webrtc-version-updater
206bdaf26c
Update WebRTC code version (2024-03-04T04:13:18).
...
Bug: None
Change-Id: Ie6d93e49c7ea04ab5f80ea6c17168919a2ab753f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341767
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Cr-Commit-Position: refs/heads/main@{#41855}
2024-03-04 05:20:38 +00:00
chromium-webrtc-autoroll
04d22681e3
Roll chromium_revision 0bfdc8c539..67f77562a2 (1267549:1267659)
...
Change log: 0bfdc8c539..67f77562a20bfdc8c539..67f77562a23915ccffa2..c06d7b5cb4e2900fac8e..ab0ead57afe61acf937c..174a3b4a8b93a213c07f..9cc615980b0bfdc8c539..67f77562a2https://webrtc-review.googlesource.com/c/src/+/341785
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Cr-Commit-Position: refs/heads/main@{#41854}
2024-03-04 02:45:22 +00:00
webrtc-version-updater
89e62f305a
Update WebRTC code version (2024-03-03T04:12:48).
...
Bug: None
Change-Id: I2fd7942657d24718c1baf8bac89ce4211a56cf55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341760
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@{#41853}
2024-03-03 05:46:17 +00:00
chromium-webrtc-autoroll
c8068f68f2
Roll chromium_revision 16ca06c8c8..0bfdc8c539 (1267445:1267549)
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Change log: 16ca06c8c8..0bfdc8c53916ca06c8c8..0bfdc8c539fef2b5e6b7..81977015e5410689e90a..e2900fac8e336e6a4e68..e61acf937c98921c2a0c..22d2e541be546c584d90..93a213c07f16ca06c8c8..0bfdc8c539https://webrtc-review.googlesource.com/c/src/+/341697
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2024-03-02 19:38:37 +00:00
webrtc-version-updater
77590862d5
Update WebRTC code version (2024-03-02T04:12:36).
...
Bug: None
Change-Id: I3cb435804e4510ccc7f15b45853faf212a911299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341690
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2024-03-02 05:58:58 +00:00
chromium-webrtc-autoroll
572ce2719c
Roll chromium_revision 6312fa2472..16ca06c8c8 (1267340:1267445)
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Change log: 6312fa2472..16ca06c8c86312fa2472..16ca06c8c80f6697fc2b..3915ccffa254f2661b52..410689e90a7563c75d12..336e6a4e68https://chromium.googlesource.com/catapult.git/+log/2ee42b4fee..f226e76aa5
* src/third_party/perfetto: 77ac4b7528..98921c2a0c158705d708..546c584d906312fa2472..16ca06c8c8https://webrtc-review.googlesource.com/c/src/+/341702
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2024-03-01 19:36:23 +00:00
chromium-webrtc-autoroll
59ed9e8ebe
Roll chromium_revision 38dcc53cf5..6312fa2472 (1267235:1267340)
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Change log: 38dcc53cf5..6312fa247238dcc53cf5..6312fa2472fa26aeb00d..fef2b5e6b7b484740dba..0f6697fc2b9ec2be606c..44a1b90ebb5d3c6792d9..54f2661b52bf93900a20..7563c75d121ac3eb7b98..fbb0301f1f76bb2afb8b..dda72ef32100e519d947..158705d70838dcc53cf5..6312fa2472https://webrtc-review.googlesource.com/c/src/+/341623
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Cr-Commit-Position: refs/heads/main@{#41849}
2024-03-01 02:50:18 +00:00
chromium-webrtc-autoroll
ede75295d4
Roll chromium_revision 1e40594b88..38dcc53cf5 (1267092:1267235)
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Change log: 1e40594b88..38dcc53cf51e40594b88..38dcc53cf53d0b3c7162..fa26aeb00dd49d4c013b..9ec2be606c9a0f787478..5d3c6792d935ff337157..bf93900a20https://aomedia.googlesource.com/aom.git/+log/a2d599c975..14010c6f0f
* src/third_party/perfetto: 3fa1408bbc..77ac4b7528d7f2f98a48..00e519d9471e40594b88..38dcc53cf5https://webrtc-review.googlesource.com/c/src/+/341660
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Cr-Commit-Position: refs/heads/main@{#41848}
2024-02-29 22:52:39 +00:00
chromium-webrtc-autoroll
e5ac106a35
Roll chromium_revision f770766245..1e40594b88 (1266950:1267092)
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Change log: f770766245..1e40594b88f770766245..1e40594b889f13d878d5..3d0b3c7162f81fcc51c8..d49d4c013b8d2ca7caa0..9a0f787478afe1d14b38..35ff337157546237e1bb..2a790a9f491553701a9f..3fa1408bbcf770766245..1e40594b88https://webrtc-review.googlesource.com/c/src/+/341622
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Cr-Commit-Position: refs/heads/main@{#41847}
2024-02-29 18:35:05 +00:00
chromium-webrtc-autoroll
015de612e9
Roll chromium_revision 248b5659e1..f770766245 (1266836:1266950)
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Change log: 248b5659e1..f770766245248b5659e1..f7707662457dfbdde7b6..9f13d878d5100a65f1dd..b484740dba53ae48db44..f81fcc51c86ac6be6e29..8d2ca7caa0dd5b48d517..afe1d14b38609cb8ef02..1553701a9ff9550c3f72..2d866a3d0776e998060b..d7f2f98a48248b5659e1..f770766245https://webrtc-review.googlesource.com/c/src/+/341549
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2024-02-29 14:31:56 +00:00
Philipp Hancke
a5cd6643f6
Add killswitch for receive-only setCodecPreferences change
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Adds a killswitch
WebRTC-SetCodecPreferences-ReceiveOnlyFilterInsteadOfThrow
to accompany the spec-change to throw when codec capabilities
are taken from the RtpSender instead of the RtpReceiver.
With the killswitch triggered, such codecs will be filtered.
BUG=webrtc:15396
Change-Id: I7d27111c72085eb7a7b2a1e66d0a08d12883ce17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341460
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41845}
2024-02-29 12:43:05 +00:00
Jan Grulich
16ac10d9f7
PipeWire camera: use length of device id instead display name
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We want to copy device id to _lastUsedDeviceName variable, but we use
length of display name instead of length of device id, which might be
longer than expected and we end up reading beyond the source string.
Bug: webrtc:15853
Change-Id: Id278ed7e361ead85475910adec18b9db51e6890b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341521
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41844}
2024-02-29 10:20:09 +00:00
Philipp Hancke
20a90295fc
sdp: set content to rejected if the list of common codecs is empty
...
which avoids throwing an error when using setCodecPreferences
to set a recvonly codec on a sendonly transceiver. See
https://github.com/w3c/webrtc-pc/issues/2936
BUG=webrtc:15396
Change-Id: I435a98c944ed2eeef87d9b8a7f791d095ec25502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41843}
2024-02-29 08:02:26 +00:00
chromium-webrtc-autoroll
e5c04cb080
Roll chromium_revision 19f774aa77..248b5659e1 (1266733:1266836)
...
Change log: 19f774aa77..248b5659e119f774aa77..248b5659e11490bf980a..7dfbdde7b663724e6b5f..100a65f1dd745b089be5..6ac6be6e29905b09eb52..dd5b48d517https://chromium.googlesource.com/catapult.git/+log/43f9843685..2ee42b4fee
* src/third_party/depot_tools: 50de666ba4..1ac3eb7b982739728a1b..76e998060b19f774aa77..248b5659e1https://webrtc-review.googlesource.com/c/src/+/341547
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Cr-Commit-Position: refs/heads/main@{#41842}
2024-02-29 07:08:13 +00:00
webrtc-version-updater
80000bc733
Update WebRTC code version (2024-02-29T04:06:33).
...
Bug: None
Change-Id: I0436dc89918ce5835430debf14355a0ff1877454
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341546
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Cr-Commit-Position: refs/heads/main@{#41841}
2024-02-29 05:35:07 +00:00
chromium-webrtc-autoroll
c935bb2141
Roll chromium_revision 07799e8663..19f774aa77 (1266593:1266733)
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Change log: 07799e8663..19f774aa7707799e8663..19f774aa7792f0d7aae7..1490bf980a443a035ed0..53ae48db448224e6bb45..745b089be542f33900fc..905b09eb52https://chromium.googlesource.com/catapult.git/+log/56ba67e870..43f9843685
* src/third_party/depot_tools: 6b84fbfb20..50de666ba42be4d2932f..609cb8ef020ac8cb8391..2739728a1b07799e8663..19f774aa77https://webrtc-review.googlesource.com/c/src/+/341544
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Cr-Commit-Position: refs/heads/main@{#41840}
2024-02-29 00:51:58 +00:00
chromium-webrtc-autoroll
3351c9c6b7
Roll chromium_revision f6623eb6d1..07799e8663 (1266346:1266593)
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Change log: f6623eb6d1..07799e8663f6623eb6d1..07799e866377fc7dc438..92f0d7aae7dd4da8c1d8..63724e6b5f135784cc4b..1db15eb4206df5e3a653..443a035ed0056ee59184..8224e6bb45dafe628de1..42f33900fc00915b6874..6b84fbfb2079a88d3393..512d2ed32faff3a0b23c..b5fe27de9315eeebda55..2be4d2932f57e626c12f..0ac8cb8391f6623eb6d1..07799e8663https://webrtc-review.googlesource.com/c/src/+/341542
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Cr-Commit-Position: refs/heads/main@{#41839}
2024-02-28 21:10:04 +00:00
Jeremy Leconte
3afa1b2ce8
Add a SimulcastStream::GetScalabilityMode2 method that returns an optional.
...
A call to GetScalabilityMode was added for logging purpose and causes an expectation failure for tests using 4 temporal layers.
Plan is to remove the old GetScalabilityMode and keep only the one that returns an optional.
Change-Id: I0e37a496bb621d9754d6572ef5838b58193aa183
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341520
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41838}
2024-02-28 17:38:46 +00:00
Danil Chapovalov
b9ce3b79fc
Delete deprecated VP8Decoder::Create
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Bug: webrtc:15791
Change-Id: Ic198706535da9f299735cd0a7bbf571cda643098
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340461
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41837}
2024-02-28 15:18:11 +00:00
chromium-webrtc-autoroll
95977d8a6a
Roll chromium_revision b2ff29df8d..f6623eb6d1 (1266241:1266346)
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Change log: b2ff29df8d..f6623eb6d1b2ff29df8d..f6623eb6d1d9194edfc2..77fc7dc4383d82e4a856..dd4da8c1d859ab976e2d..6df5e3a6536d4ac13817..056ee591849aa29ac83c..dafe628de1https://chromium.googlesource.com/catapult.git/+log/fa6a888a12..56ba67e870
* src/third_party/depot_tools: af26c1dfaa..00915b68742d8c8288d9..57e626c12fb2ff29df8d..f6623eb6d1https://webrtc-review.googlesource.com/c/src/+/341500
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Cr-Commit-Position: refs/heads/main@{#41836}
2024-02-28 12:32:13 +00:00
Harald Alvestrand
fb4ad29e3b
Continue breakup of media/rtc_media_base
...
Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
2024-02-28 12:29:54 +00:00
Danil Chapovalov
dcc1534764
Delete rtc::TaskQueue
...
All usage was updated to use TaskQueueBase interface directly bypassing rtc::TaskQueue wrapper
Bug: webrtc:14169
Change-Id: I1808afd363b50448d4014d8d8402fce41b16a3ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341082
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41834}
2024-02-28 10:22:49 +00:00
Philipp Hancke
bbff58d935
Introduce "well-known" SdpVideoFormat codecs
...
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
2024-02-28 06:57:10 +00:00
webrtc-version-updater
2825f0a7bb
Update WebRTC code version (2024-02-28T04:11:45).
...
Bug: None
Change-Id: I562e3fdf93233500b3f6e01149d63a33d3cc6459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341343
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@{#41832}
2024-02-28 05:39:36 +00:00
chromium-webrtc-autoroll
55d33f1f60
Roll chromium_revision aa1c478fe0..b2ff29df8d (1266091:1266241)
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Change log: aa1c478fe0..b2ff29df8daa1c478fe0..b2ff29df8d50deb986af..d9194edfc205bf67252f..6d4ac1381799b7011fc4..9aa29ac83chttps://chromium.googlesource.com/catapult.git/+log/7fccadad2a..fa6a888a12
* src/third_party/depot_tools: d972b831c3..af26c1dfaacd05247e5c..15eeebda557ff2c0ee3d..2d8c8288d9aa1c478fe0..b2ff29df8dhttps://webrtc-review.googlesource.com/c/src/+/341342
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Cr-Commit-Position: refs/heads/main@{#41831}
2024-02-28 04:38:03 +00:00
chromium-webrtc-autoroll
c27339fa23
Roll chromium_revision 511e819ce4..aa1c478fe0 (1265924:1266091)
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Change log: 511e819ce4..aa1c478fe0511e819ce4..aa1c478fe01cfc3d17b0..50deb986af8af52f161e..3d82e4a856f0f3d3742d..59ab976e2d6a2e50072e..05bf67252f935e0418b3..99b7011fc4https://chromium.googlesource.com/catapult.git/+log/6f4a0d6c87..7fccadad2a
* src/third_party/depot_tools: 4df61147ba..d972b831c35b35c9f06c..204deaa9c56427f365ba..cd05247e5c44a7afcd6d..7ff2c0ee3d511e819ce4..aa1c478fe0https://webrtc-review.googlesource.com/c/src/+/341361
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Cr-Commit-Position: refs/heads/main@{#41830}
2024-02-27 22:41:26 +00:00
Danil Chapovalov
c3d937b3e4
In RtpFrameReferenceFinder discard frames with too large spatial id
...
Bug: chromium:41495253
Change-Id: I681f64edfcba319ab9479a2ad10987452cf9b6d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341265
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41829}
2024-02-27 21:44:37 +00:00
Per K
8cd50cbbdc
Delete PacketTransportInternal::SignalReadPacket
...
And remove usage from tests.
Bug: webrtc:15368, webrtc:11943
Change-Id: I7f5fd6502287fb04f5f1612c7fde529996582581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341260
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41828}
2024-02-27 20:38:51 +00:00
chromium-webrtc-autoroll
a8c47276cb
Roll chromium_revision a44e4a5e32..511e819ce4 (1265726:1265924)
...
Change log: a44e4a5e32..511e819ce4a44e4a5e32..511e819ce40d5608905c..8af52f161e9eedca5b06..135784cc4b5661d22bf5..f0f3d3742d4686dee27f..6a2e50072e21431ba73b..935e0418b308b8dfd3a9..aff3a0b23c2778513f99..6427f365bacefc25bc1c..44a7afcd6da44e4a5e32..511e819ce4https://webrtc-review.googlesource.com/c/src/+/341360
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@{#41827}
2024-02-27 18:58:38 +00:00
Jan Grulich
334e9133dc
Video capture PipeWire: add support for DMABuf buffer type
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Announce that we support SPA_DATA_DmaBuf and tell PipeWire not to map
memory for us so we can handle it ourself, similar like we do in case of
screen sharing. This fixes an issue when a camera is already in use by
gstreamer (pipewiresrc), where DMABufs are used, and we try to share
same camera and get no content, as PipeWire doesn't want to mmap DMABuf
memory for us and we get NULL data pointers.
Firefox bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1876895
Bug: webrtc:15654
Change-Id: I788d8d12b2fcd5588329d7265e45b479f74bb628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338921
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41826}
2024-02-27 18:31:26 +00:00
Per K
9e0bf9b5c8
Propagate rtc::ReceivedPacket further in RtpTransport
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Bug: webrtc:15368
Change-Id: I4c8989a7b9efacbcc29c0c3331d8f4d7350774c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41825}
2024-02-27 17:46:18 +00:00
Danil Chapovalov
f7682f01bb
Delete RTCWrappedNativeVideoDecoder
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Instead implement creating native VideoDecoder via RTCNativeVideoDecoderBuilder protocol
Bug: webrtc:15791
Change-Id: Iea66d09e01eae3b064a2943932d9a3cd33e8d19c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340321
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41824}
2024-02-27 17:03:28 +00:00
Per K
f4aadf3774
Change RtpTransport and DsctTransport to receives packets through ReceivedPacketCallback
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Instead of using PacketTransportInternal::SignalReadPacket.
Bug: webrtc:15368
Change-Id: Icdc2d7f85df6db944f0ba0232891e6c5a8986a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340440
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41823}
2024-02-27 15:55:02 +00:00
Florent Castelli
524a06bc54
Change BuiltInNetworkBehaviorConfig.loss_percent to double
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This should allow greater precision in the lower ranges of packet loss.
Bug: chromium:41175925
Change-Id: Ia35059ad673a3782443b23772511b0b952b07ce7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341263
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41822}
2024-02-27 14:38:31 +00:00
chromium-webrtc-autoroll
e39f6fd7fd
Roll chromium_revision af00cd208a..a44e4a5e32 (1264869:1265726)
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Change log: af00cd208a..a44e4a5e32af00cd208a..a44e4a5e3265e713f121..1cfc3d17b0d50a0a69f0..0d5608905c90b471f681..9eedca5b06272a109a71..5661d22bf52f53c3ca75..4686dee27f646491e14a..21431ba73bhttps://chromium.googlesource.com/catapult.git/+log/4dc7721a14..6f4a0d6c87
* src/third_party/depot_tools: cb43b5d82d..4df61147ba61d95200e7..bddcd9f77baf29db7ec2..76bb2afb8b5a3d13ed42..08b8dfd3a9a7b3d968a3..5b35c9f06c3316c11240..d191c5f9844183dabcac..2778513f99ac20c40fa9..cefc25bc1caf00cd208a..a44e4a5e32https://webrtc-review.googlesource.com/c/src/+/341300
Owners-Override: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41821}
2024-02-27 13:31:12 +00:00
Erik Språng
2514dd7a20
Increase WebRTC default receive buffer size to 1MB.
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The previous default size was 256kB.
The increase reduces packet loss at very high/bursty receive rates.
Bug: chromium:41485050
Change-Id: I2cf24b14e704bfd855701461afd3060ac078df70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340340
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41820}
2024-02-27 12:35:45 +00:00
Christoffer Dewerin
7a008822f9
Disable checks for googletest for WebRTC
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Bug: chromium:326607005
Change-Id: Iaa97ef823509cd978ed462bcf8ea2945a7baf7ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341262
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#41819}
2024-02-27 12:05:12 +00:00
Philipp Hancke
51532fd355
Test handling of rejected m-lines without transport description
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adds a unit test for
https://webrtc-review.googlesource.com/c/src/+/340322
which is a single m-line variant of the original
fiddle that does not require renegotiation
BUG=chromium:326493639
Change-Id: Icc5ebb1dda6502b00828a77e13b9f5fc865d34c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340500
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41818}
2024-02-27 08:28:36 +00:00
Jan Grulich
058bfe3ae3
PipeWire capturer: set capturer as failed when session is closed
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Marking capturer as failed will indicate consumers will not be getting
any new frames by sending back ERROR_PERMANENT and let them know that
screencast can be stopped from their side. This will make screencast to
stop when a window we share is closed or when screencast is closed from
system tray.
Bug: chromium:40276865
Change-Id: Ia2c13461bd3126cab9c4838b8aa6840578562e9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339560
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41817}
2024-02-27 07:41:41 +00:00
Per K
8e137d0509
Replace use of SignalReadPacket in DtlsTransport
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Instead use PacketTransportInternal::NotifyPacketReceived
Bug: webrtc:15368
Change-Id: I70a83865c9b564429366bd297abc7dbd50da02e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340301
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41816}
2024-02-27 06:17:41 +00:00
webrtc-version-updater
09e81ccb27
Update WebRTC code version (2024-02-27T04:11:00).
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Bug: None
Change-Id: I39d2496e8a2354b7ad79c9f6dff6cd60a79e7351
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341181
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@{#41815}
2024-02-27 05:37:54 +00:00
Qiu Jianlin
c32a509da0
Export h.265 bitstream parser APIs.
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This exports neccessary API as dependency for h.265 parameter sets tracker to be submitted at CL:5307256.
Bug: webrtc:13485
Change-Id: I042599472f17d12ece4fa862c3715497502a5d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340004
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41814}
2024-02-27 01:36:42 +00:00
Danil Chapovalov
91ebd5fd12
Add missing absl::optional includes
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Bug: None
Change-Id: I4abece77b021a866175253cbb2bd212ff618910c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341022
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41813}
2024-02-26 18:21:16 +00:00
Christoffer Dewerin
7098d110e1
Revert "p2p: separate ICE tie breaker and foundation seed"
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This reverts commit d99499abbae94793a02944a1f28f7015816447f5.
Reason for revert: Breaks downstream projects and I can also repro locally when running the rtc_unittest test target (it does however pass in isolation indicating test cleanup/setup needs to be fixed)
Original change's description:
> p2p: separate ICE tie breaker and foundation seed
>
> BUG=webrtc:14626
>
> Change-Id: I189a708192c9cef0b50c3fcbe798b30376d3b547
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338982
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41806}
Bug: webrtc:14626
Change-Id: If45f8a33395c562c9388b3d3748e8566efa87ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341081
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41812}
2024-02-26 16:05:15 +00:00
Per K
14613639bf
Make PeerConnectionInteface methods pure virtual.
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Bug: none
Change-Id: I3fc8e6f87d3559544e8dfb7cacdd4d0d47ebad7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340143
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Original-Commit-Position: refs/heads/main@{#41782}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340961
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41811}
2024-02-26 13:01:27 +00:00
Danil Chapovalov
5261619ad2
Remove rtc::TaskQueue in AudioDeviceBuffer
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Instead stop/delete TaskQueueBase in destructor explicitly and explain potential race.
Bug: webrtc:14169
Change-Id: Ica7a78f149be11ba1a82cbf79d4244c918aa9d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335360
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41810}
2024-02-26 12:55:27 +00:00
Danil Chapovalov
3f7566abda
Cleanup rtc::TaskQueue in AsyncAudioProcessing
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use TaskQueueBase directly - rtc::TaskQueue wrapper adds no benefit here.
Bug: webrtc:14169
Change-Id: If3d4feb11ffa507919a8ce4d7545172a25f0aa86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335322
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41809}
2024-02-26 12:22:56 +00:00
Markus Handell
97df932ecc
Remove multiplex codec.
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The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
2024-02-26 11:26:04 +00:00
Philipp Hancke
db2f52ba88
Reland "Make setCodecPreferences only look at receive codecs"
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This is a reland of commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b
after updating the WPT that broke on Mac.
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I0c7b17f00de02286f176b500460e17980b83b35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339541
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41807}
2024-02-26 10:52:23 +00:00
Philipp Hancke
d99499abba
p2p: separate ICE tie breaker and foundation seed
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BUG=webrtc:14626
Change-Id: I189a708192c9cef0b50c3fcbe798b30376d3b547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338982
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41806}
2024-02-26 07:47:28 +00:00
