admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
40,229 Commits 1 Branch 0 Tags
Commit Graph

2702 Commits

Author SHA1 Message Date
Tommi
8f7a17f80f Remove unsupported configuration value, allow_codec_switching 
Bug: webrtc:11341
Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40995}
 2023-10-24 05:07:25 +00:00
Tommi
7c1ddb760c Support initializing a SequenceChecker with a provided TaskQueue. 
Bug: none
Change-Id: I5106f29ab7f9ed8530626f33f6259eb7aeb9e779
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324260
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40993}
 2023-10-23 14:43:04 +00:00
Tommi
5b186e98bc Remove effectively dead code for allow_codec_switching 
Bug: webrtc:11341
Change-Id: I88e3c1059f5ebcc9d693c0719534aaacd4b9199b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324283
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40990}
 2023-10-23 14:08:11 +00:00
Henrik Boström
8005d5613e Add stats-related TODOs with crbugs. 
Someone wondered why framesRendered has not been implemented. I had a
look, and discovered that a) we need to implement it, and b) our entire
inter-frame, pause, and freeze metrics are measured at the wrong time
because what WebRTC considered "OnRenderedFrame()" is not actually when
the frame was rendered.

So that we don't forget this again, I filed two crbugs and added TODOs
in the code for future reference to anyone interested in these metrics.

Bug: webrtc:15600, webrtc:15601
Change-Id: Id38df7874df715e9b9c0410efa4a9bc2af5d6232
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324306
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40987}
 2023-10-23 09:53:11 +00:00
Harald Alvestrand
aa653c0d76 Reland "Deprecate all classes related to AsyncResolver" 
This reverts commit 08d431ec34a1c7ab52557702f2cebd9fdfacae9e.

Reason for revert: Last (hopefully) Chrome blocker removed

Original change's description:
> Revert "Deprecate all classes related to AsyncResolver"
>
> This reverts commit 61a442809cc06de93a613186084d2dfa9c934d81.
>
> Reason for revert: Breaks roll into Chromium
>
> Original change's description:
> > Deprecate all classes related to AsyncResolver
> >
> > and remove internal usage.
> >
> > Bug: webrtc:12598
> > Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40919}
>
> Bug: webrtc:12598
> Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40927}

Bug: webrtc:12598
Change-Id: I3c7b07c831eb9ff808368433d9b9ae8ec4b2afb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323720
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40944}
 2023-10-17 07:08:57 +00:00
Jeremy Leconte
49dcb200ab Export FieldTrialBasedConfig in order to use it in Chromium. 
Goal is to initialize peerconnections in Chromium using this based
field trial config until a proper config that doesn't rely on the
global field trial string can be used (https://crrev.com/c/4936314).

Change-Id: I3d006e2445ccc4880b73b564c8ad4408242d3696
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323621
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40941}
 2023-10-16 16:13:48 +00:00
Danil Chapovalov
a3ce407023 Cleanup Call construction 
Return unique_ptr to clearly communicate ownership is transfered.
Remove Call::Config alias

Bug: None
Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40934}
 2023-10-16 06:34:26 +00:00
Harald Alvestrand
08d431ec34 Revert "Deprecate all classes related to AsyncResolver" 
This reverts commit 61a442809cc06de93a613186084d2dfa9c934d81.

Reason for revert: Breaks roll into Chromium

Original change's description:
> Deprecate all classes related to AsyncResolver
>
> and remove internal usage.
>
> Bug: webrtc:12598
> Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Auto-Submit: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40919}

Bug: webrtc:12598
Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40927}
 2023-10-13 11:20:51 +00:00
Danil Chapovalov
c9880805ed Delete deprecated Timestamp accessor and setter in EncodedImage 
Bug: webrtc:9378
Change-Id: I5c67cca733f2fd646e73694524abf6b33438e8a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321860
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40926}
 2023-10-13 10:22:28 +00:00
Harald Alvestrand
61a442809c Deprecate all classes related to AsyncResolver 
and remove internal usage.

Bug: webrtc:12598
Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40919}
 2023-10-12 14:07:37 +00:00
Shuhai Peng
adb855cf8d Added AsString() function for color space for easier debugging 
Change-Id: I517a435769795de26502aea0dd3e01c1ed867616
Bug: chromium:1449570
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320166
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40898}
 2023-10-10 09:08:28 +00:00
Markus Handell
a84368c3d3 TaskQueueBase: use C++17. 
Fixed: webrtc:15371
Bug: webrtc:15371
Change-Id: I8bdcd733a7f43eb5e91986dfa478ad41e5074afd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320020
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40869}
 2023-10-05 10:04:47 +00:00
Sergey Sukhanov
4b84f01fe2 Change the type of PacedPacketInfo::send_bitrate_bps from int to strongly-typed DataRate. 
Bug: webrtc:15532
Change-Id: I84a6b9860d582d68beccdcfde4a12923b2cdbe8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322181
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40865}
 2023-10-04 14:20:22 +00:00
Jeremy Leconte
532a11ee72 [PCLF] Add an API to configure field trials. 
Change-Id: Ic37a4b2f86976b2f2eace225b6eefacd1c3bd046
Bug: b/303192724
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322321
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40864}
 2023-10-04 12:08:37 +00:00
Saúl Ibarra Corretgé
4408575d18 Reland "Enable SRTP GCM ciphers by default" 
This is a reland of commit d8633868b34dc1d841f0a9fd1afe2bc22aa8bde6

Original change's description:
> Enable SRTP GCM ciphers by default
>
> Bug: webrtc:15178
> Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40828}

Bug: webrtc:15178
Change-Id: I5ea939ed6263547ebc177d9dd1763ba888936866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321961
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40862}
 2023-10-03 22:39:48 +00:00
Manashi Sarkar
c2bbe4b952 Revert "Enable SRTP GCM ciphers by default" 
This reverts commit d8633868b34dc1d841f0a9fd1afe2bc22aa8bde6.

Reason for revert: Breaks downstream project.

Original change's description:
> Enable SRTP GCM ciphers by default
>
> Bug: webrtc:15178
> Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40828}

Bug: webrtc:15178
Change-Id: I88433e899cb4b705eafa3fceff3edc520629f603
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321863
Owners-Override: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Manashi Sarkar <manashi@google.com>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40832}
 2023-09-28 14:42:18 +00:00
Saúl Ibarra Corretgé
d8633868b3 Enable SRTP GCM ciphers by default 
Bug: webrtc:15178
Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40828}
 2023-09-28 10:18:56 +00:00
Danil Chapovalov
2d508f10d3 Deprecate old names for EncodedImage::RtpTimestamp accessor and setter 
Replace remaining webrtc usage of the deprecated names.

Bug: webrtc:9378
Change-Id: Ie5bd2d3eaf68316e7c827fc35c7c7d8e2eadeb9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321584
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40824}
 2023-09-28 07:29:22 +00:00
Palak Agarwal
0505115b5c Pass the correct abs_capture_timestamp while cloning audio frame 
This change replaces type of absolute_capture_timestamp_ms_ in
TransformableOutgoingAudioFrame from int to optional uint and makes
the function AbsoluteCaptureTimestamp() inside
TransformableAudioFrameInterface pure virtual.

Bug: webrtc:14949
Change-Id: Id3bdbcba63a5f91105ab198208e4f2b11eb3c7db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319000
Commit-Queue: Palak Agarwal <agpalak@google.com>
Reviewed-by: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40814}
 2023-09-26 15:57:52 +00:00
Tommi
6bf2d31e71 Change PortInterface::Type to string_view and make type_ member const 
Bug: none
Change-Id: Id1b0298eede5d2ae5010cc450d7bcb9eadd7b874
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318080
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40801}
 2023-09-25 13:48:50 +00:00
Björn Terelius
b4d4bbcebd Revert "Clean up last_packet_received_time_ as it's no longer used." 
This reverts commit 2f4bc6416651be40ef8f95a4695e6b7c41f18666.

Reason for revert: Breaks downstream test

Original change's description:
> Clean up last_packet_received_time_ as it's no longer used.
>
> Bug: webrtc:15377
> Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40792}

Bug: webrtc:15377
Change-Id: Ifa57671cc479cdd86f543c4edc236221beb76f90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321340
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40797}
 2023-09-25 08:49:53 +00:00
Danil Chapovalov
9c58483b5a Rename EncodedImage property Timetamp to RtpTimestamp 
To avoid name collision with Timestamp type,
To avoid confusion with capture time represented as Timestamp

Bug: webrtc:9378
Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40796}
 2023-09-24 20:06:48 +00:00
Ying Wang
2f4bc64166 Clean up last_packet_received_time_ as it's no longer used. 
Bug: webrtc:15377
Change-Id: I5453b9fd572a04dbea3241a2eb1c8ad8bb8b1186
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320560
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40792}
 2023-09-23 00:03:11 +00:00
qwu16
ae82df718c Add codec name H265 to support H265 in WebRTC 
Bug: webrtc:13485
Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40773}
 2023-09-20 09:25:32 +00:00
Harald Alvestrand
e14d122a7b Remove deprecated SendRtp and SendRtcp functions 
and delete remaining usages.

Bug: webrtc:15410
Change-Id: I912bedca80a5a446a3f770211d164a5eb0af02bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320520
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40766}
 2023-09-19 10:59:26 +00:00
Danil Chapovalov
090699a01b Delete deprecated RtpSource timestamp_ms constructor and accessor 
Bug: webrtc:13756
Change-Id: Ic43cf82451785b4fbe184fce466e77b16b2c781a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319581
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40765}
 2023-09-19 09:00:03 +00:00
philipel
31718d7ce2 Reland "Add option to disable quality scaling for AV1." 
This reverts commit 83102d39077f82f2d4539c160c659dcf789a5fdb.

Reason for revert: reland with fix

Original change's description:
> Revert "Add option to disable quality scaling for AV1."
>
> This reverts commit 446dbc66fde7e9d5e684d3f71e357c2076a91740.
>
> Reason for revert: downstream break
>
> Original change's description:
> > Add option to disable quality scaling for AV1.
> >
> > The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
> >
> > Bug: b/295129711
> > Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#40709}
>
> Bug: b/295129711
> Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Owners-Override: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40742}

Bug: b/295129711
Change-Id: Iab4846c2cd6074f50a3ebe9551432d449243b5d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320120
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40743}
 2023-09-13 15:19:36 +00:00
Philip Eliasson
83102d3907 Revert "Add option to disable quality scaling for AV1." 
This reverts commit 446dbc66fde7e9d5e684d3f71e357c2076a91740.

Reason for revert: downstream break

Original change's description:
> Add option to disable quality scaling for AV1.
>
> The main goal of this change is to disable the quality scaler when multiple spatial layers are used.
>
> Bug: b/295129711
> Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40709}

Bug: b/295129711
Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Owners-Override: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40742}
 2023-09-13 12:21:31 +00:00
philipel
46bbf7ec48 Add ScalabilityModeStringToEnum helper function. 
Bug: none
Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40732}
 2023-09-11 17:04:27 +00:00
Danil Chapovalov
564e8d395d Mark RtpSource timestamp_ms constructor and accessor deprecated 
Bug: webrtc:13757
Change-Id: Ica680dfc0b7420e0d168b4854a07bd73e367218e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319281
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40718}
 2023-09-08 07:38:27 +00:00
Danil Chapovalov
541756ff6b Discourage structs in api 
Structs make api harder to evolve:
deprecated unused properties,
change how data is represented.

Classes with accessors allow more graduated and safer api evolution.

Bug: None
Change-Id: I8ebd5e072d51cf7f5800666cfdac523d0f9a937f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40714}
 2023-09-07 10:41:49 +00:00
philipel
446dbc66fd Add option to disable quality scaling for AV1. 
The main goal of this change is to disable the quality scaler when multiple spatial layers are used.

Bug: b/295129711
Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40709}
 2023-09-06 12:37:22 +00:00
Harald Alvestrand
96e1882860 Convert AsyncDnsResolver to use absl::AnyInvocable 
Bug: webrtc:12598
Change-Id: I0950231d6de7cf53116a573dcd97a3cf5514946c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318400
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40670}
 2023-08-31 08:50:40 +00:00
Philipp Hancke
df3683e9a7 Remove public GenerateKeyFrame(list-of-rids) API from RtpSender 
since the spec and implementation took a different route

BUG=chromium:1354101

Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40665}
 2023-08-30 14:54:17 +00:00
Tony Herre
55b593fb6b Remove EncodedFrame::MissingFrame and start removing Decode() param 
Remove EncodedFrame::MissingFrame, as it was always false in actual
in-use code anyway, and remove usages of the Decode missing_frames param
within WebRTC. Uses/overrides in other projects will be cleaned up
shortly, allowing that variant to be removed from the interface.

Bug: webrtc:15444
Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#40662}
 2023-08-30 10:38:35 +00:00
Qiu Jianlin
44943c8064 Add H265 codec name and profile/tier/level utils. 
This adds H265 codec name and profile/tier/level handling needed for
H265 SDP negotiation.

Bug: webrtc:13485
Change-Id: I838b910042ce36f8ae3979c41a73ee46935c57d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#40661}
 2023-08-30 08:49:09 +00:00
Philipp Hancke
82c56ca794 Request keyframe via setParameters 
after the W3C changes in approach documented here:
  https://github.com/w3c/webrtc-extensions/pull/167

chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4643591

Note that this does not follow the route taken by the W3C API but
still considers this flag a part of the encodingParameters.

BUG=chromium:1354101

Change-Id: If0f0ec09bebddea1f01dd8afbe4747c21afe6793
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286741
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40656}
 2023-08-29 13:53:33 +00:00
Danil Chapovalov
f53597140f In RtpSource represent time with Timestamp type instead of int64_t 
Bug: webrtc:13757
Change-Id: I5d7da9c9aee489e4b57d361de174c59713cb2b14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40650}
 2023-08-29 10:05:03 +00:00
Tony Herre
5f14f9e6ed Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance 
Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by
- moving getters for EncodedImage fields up to EncodedImage
- copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame
- Removing EncodedFrame's inheritance of VCMEncodedFrame

We leave VCMEncodedFrame as part of the (near) deprecated
VideoCodingModule code. The only place which needs to accept either is
in the generic decoder.

Bug: webrtc:9378, b:296992877
Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40639}
 2023-08-28 11:46:48 +00:00
Harald Alvestrand
4d25a77fd3 Deprecate AsyncResolver config fields and remove internal usage. 
Bug: webrtc:12598
Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40627}
 2023-08-25 14:02:27 +00:00
Tommi
66bf3f472c Make PendingTaskSafetyFlag compatible with component builds 
Bug: chromium:1470992
Change-Id: I06cec9cda36c9de75b970eaf709f9ed3b9f466b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40618}
 2023-08-24 14:24:49 +00:00
Florent Castelli
43a5dd86c2 Implement codec selection api 
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.

Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
 2023-08-24 13:18:04 +00:00
Danil Chapovalov
7084e1b6d9 In VideoPlayoutDelay delete access to integer representation of min/max values 
Bug: webrtc:13756
Change-Id: I1a81c25e5e3fab68a44e94a5ab93e8184c824683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316864
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40612}
 2023-08-23 16:14:26 +00:00
Arthur Sonzogni
47faf32287 Add rtc_common_public_deps 
When built for chromium, some webrtc implementations are overridden and
are implemented by chrome's "//base". For instance webrtc::Location is
implemented by base::Location. So far so good, the affected targets are
correctly defined in GN to depend on base.

The problem: Most targets in webrtc do not declare correctly their
public_deps. When a public header of a target includes one from its
dependency, the dependency must be a public_deps. The public_deps
instruct GN to forward the capability to use code from the dependency
toward the dependent.

Unfortunately, it is not possible to fix the `public_deps` in webrtc,
because its is disallowed via a presubmit. See:
https://webrtc-review.googlesource.com/c/src/+/30262

WebRTC developers decided not to use `public_deps`, because GN config
are "translated" toward different kind of downstream build system who do
not really support the `public` dependencies concept. Instead WebRTC is
using some "common" configuration applied to all of its targets.

This patch add `rtc_common_public_deps` argument, to let embedders
add the dependencies WebRTC depends on.

Bug: chromium:1467773
Change-Id: I7de43372414a09886fcb07905451e6339c8ecc64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316660
Commit-Queue: Arthur Sonzogni <arthursonzogni@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40595}
 2023-08-22 11:32:06 +00:00
Danil Chapovalov
06717773a5 Move EncodedImage::playout_delay_ to private section of the class 
Remove code where integer -1 as delay is used to represent unset value.

Bug: webrtc:13756
Change-Id: I16a01e12c25a09ce21a971c9edabf47af5936662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316923
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40592}
 2023-08-22 08:24:37 +00:00
Tony Herre
960882047d Add mock of GetCaptureTimeIdentifier to MockTransformableVideoFrame 
Bug: webrtc:14878
Change-Id: I2dffad0932aee4d2ba37c8d57a3c28330e3cc294
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40585}
 2023-08-21 14:42:25 +00:00
Danil Chapovalov
b0b03a87b7 Mark api video timing classes with RTC_EXPORT 
Bug: None
Change-Id: Icf99dcdef7278b6051f040c51583a5e164e8f22e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316921
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40584}
 2023-08-21 12:55:08 +00:00
Artem Titov
1997837d16 Add stream label to test video source for better debugablity and testability 
Bug: b/294812400
Change-Id: I830515b797100ca2dc0e68dd3b79d5a1bb4068da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316221
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40581}
 2023-08-21 08:25:38 +00:00
Danil Chapovalov
c146b5f77b Represent unset VideoPlayoutDelay with nullopt rather than special value 
Remove support for setting one limit without another limit
because related rtp header extension doesn't support such values.

Start morphing VideoPlayouDelay into a class and stricter type: add accessors returning TimeDelta

Bug: webrtc:13756
Change-Id: If0dd02620528dc870b015beeff3a8103e04022ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40570}
 2023-08-18 13:17:50 +00:00
Tony Herre
392e4714e7 Remove deprecated TransformableAudioFrameInterface::getHeader() method 
Fixed: chromium:1456628
Change-Id: I12ea08070578de846f042c64f2808b55de1603a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315960
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40555}
 2023-08-15 16:31:02 +00:00