16 Commits

Author SHA1 Message Date
Harald Alvestrand
882b32d00f Reland "Use PayloadTypePicker for video PT assignment"
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.

Reason for revert: Revised codec matching to fix issue.

Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).

Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> 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@{#43490}

Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
2024-12-12 16:37:30 -08:00
Harald Alvestrand
e046787a5a Revert "Use PayloadTypePicker for video PT assignment"
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.

Reason for revert: Broke internal tests.

Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}

Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
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@{#43490}
2024-12-03 22:24:21 +00:00
Harald Alvestrand
e5048949b0 Use PayloadTypePicker for video PT assignment
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.

Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
2024-12-03 18:18:28 +00:00
Harald Alvestrand
b7abaee819 Revert "Use Payload Type suggester for all codec merging"
This reverts commit 0bac2aae596771db020f01a57fee4828081fbc38.

Reason for revert: Suspected breakages downstream

Original change's description:
> Use Payload Type suggester for all codec merging
>
> Bug: webrtc:360058654
> Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43267}

Bug: webrtc:360058654, b/375132036
Change-Id: Ieda626270193e7e6c93903b3c03a691b2bf0c1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43290}
2024-10-23 11:37:18 +00:00
Harald Alvestrand
0bac2aae59 Use Payload Type suggester for all codec merging
Bug: webrtc:360058654
Change-Id: Id475762253c427c1800c2352a60fc0121c2dc388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364783
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43267}
2024-10-18 16:58:42 +00:00
Harald Alvestrand
19bbd6f02f Move some codec-comparing functions to a single file.
This CL is a pure move; later CLs will try to increase consistency
between the functions.

Bug: webrtc:360058654
Change-Id: I6662b3d35f8e2dab60c2778a4755454fe3029fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43210}
2024-10-09 22:10:36 +00:00
Harald Alvestrand
c17ca01f54 Move the payload type picker to call/
Since media/ and pc/ both have to use this, and both
depend on call/, this seems to be the right place to put it.

Also factor out the interface that media will use in a separate
interface class.

Bug: webrtc:360058654
Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 12:36:50 +00:00
Harald Alvestrand
93c9aa1914 Apply include-cleaner to call/
with downstream fixes.

Bug: webrtc:42226242
Change-Id: I88d7b5ffc1f86c01ea13948c27b4210d032f4190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361360
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42921}
2024-09-03 07:51:03 +00:00
Per K
569849e885 Move call/simulated_network to test/network
Old target and call/simulated.h exist but refer to new target in test/network.

Bug: webrtc:14525
Change-Id: Ida04cef17913f2f829d7e925ae454dc40d5e8240
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349264
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42191}
2024-04-29 09:55:06 +00:00
Åsa Persson
59947d2871 SimulcastEncoderAdapter: Use FramerateController instead of FramerateControllerDeprecated.
Results from test (CallPerfTest.TestEncodeFramerateVp8Simulcast):
Simulcast streams:
0: max_fps:20 -> StreamStats.encode_frame_rate:15 (before), 20 (after)
1: max_fps:30

Bug: webrtc:13031
Change-Id: I30e6b2dcb2746859bd3e21b098bfa7b0fb3b2dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34867}
2021-08-30 10:20:55 +00:00
Olga Sharonova
09ceed2165 Async audio processing API
API to injecting a heavy audio processing operation into WebRTC audio capture pipeline

Bug: webrtc:12003
Change-Id: I9f6f58f468bd84efd0a9d53d703db6229a03959e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165788
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32291}
2020-10-02 12:33:34 +00:00
Niels Möller
a8327d4415 Move FrameCounts and FrameCountObserver to common_video/frame_counts.h
Bug: webrtc:7660
Change-Id: Ic42227508e0e26b5443de0ef0e0722126cbbb8fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182501
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32005}
2020-08-27 09:53:18 +00:00
Ying Wang
3b790f316c Make fec controller plug-able.
Bug: webrtc:8656
Change-Id: I3d42ffc92a7c95266e5d53bab03f388bd0de2592
Reviewed-on: https://webrtc-review.googlesource.com/39760
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21710}
2018-01-22 11:48:16 +00:00
Fredrik Solenberg
a8b7c7f4c6 Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 13:27:47 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00