314 Commits

Author SHA1 Message Date
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
Johannes Kron
fbf168386b Add HdrMetadata to VideoFrame
Bug: webrtc:8651
Change-Id: I28aacbc63e346328633fb862662343f47e966bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/108320
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25513}
2018-11-06 09:29:54 +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
Johannes Kron
23524ced41 Add HDR metadata struct
Bug: webrtc:8651
Change-Id: Ie7e263e945eedb47776e36e2e817991977e6ef6d
Reviewed-on: https://webrtc-review.googlesource.com/c/107709
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25359}
2018-10-25 12:04:37 +00:00
Mirko Bonadei
977b46a59b Export symbols needed by the Chromium component build (part 7).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I1081af5ecf7ba55a7415e09e45357b783cf300aa
Reviewed-on: https://webrtc-review.googlesource.com/c/107708
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25358}
2018-10-25 11:41:16 +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
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Mirko Bonadei
ac19414512 Export symbols needed by the Chromium component build (part 6).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5
Reviewed-on: https://webrtc-review.googlesource.com/c/107400
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25298}
2018-10-23 06:48:51 +00:00
Yves Gerey
2e00abc98e Reland "[cleanup] Remove useless includes."
Reason for reland: Downstream project fixed.

Original change's description:

> [cleanup] Remove useless includes.
>
> Manual cleanup guided by include-what-you-use diagnostic.
>
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

Bug: webrtc:8311
Change-Id: Id6ec4aeb798886a90ace640a190eaf16497ba31b
Reviewed-on: https://webrtc-review.googlesource.com/c/104120
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25034}
2018-10-08 07:44:19 +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
Oleh Prypin
96a0f61917 Revert "[cleanup] Remove useless includes."
This reverts commit be8b5348c76105f8fe869b0cae4065ddca106419.

Reason for revert: Breaks downstream project

Original change's description:
> [cleanup] Remove useless includes.
> 
> Manual cleanup guided by include-what-you-use diagnostic.
> 
> Bug: webrtc:8311
> Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
> Reviewed-on: https://webrtc-review.googlesource.com/c/103320
> Commit-Queue: Yves Gerey <yvesg@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25013}

TBR=phoglund@google.com,phoglund@webrtc.org,yvesg@webrtc.org

Change-Id: I7a6e1cdfef685173b76f234ad598083043dcd9a0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8311
Reviewed-on: https://webrtc-review.googlesource.com/c/104022
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25015}
2018-10-05 13:13:45 +00:00
Yves Gerey
be8b5348c7 [cleanup] Remove useless includes.
Manual cleanup guided by include-what-you-use diagnostic.

Bug: webrtc:8311
Change-Id: I00be03392cc7ee005101427ea7dc701621ccea68
Reviewed-on: https://webrtc-review.googlesource.com/c/103320
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25013}
2018-10-05 11:51:06 +00:00
Johnny Lee
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension)
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 22:35:30 +00:00
Mirko Bonadei
8fdcac3f06 Remove clang:find_bad_constructs suppression from call:call.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
2018-08-29 11:57:00 +00:00
Jiawei Ou
e4749c2bf0 The default logic for creating video bitrate allocator.
It is a mirror of `VideoCodecInitializer::CreateBitrateAllocator`

Bug: webrtc:9513
Change-Id: Ib2e83e9f757387a2f6f6101d5d21512f1d507a95
Reviewed-on: https://webrtc-review.googlesource.com/92320
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24413}
2018-08-23 20:50:32 +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
Jiawei Ou
51ef008ef6 Video bitrate allocator factory interface for custom video bitrate allocator
Bug: webrtc:9513
Change-Id: If5719f4f71a2a9e44b2aaa0b592d3921b68ba5d9
Reviewed-on: https://webrtc-review.googlesource.com/92280
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#24279}
2018-08-14 17:57:22 +00:00
Karl Wiberg
c2342031f4 Remove rtc::{Make,Wrap}Unique and their header file + unit tests
We've switched to absl::make_unique and absl::WrapUnique.

Bug: webrtc:9473
Change-Id: I08aef72d52b571c511c0f4adb4c68d6cc2654192
Reviewed-on: https://webrtc-review.googlesource.com/87262
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24225}
2018-08-08 10:45:52 +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
Emircan Uysaler
d4c16b131f Extract color space from H264 decoder
Makes use of ColorSpace class to extract info from H264 stream.

Bug: webrtc:9522
Change-Id: I651d16707260bb2867b1eda95dd4956d62c47279
Reviewed-on: https://webrtc-review.googlesource.com/90180
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24085}
2018-07-24 17:07:16 +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
Mirko Bonadei
5e6dcb2e64 Enable clang::find_bad_constructs for api/video/test.
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

TBR: magjed@webrtc.org
Bug: webrtc:9251, webrtc:163
Change-Id: Ida449dae66ed5a9e1b36b14a988dee8ecfbea18f
Reviewed-on: https://webrtc-review.googlesource.com/89582
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24052}
2018-07-20 12:41:07 +00:00
Stefan Holmer
dbdb3a0079 Refactoring PayloadRouter.
- 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}
2018-07-17 14:46:15 +00:00
Stefan Holmer
f70446874a Reland "Move allocation and rtp conversion logic out of payload router."
This reverts commit c2406e4eaf7703c6c64d21318186adda791e09fd.

Reason for revert: Reland by removing the conflict with the broken CL.

Original change's description:
> Revert "Move allocation and rtp conversion logic out of payload router."
> 
> This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.
> 
> Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220
> 
> This causes a merge conflict. So need to revert this first.
> 
> Original change's description:
> > Move allocation and rtp conversion logic out of payload router.
> > 
> > Makes it easier to write tests, and allows for moving rtp module
> > ownership into the payload router in the future.
> > 
> > The RtpPayloadParams class is split into declaration and definition and
> > moved into separate files.
> > 
> > Bug: webrtc:9517
> > Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> > Reviewed-on: https://webrtc-review.googlesource.com/88564
> > Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23983}
> 
> TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org
> 
> Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9517
> Reviewed-on: https://webrtc-review.googlesource.com/88821
> Reviewed-by: JT Teh <jtteh@webrtc.org>
> Commit-Queue: JT Teh <jtteh@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23991}

TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,lliuu@webrtc.org,jtteh@webrtc.org,tkchin@webrtc.org

Change-Id: I154145cdbc668feee86dbe78860147a6954fee6c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/89020
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23996}
2018-07-17 08:17:44 +00:00
JT Teh
c2406e4eaf Revert "Move allocation and rtp conversion logic out of payload router."
This reverts commit 1da4d79ba3275b3fa48cad3b2c0949e0d3b7afe7.

Reason for revert: Need to revert https://webrtc-review.googlesource.com/c/src/+/88220

This causes a merge conflict. So need to revert this first.

Original change's description:
> Move allocation and rtp conversion logic out of payload router.
> 
> Makes it easier to write tests, and allows for moving rtp module
> ownership into the payload router in the future.
> 
> The RtpPayloadParams class is split into declaration and definition and
> moved into separate files.
> 
> Bug: webrtc:9517
> Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
> Reviewed-on: https://webrtc-review.googlesource.com/88564
> Commit-Queue: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23983}

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

Change-Id: I342c4bf483d975c87c706fe7f76f44e2dc60fe4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9517
Reviewed-on: https://webrtc-review.googlesource.com/88821
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23991}
2018-07-16 20:38:02 +00:00
Emircan Uysaler
800787f03b Add color space information to webrtc::VideoFrame and extract from VP9
This CL is the first step for introducing color space information in webrtc.
- Add ColorSpace class listing color profiles.
- Add ColorSpace as a member of webrtc::VideoFrame.
- Make use of this class by extracting info from VP9 decoder.

Bug: webrtc:9522
Change-Id: I5e2514efee2a193bddb4459261387f2d40e936ad
Reviewed-on: https://webrtc-review.googlesource.com/88540
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23988}
2018-07-16 17:28:17 +00:00
Stefan Holmer
1da4d79ba3 Move allocation and rtp conversion logic out of payload router.
Makes it easier to write tests, and allows for moving rtp module
ownership into the payload router in the future.

The RtpPayloadParams class is split into declaration and definition and
moved into separate files.

Bug: webrtc:9517
Change-Id: I8700628edff19abcacfe8d3a20e4ba7476f712ad
Reviewed-on: https://webrtc-review.googlesource.com/88564
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23983}
2018-07-16 13:34:37 +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
Emircan Uysaler
901e0ffc33 Add bit depth information to PlanarYuvBuffer
For HDR codecs, we expect to receive input that has 10-bit color depth. But
currently, WebRTC assumes only 8-bit input and output. This CL adds k010
format that represent this input.

Bug: webrtc:9376
Change-Id: Ie7df64b0eddb0752b493e0457a49083a1e87ba51
Reviewed-on: https://webrtc-review.googlesource.com/81920
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23749}
2018-06-26 20:23:37 +00:00
Björn Terelius
52f53d5419 Revert "Add Timestamp accessor methods to the EncodedImage class."
This reverts commit f34d467b03da4f20a1d036a20966fcad43d2433f.

Reason for revert: Seems to break downstream project.

Original change's description:
> Add Timestamp accessor methods to the EncodedImage class.
> 
> Bug: webrtc:9378
> Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
> Reviewed-on: https://webrtc-review.googlesource.com/82100
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23734}

TBR=brandtr@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org

Change-Id: I3aa0c0119426886bc583c918aae862eb7f4b6b63
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9378
Reviewed-on: https://webrtc-review.googlesource.com/85600
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23739}
2018-06-26 11:52:45 +00:00
Niels Möller
f34d467b03 Add Timestamp accessor methods to the EncodedImage class.
Bug: webrtc:9378
Change-Id: I59bf14f631f92f0f4e05f60d4af25641a23a53f9
Reviewed-on: https://webrtc-review.googlesource.com/82100
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23734}
2018-06-26 09:40:18 +00:00
Danil Chapovalov
0bc58cf876 Replace rtc::Optional with absl::optional in api
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'api'
Then undo changes to optional target itself and optional_unittests

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: I44093da213369d6a502e33792c694f620f53b779
Reviewed-on: https://webrtc-review.googlesource.com/84621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23707}
2018-06-21 12:50:03 +00:00
Niels Möller
7e9a619599 Add setter method EncodedFrame::SetTimestamp.
Similar to the getter method (see
https://webrtc-review.googlesource.com/83123), this is a preparation
for inheriting the method from the base class, and delete the
corresponding redundant timestamp member.

Bug: webrtc:9378
Change-Id: Idbb48e7058c94d6d4aa9a2b19e608ef08c2e35b4
Reviewed-on: https://webrtc-review.googlesource.com/83726
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23665}
2018-06-19 15:10:29 +00:00
Niels Möller
709c82200f Add nisse@ as owner of api/video/
nisse has been involved in the design of most of the interfaces here.

NOTRY=True

Bug: None
Change-Id: I37b7eb45892038c1c6d567fce1793bf0bcaca082
Reviewed-on: https://webrtc-review.googlesource.com/83981
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23661}
2018-06-19 14:02:11 +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
Niels Möller
037b37a192 Add implementation of EncodedFrame::Timestamp.
This is a preparation for inheriting the method from the base class,
and delete the corresponding redundant timestamp member.

TBR: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: I27a0ec83fb20ac3da58ba32b86cf794a154deca0
Reviewed-on: https://webrtc-review.googlesource.com/83123
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23594}
2018-06-13 12:23:49 +00:00
Niels Möller
2ac64467c4 Document that preferred VideoFrame constructor takes no RTP timestamp.
And update most internal calls to use it.

Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
2018-06-11 18:42:40 +00:00
Ilya Nikolaevskiy
6bac5c182a Remove TimingFrameFlags from its old place after it was moved
Bug: webrtc:9351
Change-Id: I97df702adddb600804fe643e17d84331073f4ba1
Reviewed-on: https://webrtc-review.googlesource.com/81681
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23564}
2018-06-11 16:38:49 +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
Ilya Nikolaevskiy
3a79a9a290 Remove deprecated API methods in video pipeline
Bug: none
Change-Id: I3c3d493f9e14a93868c86fa94ef7269126bd9877
Reviewed-on: https://webrtc-review.googlesource.com/80482
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23514}
2018-06-05 08:26:05 +00:00
Anders Carlsson
51e23aed9e Remove built-in sw codecs from decoder_database.
All decoders are injectable, no need to create built-in codecs from
there.

Bug: webrtc:7925
Change-Id: Iabf3d59a8e4d721ad29386acbf138b7e5992ce5e
Reviewed-on: https://webrtc-review.googlesource.com/72441
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23397}
2018-05-25 09:54:18 +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
Jonas Olsson
866d6dcdba Remove the remaining non-test stringstreams from api/
Bug: webrtc:8982
Change-Id: Ie54ed24a609398228a69bdd92728ebf679cf3fe3
Reviewed-on: https://webrtc-review.googlesource.com/76561
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23239}
2018-05-15 11:03:01 +00:00
Niels Möller
c6ce9c5938 New file api/video/BUILD.gn
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.

Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
Erik Språng
566124a6df Move BitrateAllocation to api/ and rename it VideoBitrateAllocation
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}
2018-04-23 15:31:27 +00:00
Niels Möller
be682d47ac Fix chromium warnings for SdpVideoFormat.
Bug: webrtc:163
Change-Id: I29ad3c00116692f047456df7721ba636bbb2ca89
Reviewed-on: https://webrtc-review.googlesource.com/64723
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22618}
2018-03-27 08:11:21 +00:00
philipel
9718711dee VideoStreamDecoderImpl implementation, part 1.
In this CL the OnFrame function is implemented.

Bug: webrtc:8909
Change-Id: I68488a033e86eadd0b16d091faad14e9cda7cc36
Reviewed-on: https://webrtc-review.googlesource.com/64121
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22583}
2018-03-23 13:58:55 +00:00
Karl Wiberg
29e7bee330 Move aligned memory utilities to rtc_base/memory/
This moves them from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.

BUG=webrtc:8445

Change-Id: I6dc34fe662f5d87b3b5288d33055345bc6bf91db
Reviewed-on: https://webrtc-review.googlesource.com/21164
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22567}
2018-03-22 14:13:24 +00:00