5b298ab9dd
For VP9 assume max number of spatial layers to simulate generic descriptor
...
VP9 allows to increase number of spatial layers on delta frame, which
is not supported by dependency descriptor.
Thus to generate DD compatible generic header, simulator would set max
number of spatial layers, while number of active spatial layers would be
communicated with active_decode_target bitmask
Bug: webrtc:14042
Change-Id: I4da63fa7c38b0f17758a7a6243640f444470b40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265164
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org >
Reviewed-by: Philip Eliasson <philipel@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37151}
2022-06-08 11:36:54 +00:00
6545516a14
RtpSenderInterface::SetEncoderSelector
...
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.
The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.
Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.
Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Jonas Oreland <jonaso@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:06:52 +00:00
1622d35b30
Update WebRTC code version (2022-06-08T04:01:49).
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Bug: None
Change-Id: I666467a288c5cf96edf18291c415ac8ad25ba6ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265340
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Cr-Commit-Position: refs/heads/main@{#37146}
2022-06-08 05:36:22 +00:00
eef9d5d6e1
Update WebRTC code version (2022-06-07T04:02:21).
...
Bug: None
Change-Id: I693fe98b515888a97bb6d1f0492339cef3af3158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265221
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Cr-Commit-Position: refs/heads/main@{#37135}
2022-06-07 05:53:27 +00:00
d589fbdaca
Update WebRTC code version (2022-06-06T04:04:36).
...
Bug: None
Change-Id: If1b5d6b0f12696eb738925df4a862bde846355cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265148
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Cr-Commit-Position: refs/heads/main@{#37130}
2022-06-06 05:22:55 +00:00
5f5bdf1880
Update WebRTC code version (2022-06-05T04:05:34).
...
Bug: None
Change-Id: Iff90bd4daa09cf5f48340540a0b60a403987b147
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265095
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37127}
2022-06-05 05:41:45 +00:00
1062c15c2d
Update WebRTC code version (2022-06-04T04:04:24).
...
Bug: None
Change-Id: I74d408854f8bd76f2359701d58e5246ee3df2190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265083
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37125}
2022-06-04 06:13:04 +00:00
55a5588f77
Update OWNERS for call/
...
No-try: True
Bug: None
Change-Id: I53103a75c3c5e1d9c798c68bcca7c2041129d32a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265001
Commit-Queue: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37112}
2022-06-03 12:01:46 +00:00
18b82e22ee
Update WebRTC code version (2022-06-03T04:04:21).
...
Bug: None
Change-Id: I533f3455ed016502ab1acdfa42c4988db2000ecc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264891
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37102}
2022-06-03 05:40:37 +00:00
709ab8b536
Update WebRTC code version (2022-06-02T04:05:29).
...
Bug: None
Change-Id: Ice47703cbbbaf183d406e84d09613c00e59a5786
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264761
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Cr-Commit-Position: refs/heads/main@{#37085}
2022-06-02 05:34:46 +00:00
9b9d533d48
Remove deprecated VideoReceiveStream alias
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Bug: webrtc:7484
Change-Id: Id1b3c5e30259ffdad92a1a6ead94dd1acab63cff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264563
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37072}
2022-06-01 10:17:55 +00:00
563dfd1948
Update WebRTC code version (2022-06-01T04:04:59).
...
Bug: None
Change-Id: Ie287ab84ab2760efca0546c637d7742b2c0d14fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264601
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37070}
2022-06-01 06:44:17 +00:00
af785d9759
Deprecate setter RtpRtcpInterface::SetRid
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This setter method is replaced by a construction-time config setting.
Bug: None
Change-Id: Iddffaeeb719a56328bccde3c4a1a0a852d2131b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264501
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37060}
2022-05-31 12:41:13 +00:00
62fabd001b
Update WebRTC code version (2022-05-31T04:04:22).
...
Bug: None
Change-Id: I1f1b63a71c5cd6de2a79e83dbc1401d0da2160ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264418
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@{#37049}
2022-05-31 05:05:24 +00:00
a136ed4085
Add SetTransportCc to ReceiveStreamInterface.
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Setting the transport cc flag was only possible post creation for
audio receive streams, while video receive streams need to be recreated.
This CL moves the setter for transport_cc() to where the getter is and
adds boiler plate implementations for the video streams. For audio
streams this splits "SetUseTransportCcAndNackHistory" into two methods,
SetTransportCc and SetNackHistory.
Bug: none
Change-Id: Idbec8217aef10ee77907cebaecdc27b4b0fb18e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264443
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37038}
2022-05-30 14:07:04 +00:00
5ac19dfefc
Remove deprecated alias, AudioReceiveStream
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Bug: webrtc:7484
Change-Id: If7351a59f384bec04e95e96e5aa0606eca2654f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264440
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37032}
2022-05-30 09:45:03 +00:00
60de8aab46
Remove unused VideoReceiveStreamInterface::Config::target_delay_ms field.
...
Bug: webrtc:14128
Change-Id: I83aa23124ed260b836930463aa712ddd097cef84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263142
Reviewed-by: Artem Titov <titovartem@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37031}
2022-05-30 09:30:23 +00:00
fc2c24ef44
[FlexfecReceiveStream] Use explicit member variables for state.
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This changes FlexfecReceiveStreamImpl so that instead of holding on to
the entire config structure, the state is broken down into member
variables whose constness and thread access can be individually set.
Bug: none
Change-Id: I497b5816d40678774dee76d8a97012e8539629b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263723
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#37027}
2022-05-30 07:37:03 +00:00
34fb92f09a
Update WebRTC code version (2022-05-30T04:04:59).
...
Bug: None
Change-Id: I0a3332f6bc02bc340c4e6c8d3c08089856e2bfce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264404
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37026}
2022-05-30 05:44:23 +00:00
e4374a6020
Update WebRTC code version (2022-05-29T04:03:53).
...
Bug: None
Change-Id: I1857214a6db403d8048167ff131a958007a9c310
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264349
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37024}
2022-05-29 05:26:36 +00:00
e0ad779aaa
Update WebRTC code version (2022-05-28T04:02:55).
...
Bug: None
Change-Id: Id8b9b4e3ba0142f2ff4c7181bd0a81cd7b4ee21e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264302
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Cr-Commit-Position: refs/heads/main@{#37022}
2022-05-28 06:03:10 +00:00
449f5229d6
Update WebRTC code version (2022-05-27T04:04:31).
...
Bug: None
Change-Id: Idcf7c92accf64601da6840f5380eedc520247aed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264081
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#37014}
2022-05-27 07:12:56 +00:00
37623110de
Update WebRTC code version (2022-05-26T04:04:22).
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Bug: None
Change-Id: I05130fb87a7678cb2f25b4f3c151e82c51cddc34
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263860
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Cr-Commit-Position: refs/heads/main@{#37008}
2022-05-26 06:34:19 +00:00
cfc79174f2
Remove unused FlexfecReceiveStream::Stats struct
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Bug: webrtc:14109
Change-Id: Ie06c267c15b21eff15803ead11b6deb661d17523
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262944
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org >
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36996}
2022-05-25 07:02:39 +00:00
ff0c0339a2
Update WebRTC code version (2022-05-25T04:04:41).
...
Bug: None
Change-Id: Ie02f3aa8db32e10c89be6880ddcfa3bf544a1f90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263466
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Cr-Commit-Position: refs/heads/main@{#36993}
2022-05-25 05:03:01 +00:00
11cf37c3ba
Remove ReceiveStream definition.
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Bug: webrtc:7484
Change-Id: I17c7617d14e28cbe4a54256ee11e9b3fd4346ec7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262961
Reviewed-by: Björn Terelius <terelius@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36986}
2022-05-24 14:53:24 +00:00
794c54faf0
Update WebRTC code version (2022-05-24T04:03:53).
...
Bug: None
Change-Id: I77db68cb4ba32316c528731e0a4da388100527e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263345
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Cr-Commit-Position: refs/heads/main@{#36981}
2022-05-24 05:52:56 +00:00
3176ef79e9
Rename AudioReceiveStream to AudioReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I22eaa7a9e082fc575cf7471d7a2f4f706564d54f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262805
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36965}
2022-05-23 08:44:26 +00:00
f8acaabe8b
Update WebRTC code version (2022-05-23T04:04:57).
...
Bug: None
Change-Id: Ieaba54afa2d7fe2f52346d388b879d4a98d9d16b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263090
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36961}
2022-05-23 05:16:29 +00:00
dddbbebe2b
Rename internal::AudioReceiveStream to AudioReceiveStreamImpl
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Bug: webrtc:7484
Change-Id: Id0836a7fdd6fabbdc9bdc3b15e9965d9102bffa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262803
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36959}
2022-05-22 12:22:18 +00:00
f6f4543304
Rename VideoReceiveStream to VideoReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I653cfe46486e0396897dd333069a894d67e3c07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262769
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36958}
2022-05-22 10:54:38 +00:00
d91996dd00
Update WebRTC code version (2022-05-22T04:05:03).
...
Bug: None
Change-Id: I15a804906a90a50f2ea083bcad0c18529e1ca3d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263061
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36957}
2022-05-22 05:18:28 +00:00
1def899931
Remove legacy (unused) config param: jitter_buffer_enable_rtx_handling
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Bug: none
Change-Id: I14164546950cc63c37e54544cdc80bfd4eddf211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262962
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36955}
2022-05-21 23:06:21 +00:00
e508ebc645
Update WebRTC code version (2022-05-21T04:04:39).
...
Bug: None
Change-Id: I072c54f3afd47f1cb3320dab9150a48dbc2091b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263021
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36954}
2022-05-21 05:29:51 +00:00
83830f316e
Delete TestListener and top-level thread wrapping.
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Instead use rtc::AutoThread in tests that need that.
Bug: webrtc:9714
Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36950}
2022-05-20 15:21:21 +00:00
6fb674ea5a
Rename MediaReceiveStream to MediaReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I0bc4bc57e8c4450c503ae4d5a41f9bbe243b00e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262768
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36947}
2022-05-20 13:17:52 +00:00
f2c710852e
Update WebRTC code version (2022-05-20T04:02:10).
...
Bug: None
Change-Id: I7e157362c58193fa64316f7bbe02da6ad94f1f06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262902
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36937}
2022-05-20 05:23:07 +00:00
65b2d8ad21
Move RunLoop test class to its own build target
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To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36932}
2022-05-19 15:51:39 +00:00
6c81a42eb1
Simulate generic dependency structure for VP8
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This will be used as a fall-back when the encoder adapter doesn't
provide any dependency structure. This ensures we can always generate a
dependency descriptor RTP header extension for VP8.
Before, when switching between encoder adapters where the old one
generated a dependency structure but the new one didn't we had to make
sure the structure was cleared so that packets weren't sent with the
dependency structure from the previous adapter. This will not be a
problem anymore since the new adapter will use the simulated dependency
structure.
Bug: b/227749056
Change-Id: I8463c48a9dcde4b8d32c519819dd8a92acd8e43b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262765
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org >
Reviewed-by: Erik Språng <sprang@webrtc.org >
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36930}
2022-05-19 11:53:08 +00:00
42cf83cc61
Update WebRTC code version (2022-05-19T04:04:47).
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Bug: None
Change-Id: I8eff6cd60b9416f81b32a1a51ca56286fedb7919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262736
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Cr-Commit-Position: refs/heads/main@{#36927}
2022-05-19 05:49:10 +00:00
0359ba2225
stats: add frame assembly time stats
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implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00
0601db9a48
Rename ReceiveStream to ReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I41176a66b8399f6c8cf568630f2808eb95cf6247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262767
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36917}
2022-05-18 07:26:50 +00:00
79dc0a223e
Update WebRTC code version (2022-05-18T04:01:44).
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Bug: None
Change-Id: Ib10b8b8236ab5905ed6c524d058a39a356eebebd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262732
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36915}
2022-05-18 05:16:30 +00:00
88af20356f
Use ProbeClusterConfig in BitrateProber from GoogCC
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Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36911}
2022-05-17 12:29:25 +00:00
641a1b11b6
Adopt absl::string_view in call/
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Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Ali Tofigh <alito@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36910}
2022-05-17 12:00:45 +00:00
1331c1821c
Reland: Update local_ssrc without needing to recreate video streams.
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This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
This is a reland of commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca
with an additional fix in Patchset 2. Another problem turned out to be
in RTCPReceiver, which is fixed in:
https://webrtc-review.googlesource.com/c/src/+/262663
Bug: webrtc:11993
Change-Id: I63c7cf62a6dd50f88b491fea3ba866697552ef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262665
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36907}
2022-05-17 10:59:54 +00:00
2e521f5a25
Update WebRTC code version (2022-05-17T04:04:40).
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Bug: None
Change-Id: I323023722ba3d5d467439a57cc23cb9f16b08b95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262724
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Cr-Commit-Position: refs/heads/main@{#36903}
2022-05-17 05:17:46 +00:00
c92ee5f3c3
Revert "Update local_ssrc without needing to recreate video streams."
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This reverts commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca.
Reason for revert: Checking if this is blocking the Chromium autoroller.
Original change's description:
> Update local_ssrc without needing to recreate video streams.
>
> This is comparable to this change done previously for for audio streams:
> https://webrtc-review.googlesource.com/c/src/+/222042
>
> Bug: webrtc:11993
> Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#36876}
Bug: webrtc:11993
Change-Id: I3a8d2f6a7e89b6784754d8e891a4e01479807c2d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262422
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com >
Cr-Commit-Position: refs/heads/main@{#36892}
2022-05-13 22:30:44 +00:00
f3f3a61167
Remove legacy PacedSender.
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The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36890}
2022-05-13 20:31:06 +00:00
16a8b25d80
Update local_ssrc without needing to recreate video streams.
...
This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
Bug: webrtc:11993
Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36876}
2022-05-13 10:08:54 +00:00
