99 Commits

Author SHA1 Message Date
Johnny Lee
1a1c52baf9 H.264 temporal layers w/frame marking (PART 2/3)
Bug: None
Change-Id: Id1381d895377d39c3969635e1a59591214aabb71
Reviewed-on: https://webrtc-review.googlesource.com/c/86140
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26624}
2019-02-09 16:47:09 +00:00
Erik Språng
a8d48ab87b Fix incorrect FPS measure when frame dropper kicks in
Bug: webrtc:10302
Change-Id: I4f8df7d41d8750e0810c2300fcd90b3eff7fb56d
Reviewed-on: https://webrtc-review.googlesource.com/c/121954
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26610}
2019-02-08 13:50:10 +00:00
Erik Språng
7ca375c8ca Implement encoder overshoot detector and rate adjuster.
The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.

Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.

Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26573}
2019-02-06 15:54:11 +00:00
Ilya Nikolaevskiy
b3032b6e33 Revert "Partial frame capture API part 4"
This reverts commit 62b9fb44aa9a05ef0e4866bcc0580779456c4cf7.

Reason for revert: Speculative revert for broken bots

Original change's description:
> Partial frame capture API part 4
> 
> Wire-up PartialFrameCompressor to VideoStreamEncoder.
> 
> Bug: webrtc:10152
> Change-Id: I6a3df28e392cf3f47aec1c97abb1d4d73d5f7e2a
> Reviewed-on: https://webrtc-review.googlesource.com/c/120409
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26548}

TBR=ilnik@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: Ib26fbf1b49f21f9f55b9b3e54fa6e6e33bf26dd2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10152
Reviewed-on: https://webrtc-review.googlesource.com/c/121564
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26555}
2019-02-05 15:33:55 +00:00
Ilya Nikolaevskiy
62b9fb44aa Partial frame capture API part 4
Wire-up PartialFrameCompressor to VideoStreamEncoder.

Bug: webrtc:10152
Change-Id: I6a3df28e392cf3f47aec1c97abb1d4d73d5f7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/120409
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26548}
2019-02-05 13:56:53 +00:00
Rasmus Brandt
c402dbe2b0 Account for simulcast hysteresis in padding rate calculation.
Bug: webrtc:10271
Change-Id: If0b0eb7d94fb1c892880ff4745f34c43fcdeee54
Reviewed-on: https://webrtc-review.googlesource.com/c/120661
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26527}
2019-02-04 10:49:04 +00:00
Mirta Dvornicic
ccc1b57e32 Poll is_hardware_accelerated from VideoEncoder instead of VideoEncoderFactory.
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.

Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
2019-01-15 14:12:12 +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
Niels Möller
6bb5ab9740 Reland "Refactor and remove media_optimization::MediaOptimization."
This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499.

Reason for revert: This change seemed innocent after all, so undoing speculative revert.

Original change's description:
> Revert "Refactor and remove media_optimization::MediaOptimization."
> 
> This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.
> 
> Reason for revert: Speculative revert due to downstream crashes.
> 
> Original change's description:
> > Refactor and remove media_optimization::MediaOptimization.
> > 
> > This CL removes MediaOptmization and folds some of its functionality
> > into VideoStreamEncoder.
> > 
> > The FPS tracking is now handled by a RateStatistics instance. Frame
> > dropping is still handled by FrameDropper. Both of these now live
> > directly in VideoStreamEncoder.
> > There is no intended change in behavior from this CL, but due to a new
> > way of measuring frame rate, some minor perf changes can be expected.
> > 
> > A small change in behavior is that OnBitrateUpdated is now called
> > directly rather than on the next frame. Since both encoding frame and
> > setting rate allocations happen on the encoder worker thread, there's
> > really no reason to cache bitrates and wait until the next frame.
> > An edge case though is that if a new bitrate is set before the first
> > frame, we must remember that bitrate and then apply it after the video
> > bitrate allocator has been first created.
> > 
> > In addition to existing unit tests, manual tests have been used to
> > confirm that frame dropping works as expected with misbehaving encoders.
> > 
> > Bug: webrtc:10164
> > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> > Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> > Commit-Queue: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26147}
> 
> TBR=nisse@webrtc.org,sprang@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: webrtc:10164
> Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
> Reviewed-on: https://webrtc-review.googlesource.com/c/116780
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26191}

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

Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10164
Reviewed-on: https://webrtc-review.googlesource.com/c/116960
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26213}
2019-01-11 10:46:33 +00:00
Niels Möller
6613f8e98a Revert "Refactor and remove media_optimization::MediaOptimization."
This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b.

Reason for revert: Speculative revert due to downstream crashes.

Original change's description:
> Refactor and remove media_optimization::MediaOptimization.
> 
> This CL removes MediaOptmization and folds some of its functionality
> into VideoStreamEncoder.
> 
> The FPS tracking is now handled by a RateStatistics instance. Frame
> dropping is still handled by FrameDropper. Both of these now live
> directly in VideoStreamEncoder.
> There is no intended change in behavior from this CL, but due to a new
> way of measuring frame rate, some minor perf changes can be expected.
> 
> A small change in behavior is that OnBitrateUpdated is now called
> directly rather than on the next frame. Since both encoding frame and
> setting rate allocations happen on the encoder worker thread, there's
> really no reason to cache bitrates and wait until the next frame.
> An edge case though is that if a new bitrate is set before the first
> frame, we must remember that bitrate and then apply it after the video
> bitrate allocator has been first created.
> 
> In addition to existing unit tests, manual tests have been used to
> confirm that frame dropping works as expected with misbehaving encoders.
> 
> Bug: webrtc:10164
> Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
> Reviewed-on: https://webrtc-review.googlesource.com/c/115620
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26147}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10164
Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c
Reviewed-on: https://webrtc-review.googlesource.com/c/116780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26191}
2019-01-10 11:39:24 +00:00
Erik Språng
07276e4f89 Refactor and remove media_optimization::MediaOptimization.
This CL removes MediaOptmization and folds some of its functionality
into VideoStreamEncoder.

The FPS tracking is now handled by a RateStatistics instance. Frame
dropping is still handled by FrameDropper. Both of these now live
directly in VideoStreamEncoder.
There is no intended change in behavior from this CL, but due to a new
way of measuring frame rate, some minor perf changes can be expected.

A small change in behavior is that OnBitrateUpdated is now called
directly rather than on the next frame. Since both encoding frame and
setting rate allocations happen on the encoder worker thread, there's
really no reason to cache bitrates and wait until the next frame.
An edge case though is that if a new bitrate is set before the first
frame, we must remember that bitrate and then apply it after the video
bitrate allocator has been first created.

In addition to existing unit tests, manual tests have been used to
confirm that frame dropping works as expected with misbehaving encoders.

Bug: webrtc:10164
Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744
Reviewed-on: https://webrtc-review.googlesource.com/c/115620
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26147}
2019-01-07 14:58:23 +00:00
Artem Titov
1ebfb6aac7 Introduce VideoFrame::id to keep track of frames inside application.
Also switch webrtc code from deprecated constructors to the builder API.

Change-Id: Ie325bf1e9b4ff1e413fef3431ced8ed9ff725107
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/114422
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26132}
2019-01-04 08:59:26 +00:00
Niels Möller
412d185b4a Delete pre_encode_callback from VideoSendStream::Config
Bug: webrtc:9864
Change-Id: I7f0c897345c99765ea9de77bc70b43ba0e4af19b
Reviewed-on: https://webrtc-review.googlesource.com/c/115320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26110}
2019-01-02 09:14:32 +00:00
Erik Språng
d96b275cd6 Refactor EncodeParameters usage, remove unused rtt/loss
Bug: webrtc:10126
Change-Id: Ib93f5e65b25540576c026197f72a5902cf43fc16
Reviewed-on: https://webrtc-review.googlesource.com/c/114281
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26001}
2018-12-13 12:15:09 +00:00
Sergey Silkin
8b9b5f98db Activate/deactivate VP9 spatial layers.
* Stop encoding spatial layers S(n >= N) if application deactivates
spatial layer N by setting RTCRtpEncodingParameters.active = false.

* Move calculation of padding bitrate to SvcRateAllocator class.

* Pad up to minimum required bitrate of base layer if ALR probing is
enabled.

Bug: webrtc:9350
Change-Id: I398284c943d43348def535c83263fc234c9767fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25945}
2018-12-10 12:55:51 +00:00
Mirta Dvornicic
1ec2a16121 Revert "Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo"
This reverts commit cdc5eb0de179dcc866ef770ea303879c64466879.

Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().

Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
> 
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
> 
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}

TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
2018-12-10 10:36:00 +00:00
Mirta Dvornicic
cdc5eb0de1 Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.

Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
2018-12-06 15:24:45 +00:00
Florent Castelli
450b548cad Encode frames at the largest layer resolution instead of last layer
The current code assumes that layers are ordered from the smallest
to the largest.
If that assumption is broken and the last layer is smaller than the
others, all layers that are bigger will be scaled up.

Bug: webrtc:10069
Change-Id: Iff87ddba741d5dfe3d0cc25a8f75d898a417eec7
Reviewed-on: https://webrtc-review.googlesource.com/c/112460
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25878}
2018-12-03 14:07:14 +00:00
Erik Språng
eee39206a2 Don't poll EncoderInfo from encoder twice per frame
Bug: webrtc:9890
Change-Id: Id4c2062a1c0c6be699f2096b4c0b334c98f3c4ba
Reviewed-on: https://webrtc-review.googlesource.com/c/111083
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25672}
2018-11-16 14:11:38 +00:00
Jiawei Ou
c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer"
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00
Qingsi Wang
59844ce57e Revert "Use the factory instead of using the builtin code path in VideoCodecInitializer."
This reverts commit be142178aaf6ab4089b4d81c88c3d59c12cca567.

Reason for revert: breaking internal projects

Original change's description:
> Use the factory instead of using the builtin code path in `VideoCodecInitializer`.
> 
> Bug: webrtc:9513
> Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
> Reviewed-on: https://webrtc-review.googlesource.com/c/94782
> Commit-Queue: Jiawei Ou <ouj@fb.com>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25456}

TBR=brandtr@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,tkchin@webrtc.org,shampson@webrtc.org,glaznev@webrtc.org,ouj@fb.com,qingsi@webrtc.org

Change-Id: I8040ccabe3ae6464d72c7696adb663c1dd275b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9513
Reviewed-on: https://webrtc-review.googlesource.com/c/108980
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25459}
2018-11-01 04:46:02 +00:00
Jiawei Ou
be142178aa Use the factory instead of using the builtin code path in VideoCodecInitializer.
Bug: webrtc:9513
Change-Id: Ia299ae1044a3ff4c91e208200938cba540bdcea6
Reviewed-on: https://webrtc-review.googlesource.com/c/94782
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25456}
2018-10-31 22:47:02 +00:00
Erik Språng
e2fd86a79c Move encoder metadata into EncoderInfo struct.
This deprecates the following methods in VideoEncoder:
  virtual ScalingSettings GetScalingSettings() const;
  virtual bool SupportsNativeHandle() const;
  virtual const char* ImplementationName() const;

Though they are not marked RTC_DEPRECATED since we still want to call
them from within the default GetEncoderInfo() until downstream
projects have been updated.

Furthmore, implementation name is changed from const char* to
std:string, which prevents some lifetime issues with dynamic encoder
names, and CodecSpecificInfo.codec_name is removed in favor of getting
the implementation name via GetEncoderInfo().

This CL removes calls to these deprecated methods, follow-ups will also
remove implementations of the methods and replace them with new
GetEncoderInfo() substitutions.

Bug: webrtc:9890
Change-Id: I6fd6e531480c0b952f53dbd5105e0b0adc3e3b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/106905
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25351}
2018-10-25 08:51:53 +00:00
Oleh Prypin
a1d9ca47f9 Revert "Add ability to specify if rate controller of video encoder is trusted."
This reverts commit 3e335d1423cab06cca8cdb4f1fadb0b16c9e7d38.

Reason for revert: breaks downstream project

Original change's description:
> Add ability to specify if rate controller of video encoder is trusted.
>
> If rate controller is trusted, we disable the frame dropper in the
> media optimization module.
>
> Bug: webrtc:9722
> Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
> Reviewed-on: https://webrtc-review.googlesource.com/c/105020
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25107}

TBR=brandtr@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org

Change-Id: Ifdb0aae684894854a184ec1e7423a7c62e7ba237
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9722
Reviewed-on: https://webrtc-review.googlesource.com/c/105360
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25117}
2018-10-11 15:37:40 +00:00
Erik Språng
3e335d1423 Add ability to specify if rate controller of video encoder is trusted.
If rate controller is trusted, we disable the frame dropper in the
media optimization module.

Bug: webrtc:9722
Change-Id: I821f21fd74a400ee9d5aa3f6b42d4e569033acbe
Reviewed-on: https://webrtc-review.googlesource.com/c/105020
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25107}
2018-10-11 09:07:34 +00:00
Niels Möller
4dc66c53d0 Move EncodedImage class to api/video/
Bug: webrtc:9378
Change-Id: I8fb3b19cad0ad428abc6c8e6b507180d461882ba
Reviewed-on: https://webrtc-review.googlesource.com/c/104002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25033}
2018-10-08 07:37:10 +00:00
Jonas Olsson
84df1c724e Make fewer copies when using StringBuilder.
Replace calls to .str() which copies with .Release which moves in cases where that's safe.

This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"

Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00
Åsa Persson
8c1bf9595a Reland "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad.

Reason for revert: downstream project fixed.

Original change's description:
> Revert "Add initial support for RtpEncodingParameters max_framerate."
>
> This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Add initial support for RtpEncodingParameters max_framerate.
> >
> > Add support to set the framerate to the maximum of |max_framerate|.
> > Different framerates are currently not supported per stream for video.
> >
> > Bug: webrtc:9597
> > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> > Reviewed-on: https://webrtc-review.googlesource.com/92392
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24270}
>
> TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9597
> Reviewed-on: https://webrtc-review.googlesource.com/94060
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24277}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Bug: webrtc:9597
Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e
Reviewed-on: https://webrtc-review.googlesource.com/100080
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:06:33 +00:00
Jonas Olsson
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
Niels Möller
2377588c82 Add accessor methods for RTP timestamp of EncodedImage.
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}
2018-08-21 09:15:51 +00:00
Kári Tristan Helgason
639602ace3 Activate initial framedrop when first BW estimate arrives
Bug: webrtc:9176, webrtc:6086
Change-Id: I420f8a0c6191697f7b50aaf780cf90a4ea365739
Reviewed-on: https://webrtc-review.googlesource.com/79580
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24306}
2018-08-16 08:41:39 +00:00
Mirko Bonadei
948b7e3755 Revert "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.

Reason for revert: Breaks downstream project.

Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
> 
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
> 
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}

TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
2018-08-14 07:25:23 +00:00
Åsa Persson
ced5cfdb35 Add initial support for RtpEncodingParameters max_framerate.
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.

Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
2018-08-13 09:59:04 +00:00
Niels Möller
213618e37e New api function CreateVideoStreamEncoder.
Bug: webrtc:8830
Change-Id: I01de86f601e48f76e6b41b4182ce006d397a190c
Reviewed-on: https://webrtc-review.googlesource.com/78260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24079}
2018-07-24 09:14:26 +00:00
Jiawei Ou
4206a0a849 Exposing video bitrate allocator into API
In order to have public video bitrate allocator factory, the video bitrate allocator has be part of
the api.

Bug: webrtc:9513
Change-Id: Ia2e5ab9eb988c710c1ac492afccc470a92544aa2
Reviewed-on: https://webrtc-review.googlesource.com/88083
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24073}
2018-07-23 21:23:21 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Yves Gerey
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
Danil Chapovalov
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
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}
2018-06-15 12:09:49 +00:00
Ilya Nikolaevskiy
fc9dcb6a00 Remove wire-up for cancelled experement on VAAPI VP8 encoding
This experiment is now wired up inside of chrome using field trial and
this passthrough is now obsolete.

Bug: chromium:794608
Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08
Reviewed-on: https://webrtc-review.googlesource.com/82780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23625}
2018-06-15 10:04:07 +00:00
Niels Möller
97e04884bd Delete unused stats for preferred_bitrate.
Bug: webrtc:8830
Change-Id: Iaa30488255f2e09e269274136d370740cd030902
Reviewed-on: https://webrtc-review.googlesource.com/78880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23529}
2018-06-07 08:11:07 +00:00
Ilya Nikolaevskiy
b6c462d4e4 Cleanup webrtc:: namespace from leaked TimingFrameFlags
Bug: webrtc:9351
Change-Id: Ifbc0a522bf13ab62a2e490b9f129eacfabe7796f
Reviewed-on: https://webrtc-review.googlesource.com/80961
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23520}
2018-06-05 13:52:04 +00:00
“Michael
277a656263 Unstable BWE due to improper bit rate padding for VP9.
Bug: webrtc:9345
Change-Id: I5b1e0b4ed7a8c1d0b942b09433017cac6d53c64b
Reviewed-on: https://webrtc-review.googlesource.com/79000
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23496}
2018-06-01 20:07:06 +00:00
Ilya Nikolaevskiy
97b4ee5b4c Wire up VAAPI VP8 experimental support in WebRTC.
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.

Artificial Sdp parameter is added to the sdp format if the flag is set.

Additionally, sdp format is propagated in vp8 simulcast adapters.

Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
2018-05-28 12:30:19 +00:00
Niels Möller
0327c2ddc1 Move VideoStreamEncoderInterface to api/.
Bug: webrtc:8830
Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0
Reviewed-on: https://webrtc-review.googlesource.com/74481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23334}
2018-05-21 19:50:37 +00:00
Taylor Brandstetter
49fcc10de6 Merge DegradationPreference enums.
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".

It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.

Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
2018-05-17 11:21:52 +00:00
Niels Möller
f133856dfa Delete nack_enabled flag in encoder configuration.
This is a followup to cl https://webrtc-review.googlesource.com/71380,
which reworked the way encoder resilience is done, and made the
nack_enabled flag unused.

Bug: webrtc:8830
Change-Id: I3de2508c97bc71e01c8f2232d16cd1f33e57fe4a
Reviewed-on: https://webrtc-review.googlesource.com/69986
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23080}
2018-05-02 16:05:27 +00:00
Niels Möller
2d06118623 Reland "Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler."
This is a reland of f394f65b7195a5f9f71c9f0dec3bb68300590b5b

Original change's description:
> Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler.
> 
> In addition restore call to ConfigureQualityScaler in SetSource, which
> is needed if degradation preferences change mid-stream.
> 
> Fixes a regressions from https://webrtc-review.googlesource.com/70740,
> The encoder's GetScalingSettings may depend on arguments to
> InitEncode, so configuring the quality scaler only at encoder creation
> time isn't enough.
> 
> Bug: webrtc:8830
> Change-Id: I48f66cde219c56272f44441fdb26ec64c6002068
> Reviewed-on: https://webrtc-review.googlesource.com/72000
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22993}

Bug: webrtc:8830
Change-Id: I96f0b8c27e3b58b9e9268117267abcb63ee7fe73
Reviewed-on: https://webrtc-review.googlesource.com/72320
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23071}
2018-05-02 07:58:56 +00:00
Åsa Persson
a945aee72e Make quality scaler downscale faster.
Include dropped frames by the encoder in the frame drop percentage.

To react faster at low framerates:
- Use ExpFilter instead of MovingAverage to filter QP values.
- Reduce sampling interval while waiting for minimum number of needed frames (when not in fast rampup mode).

A separate slower ExpFilter is used for upscaling.

Bug: webrtc:9169
Change-Id: If7ff6c3bd4201fda2da67125889838fe96ce7061
Reviewed-on: https://webrtc-review.googlesource.com/70761
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23014}
2018-04-25 09:08:21 +00:00
Mirko Bonadei
2ff7b6adf0 Revert "Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler."
This reverts commit f394f65b7195a5f9f71c9f0dec3bb68300590b5b.

Reason for revert: Breaks downstream project.

Original change's description:
> Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler.
> 
> In addition restore call to ConfigureQualityScaler in SetSource, which
> is needed if degradation preferences change mid-stream.
> 
> Fixes a regressions from https://webrtc-review.googlesource.com/70740,
> The encoder's GetScalingSettings may depend on arguments to
> InitEncode, so configuring the quality scaler only at encoder creation
> time isn't enough.
> 
> Bug: webrtc:8830
> Change-Id: I48f66cde219c56272f44441fdb26ec64c6002068
> Reviewed-on: https://webrtc-review.googlesource.com/72000
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22993}

TBR=nisse@webrtc.org,asapersson@webrtc.org

Change-Id: I8c1c12fb53a1bfd6d03c54b93dac6033b7cce081
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/72220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23011}
2018-04-24 20:26:22 +00:00
Niels Möller
f394f65b71 Make VideoStreamEncoder::ReconfigureEncoder always call ConfigureQualityScaler.
In addition restore call to ConfigureQualityScaler in SetSource, which
is needed if degradation preferences change mid-stream.

Fixes a regressions from https://webrtc-review.googlesource.com/70740,
The encoder's GetScalingSettings may depend on arguments to
InitEncode, so configuring the quality scaler only at encoder creation
time isn't enough.

Bug: webrtc:8830
Change-Id: I48f66cde219c56272f44441fdb26ec64c6002068
Reviewed-on: https://webrtc-review.googlesource.com/72000
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22993}
2018-04-24 09:06:44 +00:00