with_chromium"
Bug: b/272350185, webrtc:15081
Change-Id: Ie3f08e4ce5bec91647e802cc34040df2e01103d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39954}
2023-04-26 10:07:49 +00:00
Andreas Pehrson
28ac56a415
In VideoCaptureDS::Stop() fully stop the device
...
This makes the device light turn off when stopped.
Bug: webrtc:15109
Change-Id: I1deecbc2463e2e316e01ff1f061ab6b0313c1aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302200
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39953}
2023-04-26 08:58:46 +00:00
chromium-webrtc-autoroll
d6e6953ada
Roll chromium_revision 26dc712e58..d0ae9456ec (1135380:1135488)
...
Change log: 26dc712e58..d0ae9456ec26dc712e58..d0ae9456ec5d6d0d4d07..f723499917489b131ab0..f6692ccd70d617549f90..d23247d9e7a9eda3ac94..fd370504bac37a1309dd..fdea1c758d26dc712e58..d0ae9456echttps://webrtc-review.googlesource.com/c/src/+/303620
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Cr-Commit-Position: refs/heads/main@{#39952}
2023-04-25 20:34:42 +00:00
chromium-webrtc-autoroll
a993ff2871
Roll chromium_revision 063d347336..26dc712e58 (1135237:1135380)
...
Change log: 063d347336..26dc712e58063d347336..26dc712e58a972e3554c..489b131ab02eff2571d4..c78294c8c29bf8d1ec52..bfde407de5fce1207a83..c37a1309dd063d347336..26dc712e58https://webrtc-review.googlesource.com/c/src/+/303560
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39951}
2023-04-25 18:23:54 +00:00
chromium-webrtc-autoroll
3ac547a229
Roll chromium_revision 0c1d6778e0..063d347336 (1135085:1135237)
...
Change log: 0c1d6778e0..063d3473360c1d6778e0..063d347336fe22033c21..5d6d0d4d07a9d28a095c..a972e3554ca2df0a6e72..2eff2571d4ee4801b4e9..d617549f904f8bf4c688..a9eda3ac949806414c15..0a3836c97d2822b589bc..2175f5d05020b114cd06..f2da6df2f1bafae7909c..fce1207a830c1d6778e0..063d347336https://webrtc-review.googlesource.com/c/src/+/303442
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Cr-Commit-Position: refs/heads/main@{#39950}
2023-04-25 17:04:24 +00:00
Sameer Vijaykar
df7df199ab
Clean up IPv6 fixes field trial artifacts.
...
The fixes have been default enabled, so clean up dead code.
Bug: webrtc:14334
Change-Id: I4967d5ad451ac333c54294fc14bea6c7ba1445e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301180
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#39949}
2023-04-25 14:59:55 +00:00
Danil Chapovalov
52275845a0
Use Timestamp type instead of int64_t in Flexfec classes
...
Bug: webrtc:13757
Change-Id: Ideafea65adb827b5457de22a04e3235cda3ffd5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301260
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39948}
2023-04-25 10:53:08 +00:00
Jeremy Leconte
b035dcc0a2
Revert "Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl""
...
This reverts commit eeae96299784515f573379a64655eb07a5973a3a.
Reason for revert: breaks WebRTC Chromium FYI ios-device
https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/14896/overview
Original change's description:
> Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
>
> This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9.
>
> Reason for revert: Reland with a fix
>
> Original change's description:
> > Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
> >
> > This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b.
> >
> > Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814
> >
> > Original change's description:
> > > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl
> > >
> > > Bug: b/272350185
> > > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363
> > > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > > Cr-Commit-Position: refs/heads/main@{#39877}
> >
> > Bug: b/272350185
> > Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701
> > Commit-Queue: Jeremy Leconte <jleconte@google.com>
> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> > Reviewed-by: Jeremy Leconte <jleconte@google.com>
> > Auto-Submit: Christoffer Jansson <jansson@google.com>
> > Owners-Override: Christoffer Jansson <jansson@google.com>
> > Cr-Commit-Position: refs/heads/main@{#39881}
>
> Bug: b/272350185
> Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39936}
Bug: b/272350185
Change-Id: If0a10717bf14a0a618e52728fc3a61b9c55f3bd2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303460
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39947}
2023-04-25 10:24:56 +00:00
Artem Titov
8a9f3a8f53
Reland "Remove dependency of video_replay on TestADM."
...
This reverts commit f9e3bdd2ce410b18ca7e03b3754f94a18eb7ef3a.
Reason for revert: reland with fix
Original change's description:
> Revert "Remove dependency of video_replay on TestADM."
>
> This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67.
>
> Reason for revert: breaking CallPerfTest
> https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview
>
> Original change's description:
> > Remove dependency of video_replay on TestADM.
> >
> > This should remove requirement to build TestADM in chromium build.
> >
> > Bug: b/272350185, webrtc:15081
> > Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39934}
>
> Bug: b/272350185, webrtc:15081
> Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Jeremy Leconte <jleconte@google.com>
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39939}
Bug: b/272350185, webrtc:15081
Change-Id: I360ef3e140e60fc21d622480d1f3326e40a76f58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Auto-Submit: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39946}
2023-04-25 09:39:22 +00:00
Philipp Hancke
b11caa366c
Remove obsolete IceProtocolType enum and SetIceProtocolType
...
which only had a single member after the removal of
GICE around M42. The last downstream usage in Chromoting
was removed in
https://chromium-review.googlesource.com/c/chromium/src/+/4385113
BUG=webrtc:4299
Change-Id: Id444967822cd19b0e514ba70739a8d45a7f78fae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299600
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39945}
2023-04-25 09:25:33 +00:00
chromium-webrtc-autoroll
83c8a3b885
Roll chromium_revision fbcde4ef84..0c1d6778e0 (1134788:1135085)
...
Change log: fbcde4ef84..0c1d6778e0fbcde4ef84..0c1d6778e0304fd6d0cd..fe22033c2177af5d07d2..a9d28a095c2482155040..a2df0a6e72fae97ad698..ee4801b4e976af9e74bb..4f8bf4c688https://chromium.googlesource.com/catapult.git/+log/1aa5adbafb..cae7ec667d
* src/third_party/depot_tools: b5cec8c867..6e714e6dfeaa34142ee6..20b114cd06da5a1a8add..bafae7909cfbcde4ef84..0c1d6778e0https://webrtc-review.googlesource.com/c/src/+/303420
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Cr-Commit-Position: refs/heads/main@{#39944}
2023-04-25 09:18:46 +00:00
Tommi
cde4b67d9d
[SourceTracker] Move state to the worker thread, remove mutex.
...
This is in preparation of using the state that SourceTracker manages
for more things than only getContributingSources. Audio levels reported
via getStats(), aren't consistent with levels reported via getCS.
Since more operations will be derived from the ST owned data, moving
the management of it away from the audio thread, reduces the potential
of contention.
Bug: webrtc:14029, webrtc:7517, webrtc:15119
Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39943}
2023-04-25 08:18:42 +00:00
Erik Språng
031ebc42e6
Increase RTP send buffer size from 64kb to 256kb.
...
Assuming 15Mbps video bitrate at 30fps, a single frame is 62500 bytes.
Add to that some fluctuations in encoder output rate and capture fps,
and frames can easily become larger than 64kb.
Given enough bandwidth and the bursty pacer, it will not be uncommon to
send the entire frame in one batch - and if the send buffer is at 64kb
then you will likely get packetloss already in the IPC packet socket,
even before the packet has reached the network card!
It's not entirely clear what the optimal size is, but given that the
receive buffer size was increased from 64kb to 256kb for high bandwidth
receive scenarios and had negligible negative effects I think it's
pretty safe to bump the send buffer to match.
There is a field trial available that can be used as circuit breaker
in case things turn south: WebRTC-SendBufferSizeBytes
Bug: webrtc:14780
Change-Id: I6c786d993181a882e6dce832ff56dc92d2a8a341
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290985
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39942}
2023-04-24 21:30:26 +00:00
chromium-webrtc-autoroll
6a34c75d5d
Roll chromium_revision 43b82356e4..fbcde4ef84 (1134651:1134788)
...
Change log: 43b82356e4..fbcde4ef8443b82356e4..fbcde4ef84e6be8e9377..304fd6d0cd71e9e15d4e..77af5d07d20742b59781..2482155040e43d55ad90..fae97ad6985676009b37..76af9e74bb8154d8e2be..9806414c155437caf87b..aa34142ee6cbcf55a40f..da5a1a8add43b82356e4..fbcde4ef84https://webrtc-review.googlesource.com/c/src/+/303180
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Cr-Commit-Position: refs/heads/main@{#39941}
2023-04-24 20:44:43 +00:00
Andreas Pehrson
ba41b40461
webrtc_libyuv: Add support for more video types for consistency
...
Bug: webrtc:14830
Change-Id: I0998fb04a03745131f9f5cca878b0cdb46f3b62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291529
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39940}
2023-04-24 19:06:25 +00:00
Jeremy Leconte
f9e3bdd2ce
Revert "Remove dependency of video_replay on TestADM."
...
This reverts commit 01716663a9837a26fa292fe70fdea353cbd01a67.
Reason for revert: breaking CallPerfTest
https://ci.chromium.org/ui/p/webrtc/builders/perf/Perf%20Android32%20(R%20Pixel5)/967/overview
Original change's description:
> Remove dependency of video_replay on TestADM.
>
> This should remove requirement to build TestADM in chromium build.
>
> Bug: b/272350185, webrtc:15081
> Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39934}
Bug: b/272350185, webrtc:15081
Change-Id: I73aa0fd3c3d8c244d20e5f29f5792a4c7d7e4165
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303160
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39939}
2023-04-24 19:02:23 +00:00
chromium-webrtc-autoroll
6c2f7602ca
Roll chromium_revision 0ed338af59..43b82356e4 (1134512:1134651)
...
Change log: 0ed338af59..43b82356e40ed338af59..43b82356e461c4b10212..71e9e15d4e8de181f514..0742b59781f882a7628e..e43d55ad90cb50f56348..5676009b37df2191d9ce..5437caf87b94fc7fddea..cbcf55a40f0ed338af59..43b82356e4https://webrtc-review.googlesource.com/c/src/+/303120
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39938}
2023-04-24 17:01:41 +00:00
Philipp Hancke
6a7bf10d60
Replace "rcvd" with "received" for readability
...
following guidance in
https://google.github.io/styleguide/cppguide.html#General_Naming_Rules
BUG=None
Change-Id: I105591a7f709d65a3d3320f7f44137d432cf7ce0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39937}
2023-04-24 15:30:07 +00:00
Artem Titov
eeae962997
Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
...
This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9.
Reason for revert: Reland with a fix
Original change's description:
> Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
>
> This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b.
>
> Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814
>
> Original change's description:
> > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl
> >
> > Bug: b/272350185
> > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39877}
>
> Bug: b/272350185
> Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Auto-Submit: Christoffer Jansson <jansson@google.com>
> Owners-Override: Christoffer Jansson <jansson@google.com>
> Cr-Commit-Position: refs/heads/main@{#39881}
Bug: b/272350185
Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39936}
2023-04-24 14:42:08 +00:00
chromium-webrtc-autoroll
39e859901b
Roll chromium_revision 1cc6fa230c..0ed338af59 (1133778:1134512)
...
Change log: 1cc6fa230c..0ed338af591cc6fa230c..0ed338af59a7ef445859..e6be8e93778e40aaeb1b..61c4b102128d06fc2ffa..539a6f687364d1adcc57..307bd16360665c2e5429..2795322d5749f40887aa..8de181f51475567f4546..f882a7628e6b423834d7..cb50f56348https://chromium.googlesource.com/catapult.git/+log/f7a8b55f34..1aa5adbafb
* src/third_party/depot_tools: dec6fddc86..b5cec8c86726b0877163..df2191d9ce2a1c696af9..94fc7fddea1cc6fa230c..0ed338af59https://webrtc-review.googlesource.com/c/src/+/303040
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39935}
2023-04-24 13:50:49 +00:00
Artem Titov
01716663a9
Remove dependency of video_replay on TestADM.
...
This should remove requirement to build TestADM in chromium build.
Bug: b/272350185, webrtc:15081
Change-Id: Iceb8862aa81099c22bd378ae692229f01ab3314c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302380
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39934}
2023-04-24 13:17:45 +00:00
Jeremy Leconte
47701c8c9b
Chromium LocalRobolectricTestRunner has been removed.
...
This is a follow up on https://chromium-review.googlesource.com/c/chromium/src/+/44503 .
Change-Id: I28a0789a0af43cfac27081c9b5bcf695e9798910
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303020
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39933}
2023-04-24 11:26:11 +00:00
Tommi
94774d475b
Call PrepareShutdown in the dtor just in case Close() hasn't been called
...
Bug: b/277912909
Change-Id: I0074de59f5d16d500795589a0c94ff4840ffe57a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302384
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39932}
2023-04-24 11:06:42 +00:00
Philipp Hancke
70fc5a2e41
stats: unify optional handling to use operator*
...
following https://abseil.io/tips/181#solution
BUG=None
Change-Id: I865572e42dff172fcf722383f3dde31dcc747220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302341
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39931}
2023-04-24 10:58:04 +00:00
philipel
c22893b3f6
Add AV1 perf tests.
...
Bug: b/273502945
Change-Id: I3b1379c8757f4e1ea38d9575eb2a32d955f0643f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302401
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39930}
2023-04-24 10:45:15 +00:00
philipel
d1e5dedffe
Use DD encoder/decoder in RTC event log encoder/parser.
...
Bug: webrtc:14801
Change-Id: I7013c42765e81d147bf8284f8c29666e67fdb91f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296765
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39929}
2023-04-24 10:35:22 +00:00
Tommi
94e5817759
Guard FakeDataChannelController state with the network thread.
...
Tsan bots detected races since callbacks are being made on the network
thread but tests checked the state from the signaling thread.
Bug: none
Change-Id: If854e44159c56c0d12616e0b62ad92018291ed30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302281
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39928}
2023-04-24 10:18:27 +00:00
Victor Boivie
daaa6ab5a8
dcsctp: Add handover state for zero checksum
...
This CL can prepare downstream projects for being aware of
this new handover state.
This was extracted from change 299076.
Bug: webrtc:14997
Change-Id: I35bfbe040ffbaa5d7266eb67d58078b66083337a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302980
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39927}
2023-04-24 10:06:40 +00:00
Tommi
683f3165f9
Add slightly more constness to SourceFrame and the embedded AudioFrame
...
This makes it a bit more clear that values of member variables of
SourceFrame are never directly changed and that doing so is not an
intentional part of the design. Also made use of `SourceFrame` vs
`const SourceFrame` more consistent since the audio frame of a
`const SourceFrame` was being modified in some places.
Accessing the embedded AudioFrame can be done via the const
audio_frame() accessor or via the mutable_audio_frame() accessor when
modifying the frame is needed. This helps with clarifying later on
when downstream code paths such as ones that access the `packet_infos_`
data, can know that it won't be modified for the rest of the frame's
lifetime (thus avoiding having to make copies).
Bug: none
Change-Id: I175cec8fcdb85063239a5f9c299b7878c639f00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302383
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39926}
2023-04-24 10:05:35 +00:00
Jeremy Leconte
4e0bdf550b
iOS64 Perf bot build15-a7 is replaced by mac-254-e504.
...
Change-Id: I65a0065bee7dad345668541eddcc7a53fec2dab3
Bug: b/278663917
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302402
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39925}
2023-04-24 07:44:12 +00:00
webrtc-version-updater
b810a64db1
Update WebRTC code version (2023-04-24T04:05:22).
...
Bug: None
Change-Id: I51a2a0a8aaaffc8ff147d9055b081029776b666b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302940
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39924}
2023-04-24 06:02:33 +00:00
Victor Boivie
014cbed9d2
Revert "dcsctp: Negotiate zero checksum"
...
This reverts commit a736f30a5fddfa9a6af02a0a916da09bcac49d0d.
Due to a downstream project not supporting this
new handover state, it fails. Handover state needs
to be submitted first, then downstream project needs
to be updated, and finally the code changes can be
submitted.
Bug: webrtc:14997
Change-Id: I8551e349408d9bf4eb593cb948279d659467fe20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302821
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39923}
2023-04-23 22:25:44 +00:00
Artem Titov
e91c76875a
Complete move of TestADM into its own target
...
Bug: b/272350185, webrtc:15081
Change-Id: I1a7ffedae34790ed08c0205c713a650efd36273d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302340
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39922}
2023-04-21 18:33:33 +00:00
Sergey Silkin
0421294df0
Enable RTC mode in Google HW AV1 encoder
...
Bug: b/274179852
Change-Id: Id987fa8b73468532304f856bc97d40238f93b266
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302180
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39921}
2023-04-21 16:38:06 +00:00
Victor Boivie
a736f30a5f
dcsctp: Negotiate zero checksum
...
If configured, attempt to negotiate "zero checksum
acceptable" capability, which will make the outgoing
packets have a fixed checksum of zero. Received
packets will not be verified for a correct checksum
if it's zero.
Also includes some boilerplate:
- Setting capability in state cookie
- Adding capability to handover state
- Adding metric to know if the feature is used
This feature is not enabled by default, as it will be
first evaluated with an A/B experiment before making
it the default.
When the feature is enabled, lower CPU consumption for
both receiving and sending packets is expected. How
much depends on the architecture, as some architectures
have better support for generating CRC32 in hardware
than others.
Bug: webrtc:14997
Change-Id: If23f73e87220c7d42bd4f9a92772cda16bc18fcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299076
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39920}
2023-04-21 15:31:35 +00:00
chromium-webrtc-autoroll
325b3caac6
Roll chromium_revision 2598ff6c3a..1cc6fa230c (1133653:1133778)
...
Change log: 2598ff6c3a..1cc6fa230c2598ff6c3a..1cc6fa230c827c543161..a7ef445859f78fc195a7..8e40aaeb1ba64df6cce2..64d1adcc5748d230ebc1..49f40887aada88a644b4..75567f45462a07492c50..6b423834d74f9f0278b3..26b0877163206447d5cb..2a1c696af92598ff6c3a..1cc6fa230chttps://webrtc-review.googlesource.com/c/src/+/302362
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39919}
2023-04-21 14:45:42 +00:00
Tommi
51c632c13b
Use GlobalSimulatedTimeController in more webrtc video engine unittests
...
This removes all "_WAIT" operations in the tests and all uses
of kTimeout for loop+poll check for values.
On my linux box, this also reduces the time it takes to run all
./rtc_media_unittests (in parallel) from about 3500ms, to ~50ms.
(longest running test was WebRtcVideoChannelBaseTest.GetStats)
Bug: none
Change-Id: If544aa10cb0281cb5e5e51fb654db5f45de871da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302343
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39918}
2023-04-21 14:41:27 +00:00
Sergey Silkin
691b447c53
Fix returns from IsSameSettings and IsSameRate in codec tests
...
Swap true/false.
Bug: webrtc:14852
Change-Id: Id82c0180d33bfc4e5237f4800c3e89fe8d17693f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39917}
2023-04-21 11:29:32 +00:00
Philipp Hancke
ff6cd53303
Reject stun urls with query parameters
...
for consistency with Chromium behavior, in particular
stun:host?transport=udp
gets rejected.
BUG=chromium:1385735
Change-Id: I85a141ecf72480bfa09d8354d6dcaef8ca0cdcff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299943
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@{#39916}
2023-04-21 11:19:37 +00:00
chromium-webrtc-autoroll
7aa6a72dc6
Roll chromium_revision e0f3c7ef30..2598ff6c3a (1133548:1133653)
...
Change log: e0f3c7ef30..2598ff6c3ae0f3c7ef30..2598ff6c3a09a9fa99c1..827c5431614a2a31a17b..f78fc195a7b8ae314430..48d230ebc1592b965c70..da88a644b496a84a37ab..2a07492c5034f62b4af7..dec6fddc864d46c527b1..4f9f0278b3069e2f6512..206447d5cbe0f3c7ef30..2598ff6c3ahttps://webrtc-review.googlesource.com/c/src/+/302320
Bot-Commit: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39915}
2023-04-21 08:39:26 +00:00
Jared Siskin
802e8e5fdb
Format /rtc_base
...
git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^rtc_base/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs
Bug: webrtc:15082
Change-Id: I152228f7c7926adf95d2f3fbbe4178556fd75d0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302061
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39914}
2023-04-21 06:17:42 +00:00
Jared Siskin
6f86f6af00
Format /sdk
...
git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -e "^sdk/" | xargs clang-format -i ; git cl format
after landing: add to .git-blame-ignore-revs
Bug: webrtc:15082
Change-Id: If05d3c7555c4f2bf25e387249932787a93aa39c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39913}
2023-04-21 04:30:57 +00:00
