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}
As the rate allocation has been moved into entirely into
SimulcastRateAllocator, and the listeners are thus no longer needed,
this class doesn't fill any other purpose than to determine if
ScreenshareLayers or TemporalLayers should be created for a given
simulcast stream. This can however be done just from looking at the
VideoCodec instance, so changing this into a static factory method.
Due to dependencies from upstream projects, keep the class name and
field in VideoCodec around for now.
Bug: webrtc:9012
Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979
Reviewed-on: https://webrtc-review.googlesource.com/63264
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22529}
This work is in preparation for refactoring the TemporalLayers api.
Bug: webrtc:9012
Change-Id: I01908ee034fb79996e687ff72d10178acf102321
Reviewed-on: https://webrtc-review.googlesource.com/61781
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22445}
Overriding implementations of VideoEncoder::GetScalingSettings that
want to enable quality scaling must now provide the thresholds.
Bug: webrtc:8830
Change-Id: I75c47cb56ac1b9cf77401684980b3167e485f51c
Reviewed-on: https://webrtc-review.googlesource.com/46622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22172}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
This can be used to determine whether the bitrate of a given spatial and temporal layer has been set in the allocation, even if the value it's set to is zero.
GetBitrate still returns 0 if the queried layer does not have the bitrate set.
Bug: webrtc:8479
Change-Id: I1d982e211da9b052fcccdbf588b67da1a4550c60
Reviewed-on: https://webrtc-review.googlesource.com/17440
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20852}
All frames are checked against hard-coded dependency graph
using new helper class. It's invoked in RTC_DCHECK(). Only
DefaultTemporalLayers are fully implemented in this CL, checker
for ScreenshareLayers is not doing anything for now.
Bug: none
Change-Id: I3ec017176d8c25f7572c8f161e52f2ebfac8220f
Reviewed-on: https://webrtc-review.googlesource.com/3740
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20066}
If this is not done, the RTC_DCHECK_CALLED_SEQUENTIALLY might fire
if the encoder is used on a new VideoStreamEncoder. This happens
after VideoSendStream recreations due to changes in the SDP.
BUG=b/66590444
Change-Id: I086370526afbbe2ba629805f97f89e512ba3f472
Reviewed-on: https://webrtc-review.googlesource.com/4360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20020}
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}
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}
