12 Commits

Author SHA1 Message Date
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
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
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
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
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
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
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