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
...
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
...
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
...
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
...
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
...
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
...
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
...
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.
...
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>
Commit-Queue: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#42831}
2024-08-23 06:20:58 +00:00
Philipp Hancke
b31ade36ff
stun/turn: suppress icecandidateerror for incompatible address family
...
Suppresses the ice candidate error callback when the STUN/TURN server
address family is not compatible with the local candidate address family.
This is similar to not pairing between candidates that have different
incompatible address families as described in
https://datatracker.ietf.org/doc/html/rfc5245#section-5.7.1
The spec actually says to emit the 701 error if *no* host candidate is able to reach the server:
https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectioniceerrorevent-errorcode
Also use the same (spec) error code for STUN and TURN, see
https://github.com/webrtc/samples/issues/1215 (error 600 for TURN)
https://github.com/webrtc/samples/issues/1227 (error 701 with AF mismatch)
Drive-by: misc logging fixes
BUG=webrtc:359404135
Change-Id: I99574b7b2b79986a52ab38a7fa58ea1bebab954c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358961
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42830}
2024-08-22 21:33:45 +00:00
Harald Alvestrand
d178532ff9
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}
2024-08-22 12:11:30 +00:00
Harald Alvestrand
e2869de9ef
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}
2024-08-22 11:05:17 +00:00
Helmut Januschka
abb638899e
remove deprecated <codecvt>
...
Bug: chromium:358567090
Change-Id: I0480e8962ee38425406ad167f43d666b57418f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360142
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Helmut Januschka <helmut@januschka.com>
Cr-Commit-Position: refs/heads/main@{#42827}
2024-08-22 10:37:00 +00:00
Fanny Linderborg
c03edf6096
Add missing includes and remove unused includes
...
Unused includes in header files are not removed.
Bug: webrtc:358039777
Change-Id: I4586971cd33ff76cac2f869bcdfb063c31e9a7a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42826}
2024-08-22 08:53:12 +00:00
Denis Genestier
5e70fd3d8f
fix of a compilation error in Visual Studio 2022 due to a warning C4244.
...
Bug: webrtc:353294048
Change-Id: I8c40a9df06464b9ffd7112fd7d070dee2d0e5ca4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360140
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42825}
2024-08-22 08:51:50 +00:00
webrtc-version-updater
bbd1467126
Update WebRTC code version (2024-08-22T04:09:17).
...
Bug: None
Change-Id: Ia279e9aad972f5a44bbaed604db02b09c9dc1262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360183
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@{#42824}
2024-08-22 06:11:24 +00:00
chromium-webrtc-autoroll
69f79166c1
Roll chromium_revision bb795520d5..ef49a3ba49 (1344631:1344824)
...
Change log: bb795520d5..ef49a3ba49bb795520d5..ef49a3ba4979aeed60c4..033588ef80b8f47779f5..2fb4df2b338bf8f9a09f..431d4c67adb941ee9cc5..489a5b43e4ac1cde9015..eb289deb6a1420017c7f..6b0c5b7ee18d63401473..4a7ddbf3bf2eafa58896..2449f923cabb795520d5..ef49a3ba49https://webrtc-review.googlesource.com/c/src/+/360181
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@{#42823}
2024-08-21 17:14:20 +00:00
Henrik Boström
da72666d94
Support standard simulcast with requested_resolution.
...
According to spec, if you ask for three encodings you get three
encodings (duh). But according to legacy code, if you ask for three
encodings AND codec is VP9, then surely you meant a single encoding that
is kSVC where the other encodings influence the scalability mode of the
first encoding.
Standard simulcast support in VP9 was shipped as an opt-in feature where
you have to specify `scalability_mode` and `scale_resolution_down_by` in
order to let WebRTC know that you want to disable the legacy path.
But `scale_resolution_down_by` is not the only way to configure
resolution, there is also the `requested_resolution` code path. This CL
adds standard simulcast support for this code path as well.
Prior to this change, our parameterized test would have passed in VP8
but failed in VP9. With this change the test passes for all codecs.
Bug: webrtc:361124448
Change-Id: Ic5a7136de8abf430813fd01342862775fca145fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42822}
2024-08-21 09:35:52 +00:00
chromium-webrtc-autoroll
90bd5003a2
Roll chromium_revision 3484724f00..bb795520d5 (1344182:1344631)
...
Change log: 3484724f00..bb795520d53484724f00..bb795520d5e499c7d661..79aeed60c4b2b185d91d..b8f47779f5d6d7cf19f0..2470cca5b27bcf347b5a..8bf8f9a09fc83d4fc180..b941ee9cc5a1b5b86212..ac1cde9015https://chromium.googlesource.com/catapult.git/+log/98d1cbf9f9..48294e2bd1
* src/third_party/libc++/src: 627bf9cce8..2557638e53c0a70944d2..8d6340147341a224862f..2eafa588963484724f00..bb795520d5https://webrtc-review.googlesource.com/c/src/+/359993
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@{#42821}
2024-08-21 09:19:24 +00:00
Danil Chapovalov
1f261028c8
Adjust fuzzers group to respect build variables
...
Avoid including targets that might not be present.
Bug: None
Change-Id: I439a3ff4e34337ecd64c6986edffcaa0843611cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360121
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42820}
2024-08-21 09:11:46 +00:00
Harald Alvestrand
15717236c8
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}
2024-08-21 09:06:51 +00:00
Corby Hoback
fea60ef8e7
Fixed issue with missing network interfaces on iOS
...
Fixed issue with network interfaces due to a missing return value in the
"nw_path_enumerate_interfaces(...)" block. Exposed in iOS 18,
RTCNetworkMonitor::initWithObserver will only enumerate the first
interface, instead of all device interfaces
Bug: webrtc:359245764
Change-Id: Ifb9f28c33306c0096476a4afb0cdb4d734e87b2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359541
Auto-Submit: Corby <corby.hoback@gmail.com>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42818}
2024-08-21 06:07:06 +00:00
webrtc-version-updater
2efd4fd7f3
Update WebRTC code version (2024-08-21T04:04:12).
...
Bug: None
Change-Id: Iada282369fe98dd6782babbd52ab4f8df8db877d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359930
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@{#42817}
2024-08-21 05:55:33 +00:00
chromium-webrtc-autoroll
c1a7827084
Roll chromium_revision f031fbef87..3484724f00 (1343352:1344182)
...
Change log: f031fbef87..3484724f00f031fbef87..3484724f00c79f2cf1aa..e499c7d6614a4042e134..b2b185d91d0d282e9aae..7bcf347b5a9a7b9f09db..c83d4fc18024ddd89921..a1b5b86212https://chromium.googlesource.com/catapult.git/+log/799e21b232..98d1cbf9f9
* src/third_party/depot_tools: caa77da956..04a2e71a92https://aomedia.googlesource.com/aom.git/+log/f6b976de02..68bc71348b
* src/third_party/libc++abi/src: eb6567388e..a834cb25398cd7489ada..c0a70944d279b0e0e577..41a224862ff031fbef87..3484724f00https://webrtc-review.googlesource.com/c/src/+/359924
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@{#42816}
2024-08-20 17:06:16 +00:00
Guy Hershenbaum
f009e38fe0
Fix AudioSendStream reconfigure - stop processing during unconfigured state
...
When Reconfiguring there's a call to ResetSenderCongestionControlObjects followed by a later call to
RegisterSenderCongestionControlObjects which happen on the worker thread, while enqueuing packets is
happening on a different thread.
If packets are enqueued in between these calls we get a null dereference of the `rtp_packet_pacer_`
This change fixes it by pausing processing of incoming audio in the interim state
Bug: webrtc:358290775
Change-Id: I77cebfb131545ce2a6fdb26105dd999da3f7c443
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359080
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42815}
2024-08-20 16:22:04 +00:00
Danil Chapovalov
f2d31361d9
Reland "Include fuzzers to build by default"
...
This reverts commit 06391decf8397849f4c0dd9468612caa8f111185.
Reason for revert: [will] include change to address previous breakage
Original change's description:
> Revert "Include fuzzers to build by default"
>
> This reverts commit e777c656db6c20329529ff301805d77dba78e7c8.
>
> Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux-libfuzzer-asan-rel/1941368/overview
> Original change's description:
> > Include fuzzers to build by default
> >
> > Compilation errors in fuzzers are often overlooked when building locally
> > adding fuzzers (when configured) to build automatically helps to detect errors earlier.
> >
> > list of all fuzzers was generated with a command
> > gn ls out/Default "//test/fuzzers:*" | grep "fuzzer$" | sed 's/\/\/test\/fuzzers\(.*\)/"\1",/'
> >
> > Bug: webrtc:42223576
> > Change-Id: I6988e96f521a198657833e666428377d0851e1d4
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359762
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#42794}
>
> Bug: webrtc:42223576
> Change-Id: I273b96b1b48053a4895209710a8cabd652d9c61e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359881
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Owners-Override: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42803}
Bug: webrtc:42223576
Change-Id: I43ec20dff79eba23b292b6d9096d9136ca59bfb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359961
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42814}
2024-08-20 15:50:14 +00:00
Christoffer Dewerin
a6fad74043
Add missing optional deps
...
Bug: webrtc:42226242
Change-Id: If09580139acb52b11ac4827f68aba46929cc5755
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360080
Commit-Queue: Christoffer Dewerin <jansson@google.com>
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@{#42813}
2024-08-20 14:50:36 +00:00
Jeremy Leconte
c478ff6a49
Sync bot configs with Chromium.
...
Change-Id: Idad14d12331f0f113cfa3c4e2aa006feec407a67
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360040
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42812}
2024-08-20 08:47:43 +00:00
Patrick Reynolds
f654998b80
Add opus_audio_decoder_factory to the static library.
...
Bug: None
Change-Id: Ia8aa6594ec1d2bca6649ec7da284c931a0f68ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360000
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42811}
2024-08-20 08:39:10 +00:00
Philipp Hancke
2cfedb277a
Remove vestiges of GTURN
...
and update some usage to use the "correct" stun attribute names
BUG=webrtc:42229250
Change-Id: If0c34d1d9b399766d7073661ea2a5515100256a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359440
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42810}
2024-08-19 16:05:37 +00:00
Philipp Hancke
37c406a1fd
Clean up 32 byte mid support from the demuxer
...
the SDP parser is restricting this to 16 bytes already
BUG=webrtc:42222663
Change-Id: I26f8f15a3f15cb5a5c8eab64f9dc3e26c0ea1bbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359780
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42809}
2024-08-19 15:23:03 +00:00
Harald Alvestrand
b1ffa9bd4e
Export SdpTypeFromString
...
Needed to get rid of a form of CreateSessionDescription that is due
for deprecation.
Bug: None
Change-Id: I9717b7ded1e28cf803de4bebc852c2f255851918
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359941
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42808}
2024-08-19 13:34:49 +00:00
