If the paced queue gets too long becaused of e.g. encoder overshoot,
the encoder is paused by setting the target bitrate to 0. Don't signal
this 0-bitrate via RTCP TargetBitrate message as the overall target
bitrate is probably unchanged.
Bug: webrtc:9734
Change-Id: I77f23b707a8d4494d0c89fa05005ac1482eace52
Reviewed-on: https://webrtc-review.googlesource.com/99507
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24679}
Use in VideoQualityTest replaced by creating a wrapper for the encoder.
Bug: None
Change-Id: I5c5519e147ca7ddb97696b0d6958a8a1f5cc6e83
Reviewed-on: https://webrtc-review.googlesource.com/94152
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24533}
This is a reland of da0898dfae
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.
Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
This prevents a lot of unnecessary processing taking place when we are
not using FEC.
This CL also removes the FieldTrial that was used to disable ulpfec, as it's no longer used.
Bug: webrtc:9514
Change-Id: I8285b933f71eea971f5932cd19833455a42c8639
Reviewed-on: https://webrtc-review.googlesource.com/87848
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23952}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
Since the webrtc_common build target does not have visibility set, we
cannot easily use BitrateAllocation in other parts of Chromium.
This is currently blocking parts of chromium:794608, and I know of other
usage outside webrtc already, so moving it to api/ should be warranted.
Also, since there's some naming confusion and this class is video
specific rename it VideoBitrateAllocation. This also fits with the
standard interface for producing these: VideoBitrateAllocator.
Bug: chromium:794608
Change-Id: I4c0fae40f9365e860c605a76a4f67ecc9b9cf9fe
Reviewed-on: https://webrtc-review.googlesource.com/70783
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22986}
This makes it easier to refactor the interface in upcoming CLs.
Bug: None
Change-Id: I67d0216e24f087294e95ac96f7278f302bf69832
Reviewed-on: https://webrtc-review.googlesource.com/71041
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22933}
This is a reland of fb82fcc7f9
Original change's description:
> Move creating encoder to VideoStreamEncoder.
>
> This used to be in WebRtcVideoChannel::WebRtcVideoSendStream.
> One implication is that encoder is not created until the first
> frame arrives, and some of the tests needed updates to emit a
> frame or two.
>
> Bug: webrtc:8830
> Change-Id: I78169b2bb4dfa4197b4b4229af9fd69d0f747835
> Reviewed-on: https://webrtc-review.googlesource.com/64885
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22905}
TBR=magjed@webrtc.org,kwiberg@webrtc.org
Bug: webrtc:8830
Change-Id: I9565095ea1880fb49d15111198c08b2fcb84f18c
Reviewed-on: https://webrtc-review.googlesource.com/70740
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22930}
This prepares for adding unit tests for VideoSendStreamImpl.
Bug: None
Change-Id: I488041b09f4a455ce4cf1bdc7b8163ef6ad19a8a
Reviewed-on: https://webrtc-review.googlesource.com/70782
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22926}
