Commit graph

19 commits

Author SHA1 Message Date
Philipp Hancke
c14a2cb9cc Add nonstandard x-google-per-layer-pli fmtp for enabling per-layer keyFrames in response to PLIs
which needs to be added to the remote codecs a=fmtp:

This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.

This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.

BUG=webrtc:10107

Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
2024-05-03 10:59:22 -04:00
Jonas Oreland
6c2dae21e9 Move VideoEncoderConfig from api/ into video/config
This cl move VideoEncoderConfig from api/ to video/config.

VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.

brandt@ think that the reason these were in api/ in the
first place had to downstream project.

Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).

Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 09:44:43 +00:00
Tommi
28e9653b1f Remove dependency on RtpVideoSenderInterface from EncoderRtcpFeedback.
This removes the two step initialization and explicit circular
dependency between the sender and the observer that complicates
construction and making members const that should be.
Moving forward the encoder feedback instance will move to a different
class, so this CL is one part of making that change possible.

Also removing an unnecessary mutex and replacing with a checker.

Bug: webrtc:12840
Change-Id: I21694806b122592de0cd1e1d96f241d339a0860f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221108
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34214}
2021-06-03 10:46:49 +00:00
Markus Handell
a376518817 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: If5b2eae65c5f297f364b6e3c67f94946a09b4a96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178862
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31672}
2020-07-08 12:21:08 +00:00
Markus Handell
a827a30bb7 Revert "Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex."
This reverts commit 0eba415fb4.

Reason for revert: previously unknown lock recursion occurring downstream.

Original change's description:
> Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
> 
> Also migrates test/ partly.
> 
> Bug: webrtc:11567
> Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31653}

TBR=sprang@webrtc.org,handellm@webrtc.org

Change-Id: I13f337e0de5b8f0eb19deb57cb5623444460ec4d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178842
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31656}
2020-07-07 20:46:48 +00:00
Markus Handell
0eba415fb4 Migrate video/ except video/end_to_end_tests and video/adaptation to webrtc::Mutex.
Also migrates test/ partly.

Bug: webrtc:11567
Change-Id: I4203919615c087e5faca3b2fa1d54cba9f171e07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178813
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31653}
2020-07-07 18:01:44 +00:00
Bjorn A Mellem
7a9a092708 Delete media transport integration.
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
2019-11-26 19:19:36 +00:00
Niels Möller
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
2019-09-13 10:49:56 +00:00
Elad Alon
b6ef99bb33 Translate loss notifications and pass to encoder
Translate LossNotification RTCP messages (sequence number to
timestamp and additional information), then send the translted
message onwards to the encoder.

Bug: webrtc:10501
Change-Id: If2fd943f75c36cf813a83120318d8eefc8c595d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131950
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27545}
2019-04-10 15:31:53 +00:00
Elad Alon
0a8562e276 Forward LossNotification from RTCPReceiver to EncoderRtcpFeedback
TBR=sprang@webrtc.org

Bug: webrtc:10501
Change-Id: I09a571a65ba8515b027ee32d1f46e5cc7f699704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131325
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27513}
2019-04-09 11:13:39 +00:00
Elad Alon
14d1c9d968 Rename EncoderKeyFrameCallback back to EncoderRtcpFeedback
The name EncoderRtcpFeedback is more appropriate because:
1. "Callback" implies a single function exposed, which is
   not the case.
2. We intend to add OnLossNotification() to the same class.

Bug: webrtc:10501
Change-Id: I75b2e06eb070e85e872eba9fbc354aa4b68c7993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131289
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27490}
2019-04-08 14:33:19 +00:00
Niels Möller
fa89d84698 Register callback for key frame request from media transport.
Bug: webrtc:9719
Change-Id: Ibeadadb8e477d6d712fd69427c95e1e4f1940854
Reviewed-on: https://webrtc-review.googlesource.com/c/120340
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26473}
2019-01-30 16:26:31 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Magnus Flodman
3c62afd918 Don't throttle key-frame requests per layer.
Bug: webrtc:9688
Change-Id: Ia6f8b131412a8f46ad6fa3f0173c3285728bbeeb
Reviewed-on: https://webrtc-review.googlesource.com/100522
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24791}
2018-09-24 10:40:48 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
Sebastian Jansson
652dc915bc Adds unit tests for VideoSendStreamImpl.
Bug: None
Change-Id: Ifadad47af4769d8aca42c98832cea49a6c7977cd
Reviewed-on: https://webrtc-review.googlesource.com/71040
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22959}
2018-04-20 13:12:13 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +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
Renamed from webrtc/video/encoder_rtcp_feedback.h (Browse further)