Danil Chapovalov
a99bf7fa84
Delete deprecated AudioDecoderOpus::MakeAudioDecoder
...
Bug: webrtc:356878416
Change-Id: I2dc830c46fb5eece3b93a0354fd1e8a323a5e2ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360841
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42880}
2024-08-29 08:55:27 +00:00
Ho Cheung
f2487c0d4f
[audio] Adjust the order of some definitions in audio_processing
...
Moving defines before they are used with
unique_ptr allows to compile this file with
-std=c++2b flag.
Bug: webrtc:339074792
Change-Id: Ie7c37ab724800aea4545b72b4d2a779e12a2026a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360860
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42879}
2024-08-29 07:08:12 +00:00
webrtc-version-updater
45af5a81f0
Update WebRTC code version (2024-08-29T04:04:15).
...
Bug: None
Change-Id: I6173a9637c8c2b2558dc832f6875320ce3c7a0a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360824
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@{#42878}
2024-08-29 06:11:38 +00:00
chromium-webrtc-autoroll
2de37ef650
Roll chromium_revision c3a359139e..10ff7fa1e3 (1348059:1348232)
...
Change log: c3a359139e..10ff7fa1e3c3a359139e..10ff7fa1e363a173a5c3..053a09c8e2e5797c58d8..d8d1de384a8eccd23ec5..f39c5823f1b0f720d94c..e0f26cc194https://boringssl.googlesource.com/boringssl.git/+log/f1b043c283..d3c1afdc1b
* src/third_party/depot_tools: 1f6ef165b7..aaaca2863c05564d1572..700c45efcef8e9d8a29c..445761993ec3a359139e..10ff7fa1e3https://webrtc-review.googlesource.com/c/src/+/360823
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@{#42877}
2024-08-28 21:30:56 +00:00
chromium-webrtc-autoroll
2e1068840e
Roll chromium_revision ab7255fe8a..c3a359139e (1347197:1348059)
...
Change log: ab7255fe8a..c3a359139eab7255fe8a..c3a359139ef86fda1657..4b293ff11b5597e471cb..63a173a5c3e4421d7122..e5797c58d855fd0c28ae..8eccd23ec531d0a5b358..b0f720d94chttps://chromium.googlesource.com/catapult.git/+log/4917becfcf..c903c60cb7
* src/third_party/depot_tools: f5e1092339..1f6ef165b714d16cd143..3f8a122bfa312a9004c1..f00fa3ce7420ed12d82f..05564d157288d25846bc..f8e9d8a29cab7255fe8a..c3a359139ehttps://webrtc-review.googlesource.com/c/src/+/360880
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@{#42876}
2024-08-28 17:39:50 +00:00
Jakob Ivarsson
44df591447
Use NetEq::GetCurrentDecoderFormat in AcmReceiver.
...
This replaces the payload type tracking in AcmReceiver with the one in
NetEq and should be a noop.
Bug: None
Change-Id: Iaf124b5e56a646f994b5c2af65d349ede550b7fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360840
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42875}
2024-08-28 17:33:36 +00:00
Danil Chapovalov
4c862e781b
Implement Create instead of MakeAudioDecoder in AudioDecoderFactory template
...
Bug: webrtc:356878416
Change-Id: Iae9369eb6a2ae09a707854a18d909eed453bbfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359960
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42874}
2024-08-28 14:18:22 +00:00
Jakob Ivarsson
32dd2ed745
Improve NetEq simulation frame size estimation.
...
This fixes an edge case where the frame size changes for a DTX packet.
We should avoid having the frame size larger than the timestamp gap.
Bug: None
Change-Id: I0a384cfb06f5aebc1654c1e3d127541fd24e05c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360722
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42873}
2024-08-28 13:09:23 +00:00
Jakob Ivarsson
b6046aece2
Add NetEq API to get info about the current decoder.
...
This is currenly tracked in both AcmReceiver and NetEq. Adding this API
enables us to have it in just one place.
Bug: None
Change-Id: Ia537f87f36b0aedf19c00a57bd6cec4425a49df1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360743
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#42872}
2024-08-28 12:50:50 +00:00
Jeremy Leconte
c22a1ae9a1
Fix linux_more_configs mb config.
...
Change-Id: I5b882672e0d57b462f275918a1c4cbd28559fe7b
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360800
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42871}
2024-08-28 12:30:24 +00:00
Emil Vardar
572280fe6a
Remove redundant mapping.
...
It looks like the map `content_specific_stats_` is just carried over into `aggregated_stats` without doing any important processing. This seems to be redundant and hence, removing it in this CL.
Bug: None
Change-Id: Ia4a5de03999ecd33d387014928cba322bb11ee93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360745
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42870}
2024-08-28 11:44:43 +00:00
Erik Språng
54559d354e
Fix formatting for corruption detection header explainer.
...
Bug: webrtc:358039777
Change-Id: Ic9e860fbfdaf0bfe4b3e1d29f75913fb8d1080d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360742
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42869}
2024-08-27 15:18:27 +00:00
Per K
b60f0ffbce
Dont signal ReadyToSend in RtpTransport::SendPacket
...
Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN.
ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail.
Changing the state in the context of sending packets may cause recursive
calls and seems to cause problems with incorrect states.
It is simpler if RtpTransport::SendPacket ignore these failures and
upper layers treat these lost packets similar to if the packets had been
lost via UDP.
For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/.
Bug: webrtc:361124449 b/359989715
Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42868}
2024-08-27 14:16:53 +00:00
Emil Vardar
3f1e51d599
Aggregate and log corruption score.
...
Bug: webrtc:358039777
Change-Id: I4dade8e6daecf41e5b1f156416935c320f513d0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359160
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Cr-Commit-Position: refs/heads/main@{#42867}
2024-08-27 13:50:35 +00:00
Jeremy Leconte
0a8204b392
Set libsrtp_build_boringssl to false in 'no_build_ssl'.
...
Change-Id: Ib9023103d12f14134859b02a29490fdad9795ed6
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360744
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42866}
2024-08-27 13:48:11 +00:00
Per K
6db0db5cfd
Ensure TCPPort is notified of sent packets after reconnect
...
This fix an issue where the socket does not notify the port of sent packets after that the TCPConnection has opened a new socket. TcpConnection will open a new socket if the TCP connection has been closed.
Bug: webrtc:361124449 b/359989715
Change-Id: Id33c5fc5292ee7d1c2d1cad6c373e2d4355d4fe1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360260
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42865}
2024-08-27 12:29:52 +00:00
Danil Chapovalov
6bed21c811
Extend objc RTCVideoCodecInfo to include scalability modes
...
To match SdpVideoFormat c++ type
Bug: b/299588022
Change-Id: I67444cdb500b75fe43e3dfec47cc1333783897c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358866
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42864}
2024-08-27 09:40:36 +00:00
Jeremy Leconte
67ed656b75
Roll chromium_revision 30454db4a5..ab7255fe8a
...
Add cipd suffix to third_party/r8 and third_party/r8/d8.
Change-Id: Idc47534199a83b2859786c5040aabcf456dcbbb0
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360741
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#42863}
2024-08-27 09:07:12 +00:00
Erik Språng
c1a0d233d0
Update explainer text for corruption detection header extension.
...
Bug: webrtc:358039777
Change-Id: I6a1cffc2a5797d154bfecb50c60b4c05d4943426
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42862}
2024-08-27 08:34:31 +00:00
Fanny Linderborg
fd6f4b4e51
Add the corruption detection extension to RTPExtensionType
...
Bug: webrtc:358039777
Change-Id: Ib825593e5c37beb0cba3190c1d3bdcf1c9d957cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360144
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42861}
2024-08-27 08:27:20 +00:00
Christoffer Jansson
ad1775634f
Re-enable ApiCallDurationTest
...
Bug: b/260832909
Change-Id: I2a3e7651fbeba7a33a52a7139496b631cd2a446a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285469
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42860}
2024-08-27 08:24:47 +00:00
Harald Alvestrand
90e0829c59
Add test for PR-Answer functionality
...
Bug: None
Change-Id: I29bf1e40d47361917eb6f52424df23f7697bde0d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360721
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42859}
2024-08-27 08:17:32 +00:00
Mirko Bonadei
fd90f1aa05
Add Security Critical field to README.chromium.
...
As per [1], WebRTC qualifies as a security critical Chromium dependency.
[1] - https://chromium.googlesource.com/chromium/src/+/HEAD/docs/adding_to_third_party.md#add-a-readme_chromium
Bug: chromium:362396795
Change-Id: Ic4fc113cc2e3ec1df1c605698e1b3e83111aa14b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360740
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42858}
2024-08-27 07:38:26 +00:00
Philipp Hancke
06a49f02bd
build: add options to configure libsrtp for boringssl or other libraries
...
Depends on
https://webrtc-review.googlesource.com/c/src/+/359928
BUG=webrtc:42234521,webrtc:42224104
Change-Id: I0d6335aa5fb3f090c781bed234ed34d6c98ec299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359928
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42857}
2024-08-27 07:17:52 +00:00
Jeremy Leconte
a46f10329c
Re-enable iOS simulator from CQ and LKGR.
...
Change-Id: I734f2b6b5f24de766c933c130bf9f89def0cb8cc
Bug: b/357552398
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360720
Auto-Submit: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42856}
2024-08-27 06:05:24 +00:00
webrtc-version-updater
1d6ad045e4
Update WebRTC code version (2024-08-27T04:03:09).
...
Bug: None
Change-Id: Ia0a1d6dd6453a6242aebd8a57cbd303b82424154
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360642
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42855}
2024-08-27 06:02:14 +00:00
chromium-webrtc-autoroll
c6b556fadd
Roll chromium_revision cb10943d61..30454db4a5 (1346705:1346833)
...
Change log: cb10943d61..30454db4a5cb10943d61..30454db4a54434cfea96..47e1ac1df99c77ec35a2..e4421d7122e981b26186..0a53e28d00c677a5e718..994a093f09f92c96550a..d2612e1c3f6071843cd8..d8d9de41d77a7e64c8b5..000edd791486262cf454..88d25846bccb10943d61..30454db4a5https://webrtc-review.googlesource.com/c/src/+/360641
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@{#42854}
2024-08-26 18:51:51 +00:00
Harald Alvestrand
84ce5453ad
Reland "Add PT lookup function to JsepTransportController"
...
This reverts commit 0e3a3266afc50218747134bec7c40f1c6e82ab19.
Reason for revert: Ancestor CL fixed
Original change's description:
> Revert "Add PT lookup function to JsepTransportController"
>
> This reverts commit d178532ff9416f8b4272b9b8622afa9bab2ed558.
>
> Reason for revert: break pw-answer
>
> Original change's description:
> > Add PT lookup function to JsepTransportController
> >
> > Bug: webrtc:360058654
> > Change-Id: I9db58bf872f8659622e9f626fc21ce84993cfdfb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360143
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42829}
>
> Bug: webrtc:360058654
> Change-Id: Ic082dd3e86ed11d05b65710463fa9e57715bf07a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360360
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Jonas Oreland <jonaso@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42832}
Bug: webrtc:360058654
Change-Id: Ice9c118f9a5d4e0fa2cff89f504a25b80ec625ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360662
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42853}
2024-08-26 17:24:15 +00:00
Jeremy Leconte
37bd18fa48
Roll chromium_revision ef49a3ba49..cb10943d61 (1344824:1346705)
...
Change log: ef49a3ba49..cb10943d61ef49a3ba49..cb10943d61033588ef80..0db01afd902fb4df2b33..96625099e82470cca5b2..4434cfea96431d4c67ad..9c77ec35a2489a5b43e4..e981b26186eb289deb6a..c677a5e718https://boringssl.googlesource.com/boringssl.git/+log/11f334121f..f1b043c283
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/48294e2bd1..4917becfcf
* src/third_party/dav1d/libdav1d: 5ef6b241f0..472b31f83804a2e71a92..f5e1092339https://aomedia.googlesource.com/aom.git/+log/68bc71348b..35c90ab674
* src/third_party/libc++/src: 2557638e53..6071843cd8ccfbe1c82a..4426a8da65116c20dae6..dc70138c3e428f3104fa..312a9004c14a7ddbf3bf..20ed12d82f2449f923ca..86262cf454ef49a3ba49..cb10943d61https://webrtc-review.googlesource.com/c/src/+/360701
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42852}
2024-08-26 16:09:25 +00:00
Danil Chapovalov
c54c85fe8f
Attach Mid/Rid RTP header extension to pure padding packets
...
same as they attached to other packets.
Otherwise there is risk that ssrc will be acked after few initial pure padding packets are sent, before remote endpoint seen any mid or rid attached.
Bug: b/361257385
Change-Id: I695b379221debe2518ad33d13d65620877f0b2a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42851}
2024-08-26 16:00:16 +00:00
Benjamin Williams
ab009c27b4
Refactor WebRTC self assignments in if clauses
...
This change refactors existing self-assignments within if clauses across
the WebRTC codebase.
*Why:*
- Bug Prevention: Assignments within conditionals are frequently
unintended errors, often mistaken for equality checks.
- Clearer Code: Separating assignments from conditionals improves code
readability and reduces the risk of misinterpretation.
Change-Id: I199dc26a35ceca109a2ac569b446811314dfdf0b
Bug: chromium:361594695
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360460
Reviewed-by: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42850}
2024-08-26 15:56:43 +00:00
Harald Alvestrand
9e8652853e
Reland "Add first iteration of PayloadTypePicker.SuggestPayloadType"
...
This reverts commit 2e376cd36dc4beb224c2a4b858841a6b46a1c5df.
Reason for revert: Ancestor CL fixed.
Original change's description:
> Revert "Add first iteration of PayloadTypePicker.SuggestPayloadType"
>
> This reverts commit e2869de9efb8633e51815061242fe5eefd43dad7.
>
> Reason for revert: pr-answer
>
> Original change's description:
> > Add first iteration of PayloadTypePicker.SuggestPayloadType
> >
> > Bug: webrtc:360058654
> > Change-Id: I8f9242a97dc871a39ae72f325b8ca039b2285bae
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360061
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42828}
>
> Bug: webrtc:360058654
> Change-Id: I3a6b20595aa8420983f692048a8abdb254bf20d8
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360343
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@google.com>
> Cr-Commit-Position: refs/heads/main@{#42833}
Bug: webrtc:360058654
Change-Id: I09c16b50abe85f1423c449190e74b63654158322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360681
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42849}
2024-08-26 15:47:52 +00:00
Johannes Kron
0b91688817
Mark EncodedImage::{Set, Is}AtTargetQuality() as deprecated
...
The "at target quality" attribute is no longer set to the encoded
image in VideoStreamEncoder, see
https://webrtc-review.googlesource.com/c/src/+/359640
Mark the attribute as deprecated to avoid new dependencies and prepare
for deletion.
Bug: chromium:359410061
Change-Id: Id5a98ec9d2068099cb75a70b849bbf1c77feabb3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42848}
2024-08-26 14:48:29 +00:00
Harald Alvestrand
5308652c73
Reland "Add recording of PT->Codec mappings on setting SDP for transport"
...
This reverts commit 6793f831ffdc598e12aced80a4d97956ca50e436.
Reason for revert: Removed the check that caused the error.
Original change's description:
> Revert "Add recording of PT->Codec mappings on setting SDP for transport"
>
> This reverts commit 15717236c8621cb684bb7753acfedbf34d931c80.
>
> Reason for revert: pr-answer
>
> Original change's description:
> > Add recording of PT->Codec mappings on setting SDP for transport
> >
> > Bug: webrtc:360058654
> > Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Florent Castelli <orphis@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42819}
>
> Bug: webrtc:360058654
> Change-Id: I1fea51b3a0cecfa7e7de75f94f47a85fa064be59
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360380
> Reviewed-by: Jonas Oreland <jonaso@google.com>
> Commit-Queue: Jonas Oreland <jonaso@google.com>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42835}
Bug: webrtc:360058654
Change-Id: I2b60ccd60df3bacbeecd848c3cb86f6725b1505a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42847}
2024-08-26 11:11:43 +00:00
Victor Boivie
7348f820a9
dcsctp: Re-add lost validating in test case
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This was unintentionally removed in change
https://webrtc-review.googlesource.com/c/src/+/359681 due to a dirty
workspace.
Re-adding it.
Bug: None
Change-Id: Icff55b7a656ed9b504b0e10e7aeb947e8df78e85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360540
Auto-Submit: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42846}
2024-08-26 09:22:13 +00:00
Christoffer Dewerin
b4dc789c1f
Fix incorrect target for hamcrest and aapt2 and add back icu4j
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No-try: true
Bug: None
Change-Id: I51af2d90ca6a859247848ce26fcdb1ed8a252fae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360580
Auto-Submit: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42845}
2024-08-26 08:50:59 +00:00
Christoffer Dewerin
fc9d0cfb76
Remove deprecated DEPS
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RollError: WebRTC DEPS entries are missing from Chromium: ['src/third_party/android_build_tools/aapt2', 'src/third_party/hamcrest', 'src/third_party/icu4j', 'src/third_party/kotlin_stdlib'].
Remove them or add them to either WEBRTC_ONLY_DEPS or DONT_AUTOROLL_THESE.
Bug: None
Change-Id: I69b4912e13cb7b84ffd822856e2273594a52717c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360420
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Auto-Submit: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42844}
2024-08-26 07:11:46 +00:00
Ho Cheung
5b47a7a149
[rtc] Adjust the sequence of rtc::Network definition
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rtc::Network must be defined before classes that
use it as an instance variable.
Bug: webrtc:360158397
Change-Id: I5ca0acbc70cb5c27318d0ad01081b10b0b4dbff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360440
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Ho Cheung <hocheung@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42843}
2024-08-26 07:04:16 +00:00
Philipp Hancke
4f1dcd9d00
rename shadowing variable "offer" in unit test
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BUG=None
Change-Id: I34a17010d5ff02d0b8fad27ad0a69bc0e26b8c13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360300
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42842}
2024-08-26 05:39:55 +00:00
webrtc-version-updater
08cdf77208
Update WebRTC code version (2024-08-26T04:05:49).
...
Bug: None
Change-Id: I75e2f40fce411b9b786a301d0915992e7271634a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360560
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@{#42841}
2024-08-26 05:17:27 +00:00
webrtc-version-updater
d4e8e61dbb
Update WebRTC code version (2024-08-25T04:07:14).
...
Bug: None
Change-Id: I6613e7e162da2c19b26d21db0b94c8d25ee3cd96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360521
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@{#42840}
2024-08-25 06:12:01 +00:00
webrtc-version-updater
5a6a8fe6b3
Update WebRTC code version (2024-08-24T04:06:47).
...
Bug: None
Change-Id: I8d58bb9ba2343a7acc8834222b5aa7d34f777af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360520
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@{#42839}
2024-08-24 06:15:30 +00:00
Peter Kasting
b92345615e
[jumbo] Add begin()/end() to EncodedImage[BufferInterface].
...
This allows these types to meet the requirements of e.g.
std::ranges::range, which is necessary for them to work with the
std::span range constructor, or the "non-legacy" constructor for
Chromium's base::span.
Bug: none
Change-Id: Ia51c17690c785e0714c36d237094877129e0cbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358844
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42838}
2024-08-23 19:28:27 +00:00
Christoffer Dewerin
7e37e5fd5a
Use xcode 16 for iOS debug simulators + fix version
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Bug: chromium:361610467
Change-Id: I6ec53d7ca11c1537e8a59fdbc4a8ec277e39c626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360401
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Christoffer Dewerin <jansson@google.com>
Auto-Submit: Christoffer Dewerin <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#42837}
2024-08-23 13:50:22 +00:00
Fan Zhou
8771cf470d
Allow gap on packet buffer fix with GFD
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The current solution does not work for GFD since GFD is only parsed from the first packet of the frame. As a result, to access the generic information, we have to check every packet when traversing the packet buffer to find the first packet of frame. This fix is necessary to ensure temporal scaling works correctly with GFD.
Bug: webrtc:42225186
Change-Id: Iadda4ec690deab62c32eb6101583e6a6d75cfeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344840
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42836}
2024-08-23 09:41:14 +00:00
Jonas Oreland
6793f831ff
Revert "Add recording of PT->Codec mappings on setting SDP for transport"
...
This reverts commit 15717236c8621cb684bb7753acfedbf34d931c80.
Reason for revert: pr-answer
Original change's description:
> Add recording of PT->Codec mappings on setting SDP for transport
>
> Bug: webrtc:360058654
> Change-Id: I2aa5e0058346cd3fcda47a8ea5115848fbc4f3e2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360041
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42819}
Bug: webrtc:360058654
Change-Id: I1fea51b3a0cecfa7e7de75f94f47a85fa064be59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360380
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Jonas Oreland <jonaso@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42835}
2024-08-23 08:56:51 +00:00
Dan Tan
43c0cf9cf8
Support borrowing of underused audio bitrate.
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Controlled via added field trial WebRTC-ElasticBitrateAllocation.
Bug: webrtc:350555527
Change-Id: If57552144bd4a50421d618fd8bdab31d7c4afc35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359506
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#42834}
2024-08-23 07:44:45 +00:00
Jonas Oreland
2e376cd36d
Revert "Add first iteration of PayloadTypePicker.SuggestPayloadType"
...
This reverts commit e2869de9efb8633e51815061242fe5eefd43dad7.
Reason for revert: pr-answer
Original change's description:
> Add first iteration of PayloadTypePicker.SuggestPayloadType
>
> Bug: webrtc:360058654
> Change-Id: I8f9242a97dc871a39ae72f325b8ca039b2285bae
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360061
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42828}
Bug: webrtc:360058654
Change-Id: I3a6b20595aa8420983f692048a8abdb254bf20d8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360343
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#42833}
2024-08-23 07:23:54 +00:00
Jonas Oreland
0e3a3266af
Revert "Add PT lookup function to JsepTransportController"
...
This reverts commit d178532ff9416f8b4272b9b8622afa9bab2ed558.
Reason for revert: break pw-answer
Original change's description:
> Add PT lookup function to JsepTransportController
>
> Bug: webrtc:360058654
> Change-Id: I9db58bf872f8659622e9f626fc21ce84993cfdfb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360143
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42829}
Bug: webrtc:360058654
Change-Id: Ic082dd3e86ed11d05b65710463fa9e57715bf07a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360360
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jonas Oreland <jonaso@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42832}
2024-08-23 07:02:50 +00:00
webrtc-version-updater
a691309b23
Update WebRTC code version (2024-08-23T04:07:24).
...
Bug: None
Change-Id: Iec1a8c6c89374076ae6a6cc1dead928a5f67bd90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360320
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@{#42831}
2024-08-23 06:20:58 +00:00
