248 Commits

Author SHA1 Message Date
Taylor Brandstetter
1204448a68 Revert "Reland "Rename stereo video codec to multiplex""
This reverts commit 4954a77cf81e6793245f52d485834acd3e6eab1c.

Reason for revert: Breaks downstream build which was depending on the name "kVideoCodecStereo". Will need to do some sort of trickery to make this change without breaking the relevant code. Sorry. :(

Original change's description:
> Reland "Rename stereo video codec to multiplex"
> 
> This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
> This was reverted because of breaking internal build. I contacted sheriff
> and looked at logs but cannot find anything related to this CL. This was landed
> with #3850 build which caused exception, but 3847-3855 seem to all have failed.
> I am relanding to see if it will work this time or it will give some related
> error message that can guide me.
> 
> Original change's description:
> > Rename stereo video codec to multiplex
> >
> > This CL only does the rename from"stereo" to multiplex". With this we have a
> > better name that doesn't clash with audio's usage of stereo.
> >
> > Bug: webrtc:7671
> > Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> > Reviewed-on: https://webrtc-review.googlesource.com/43242
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21769}
> 
> TBR=niklas.enbom@webrtc.org
> 
> Bug: webrtc:7671
> Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
> Reviewed-on: https://webrtc-review.googlesource.com/44520
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21780}

TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: I0a71327c2ddfdd030b1e058cd6a41b1689836719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21783}
2018-01-27 00:45:20 +00:00
Emircan Uysaler
4954a77cf8 Reland "Rename stereo video codec to multiplex"
This is a reland of bbdabe50db0cf09f6007dda12a6476dc4602b174.
This was reverted because of breaking internal build. I contacted sheriff
and looked at logs but cannot find anything related to this CL. This was landed
with #3850 build which caused exception, but 3847-3855 seem to all have failed.
I am relanding to see if it will work this time or it will give some related
error message that can guide me.

Original change's description:
> Rename stereo video codec to multiplex
>
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
>
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}

TBR=niklas.enbom@webrtc.org

Bug: webrtc:7671
Change-Id: I5934abad1ce28acf02842ea8ee2af7768a826eb8
Reviewed-on: https://webrtc-review.googlesource.com/44520
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21780}
2018-01-26 21:11:54 +00:00
Mirko Bonadei
65ce31158f Removing useless dependencies on //testing/gmock.
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.

Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).

This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.

Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.

TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
2018-01-26 13:34:12 +00:00
Ivo Creusen
6bc7bb659e Revert "Rename stereo video codec to multiplex"
This reverts commit bbdabe50db0cf09f6007dda12a6476dc4602b174.

Reason for revert: This breaks the internal build.

Original change's description:
> Rename stereo video codec to multiplex
> 
> This CL only does the rename from"stereo" to multiplex". With this we have a
> better name that doesn't clash with audio's usage of stereo.
> 
> Bug: webrtc:7671
> Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
> Reviewed-on: https://webrtc-review.googlesource.com/43242
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21769}

TBR=sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: Icf019cb09e07de45821d31d7d8ea7707d01346ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/44360
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21774}
2018-01-26 12:44:54 +00:00
Emircan Uysaler
bbdabe50db Rename stereo video codec to multiplex
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.

Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
2018-01-25 23:16:04 +00:00
Per Kjellander
a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00
Per Kjellander
c73e1f4378 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.

Reason for revert: 

Breaks Chrome FYI:

/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
  -> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
    static_library(target_name) {
    ^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
  //third_party/webrtc/*
  //third_party/webrtc_overrides/*
]

 https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout

Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> 
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
> 
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
> 
> BUG=webrtc:8254
> 
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
2018-01-10 15:14:54 +00:00
Karl Wiberg
588c548657 GN rtc_* templates: Set default visibility to webrtc_root + "/*"
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.

API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.

BUG=webrtc:8254

Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
2018-01-10 13:08:11 +00:00
Todd Wong
6728003bcf Skip H246 scaling lists in SPS packets
This code is originally written by marc@frankensteinmotorworks.com

Bug: webrtc:8275
Change-Id: I35e6d21b12e71199e0209ff91740d95c9df3bd10
Reviewed-on: https://webrtc-review.googlesource.com/36520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21528}
2018-01-09 10:22:30 +00:00
Patrik Höglund
be214a26f8 Move videosinkinterface.h to common_video to solve a circular dep.
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.

Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
Mirko Bonadei
0594a7ca5d Stop using public_deps in common_video/.
Bug: webrtc:8603
Change-Id: I467f07a6bd07585455d1d1f9e8bcfa59f0dce9f0
Reviewed-on: https://webrtc-review.googlesource.com/34185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21359}
2017-12-19 12:50:00 +00:00
Åsa Persson
aa329e7cc3 Reland: googBandwidthLimitedResolution stat is not always set depending on configuration.
TBR=brandtr@webrtc.org,stefan@webrtc.org

Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.

Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.

Bug: webrtc:8643
Change-Id: I553cea30dcda34b753b5224f15094a1b7b70a750
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#21249}
Reviewed-on: https://webrtc-review.googlesource.com/33360
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21319}
2017-12-18 11:20:13 +00:00
Patrik Höglund
3e113438b1 Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00
Guido Urdaneta
62e9ebe589 Revert "googBandwidthLimitedResolution stat is not always set depending on configuration."
This reverts commit 59283e4c66d038a00923736685457f4b53f922fe.

Reason for revert: This CL is preventing rolls into Chromium because it fails to compile with MSVC.

Sample error log:

[13258/43857] CXX obj/third_party/webrtc/video/video/send_statistics_proxy.obj
FAILED: obj/third_party/webrtc/video/video/send_statistics_proxy.obj 
ninja -t msvc -e environment.x64 -- E:\b\c\goma_client/gomacc.exe "e:\b\c\win_toolchain\vs_files\a9e1098bba66d2acccc377d5ee81265910f29272\vc\tools\msvc\14.11.25503\bin\hostx64\x64/cl.exe" /nologo /showIncludes  @obj/third_party/webrtc/video/video/send_statistics_proxy.obj.rsp /c ../../third_party/webrtc/video/send_statistics_proxy.cc /Foobj/third_party/webrtc/video/video/send_statistics_proxy.obj /Fd"obj/third_party/webrtc/video/video_cc.pdb"
../../third_party/webrtc/video/send_statistics_proxy.cc(217): error C2220: warning treated as error - no 'object' file generated
../../third_party/webrtc/video/send_statistics_proxy.cc(217): warning C4267: 'initializing': conversion from 'size_t' to 'int', possible loss of data
../../third_party/webrtc/video/send_statistics_proxy.cc(632): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data


Original change's description:
> googBandwidthLimitedResolution stat is not always set depending on configuration.
> 
> Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
> OnEncodedImage callback.
> 
> Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
> on info that is reported to SendStatisticsProxy::OnEncodedImage.
> 
> Bug: webrtc:8643
> Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
> Reviewed-on: https://webrtc-review.googlesource.com/31460
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21249}

TBR=brandtr@webrtc.org,asapersson@webrtc.org,stefan@webrtc.org

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

Bug: webrtc:8643
Change-Id: Ib9ef55b8894ea72236a5dc1e9a839adecd401afb
Reviewed-on: https://webrtc-review.googlesource.com/33100
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21284}
2017-12-14 17:35:53 +00:00
Mirko Bonadei
401d056891 Removing $rtc_libyuv_dir and removing useless dependencies on libyuv.
This CL removes the following GN variables: rtc_build_libyuv,
rtc_libyuv_dir (as requested in webrtc:7906).
It also removes some unneeded dependencies on //third_party/libyuv.

WebRTC targets were using public_deps to depend on //third_party/libyuv
and this created a build graph where targets that were depending on
//third_party/libyuv were not declaring the dependency to GN because
they were somehow getting it from another target that was exposing
//third_party/libyuv header files even if it wasn't directly depending
on it.

Bug: webrtc:8605, webrtc:7906
Change-Id: If71f7988fd80421dc2ad887cf94c2ac66366c3fb
Reviewed-on: https://webrtc-review.googlesource.com/32201
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21275}
2017-12-14 11:18:33 +00:00
Åsa Persson
59283e4c66 googBandwidthLimitedResolution stat is not always set depending on configuration.
Currently |bw_resolutions_disabled| is set per VP8EncoderImpl instance and reported via
OnEncodedImage callback.

Instead move logic to SendStatisticsProxy to determine if resolution is bw limited or not based
on info that is reported to SendStatisticsProxy::OnEncodedImage.

Bug: webrtc:8643
Change-Id: I6c148e3507a0f04a793775b9f84ce54028b64d0f
Reviewed-on: https://webrtc-review.googlesource.com/31460
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21249}
2017-12-13 14:32:21 +00:00
Niels Möller
6b642f730c Delete EncodedFrameObserver::OnEncodeTiming.
This callback was used only by the PrintSamplesToFile feature of
video_quality_test, which looks like it has been broken for some time
(due to mixup of capture time and ntp time).

Bug: webrtc:8504
Change-Id: I7d2b55405caeffda582ae0d6fb0e7dfdfce4c5a9
Reviewed-on: https://webrtc-review.googlesource.com/31420
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21211}
2017-12-11 16:41:46 +00:00
Mirko Bonadei
654320666d Including libyuv headers using fully qualified paths.
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}
2017-12-11 15:51:26 +00:00
Mirko Bonadei
0250be51be Stop using public_deps to depend on libyuv.
A lot of WebRTC targets were depending on //third_party/libyuv using
public_deps instead of deps. This causes issues because a the
inclusion of libyuv headers is not declared to the build system and
this creates hidden dependencies that put the modularity of the project
at risk.

Bug: webrtc:8603
Change-Id: Ide0ceb84eb5640ae664dc782f3a722b55c3b601a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28120
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21039}
2017-12-04 14:16:08 +00:00
Mirko Bonadei
ad62792c5d Fixing hidden dependencies.
Header files base/videosinkinterface.h and base/videosourceinterface.h
were not part of any target (because they cause 2 dependency cycles).

This CL uncomment them so GN can keep dependencies under control, the
2 dependency cycles will be removed as part of webrtc:6828.

Bug: webrtc:6828
Change-Id: I5c5580facc010ba619e105a9b8a572ac70169a01
Reviewed-on: https://webrtc-review.googlesource.com/27621
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20970}
2017-12-01 09:30:11 +00:00
Emircan Uysaler
90612a681b Reland "Add stereo codec header and pass it through RTP"
This is a reland of 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}

TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org

Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
2017-11-30 01:44:19 +00:00
Philip Eliasson
deb866360a Revert "Add stereo codec header and pass it through RTP"
This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1.

Reason for revert: Breaks downstream project.

Original change's description:
> Add stereo codec header and pass it through RTP
> 
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
> 
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
> 
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}

TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org

Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
2017-11-29 11:39:41 +00:00
Emircan Uysaler
20f2133d5d Add stereo codec header and pass it through RTP
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.

This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
2017-11-28 18:43:43 +00:00
Oskar Sundbom
92016ce5a4 Optional: Use nullopt and implicit construction in /common_video
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=magjed@webrtc.org

Bug: None
Change-Id: I0eddc997560894dc661f521f6096e2d834216cee
Reviewed-on: https://webrtc-review.googlesource.com/23608
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20895}
2017-11-27 17:03:12 +00:00
Karl Wiberg
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
Ilya Nikolaevskiy
76f2a85027 In generic encoder ensure capture timestamp is preserved.
Because some hardware encoders may not preserve capture timestamp, use
rtp timestamp as a key to capture time storage and restore corrupted
capture timestamps whenever possible.

Bug: webrtc:8497
Change-Id: Ib3449313c65e864a89cd645a3f5dec99aa9cf8e4
Reviewed-on: https://webrtc-review.googlesource.com/23620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20732}
2017-11-17 16:06:17 +00:00
mallikarjun82
12e555b715 Delete wrapper API ConvertToI420 for YUV conversion to I420
Directly use the libyuv API for YUV conversion to I420

Bug: None
Change-Id: Iea6e8fa8f7179c800ea850305170002398cb00dc
Reviewed-on: https://webrtc-review.googlesource.com/17260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20681}
2017-11-15 11:10:20 +00:00
Patrik Höglund
b5b5bcee72 Separate i420 and i444 implementations to separate targets.
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.

Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
   new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
   video_frame_api, since it no longer contains i420 code

Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 14:27:39 +00:00
Emircan Uysaler
574eaa4cda Add alpha channel to VideoFrameBuffer containers
- Add alpha accessors to PlanarYuvBuffer interface, null by defualt.
- Add WrapI420ABuffer() that creates a container which implements these
accessors.
- Show the use via StereoDecoderAdapter.

This CL is the step 2 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: Id5691cde00088ec811b63d89080d33ad2d6e3939
Reviewed-on: https://webrtc-review.googlesource.com/21130
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20635}
2017-11-10 17:13:25 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Peter Boström
de6914508e Remove pbos@webrtc.org from all OWNERS.
Bug: None
Change-Id: I49c4df3873f359c20f46a64592a05c3d001b708d
Reviewed-on: https://webrtc-review.googlesource.com/17720
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20517}
2017-11-01 08:03:46 +00:00
VladimirTechMan
7ee54a3cfd Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv"
This fix brings common_video/BUILD.gn on a par with other BUILD.gn files
that use $rtc_libyuv_dir to include libyuv header files when building
WebRTC. Doing so eliminates the failure in the building process when
a non-default $rtc_libyuv_dir is used.

Bug: webrtc:8399
Change-Id: I1bac0285f1869a334d116d0e1371aa10204137e5
Reviewed-on: https://webrtc-review.googlesource.com/11140
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20315}
2017-10-16 16:05:37 +00:00
Niels Möller
84255bbe3b Add explicit includes of refcountedobject.h where it is used.
This is in preparation for deleting the include in rtc_base/refcount.h,
but that change has to wait for some downstream applications to 
not rely in the indirect include.

Partial reland of "Make rtc_base/refcount.h self contained, not including refcountedobject.h."

This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
> 
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
> 
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}

Bug: webrtc:8270
Change-Id: I63a42712f6c1ec83823c629d1a954fd1a04d4a6c
Reviewed-on: https://webrtc-review.googlesource.com/7281
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20185}
2017-10-06 13:00:14 +00:00
Niels Moller
fb26f85b79 Revert "Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h.""
This reverts commit bf6937a8e90f94ac0ae50a296574b9f262b0932f.

Reason for revert: Broke internal projects.

Original change's description:
> Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
> 
> This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
> Original change's description:
> > Make rtc_base/refcount.h self contained, not including refcountedobject.h.
> > 
> > The refcount.h file doesn't depend on anything from
> > refcountedobject.h. The motivation of this change to make it possible
> > to add additional declarations to refcount.h, and include it from
> > refcountedobject.h.
> > 
> > Bug: webrtc:8270
> > Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> > Reviewed-on: https://webrtc-review.googlesource.com/5760
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20106}
> 
> Bug: webrtc:8270
> Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
> Reviewed-on: https://webrtc-review.googlesource.com/5840
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20180}

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

Change-Id: I342b241f5bb707b59ccf2d15a1a5befecb53a52e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/7280
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20181}
2017-10-06 11:05:55 +00:00
Niels Möller
bf6937a8e9 Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This is a reland of b7239a9dc825ddb25dbc800aed3a065163b9a10e
Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
> 
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
> 
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}

Bug: webrtc:8270
Change-Id: I9738f6680ab52d0f43639a1a39175fdba5957681
Reviewed-on: https://webrtc-review.googlesource.com/5840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20180}
2017-10-06 10:20:48 +00:00
Niels Moller
d25fa78daf Revert "Make rtc_base/refcount.h self contained, not including refcountedobject.h."
This reverts commit b7239a9dc825ddb25dbc800aed3a065163b9a10e.

Reason for revert: Broke chromium mac build, compilation failures on content/renderer/media/webrtc/webrtc_video_frame_adapter.h.

Original change's description:
> Make rtc_base/refcount.h self contained, not including refcountedobject.h.
> 
> The refcount.h file doesn't depend on anything from
> refcountedobject.h. The motivation of this change to make it possible
> to add additional declarations to refcount.h, and include it from
> refcountedobject.h.
> 
> Bug: webrtc:8270
> Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
> Reviewed-on: https://webrtc-review.googlesource.com/5760
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20106}

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

Change-Id: I7334597cc8979ba9cfaff526967084349ef27f3c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8270
Reviewed-on: https://webrtc-review.googlesource.com/5800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20107}
2017-10-03 09:49:04 +00:00
Niels Möller
b7239a9dc8 Make rtc_base/refcount.h self contained, not including refcountedobject.h.
The refcount.h file doesn't depend on anything from
refcountedobject.h. The motivation of this change to make it possible
to add additional declarations to refcount.h, and include it from
refcountedobject.h.

Bug: webrtc:8270
Change-Id: I24f6131f471e675570968d00065ff9b1f55e3373
Reviewed-on: https://webrtc-review.googlesource.com/5760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20106}
2017-10-03 09:37:30 +00:00
Niels Möller
032f410ae0 Delete unneeded includes of pathutils.h
Bug: webrtc:6424
Change-Id: I73b2bc747c67d2fe2ad888dde9c2815a6d9aceaa
Reviewed-on: https://webrtc-review.googlesource.com/4760
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20059}
2017-10-02 07:05:19 +00:00
Niels Möller
d8970dbd42 Delete unneeded includes of fileutils.h
It is now used only by FileRotatingStream.

Bug: webrtc:6424
Change-Id: I216b20baadae836d24c39899efe4cb45c2935f41
Reviewed-on: https://webrtc-review.googlesource.com/4720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20040}
2017-09-29 12:39:09 +00:00
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
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}
2017-09-15 04:25:06 +00:00
niklase@google.com
d784e5535c git-svn-id: http://webrtc.googlecode.com/svn/trunk@160 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:26:13 +00:00
mikhal@google.com
b5427cbd35 Changing JPEG API to to accept rawImage and encodedImage; moved video_image.h from modules/video_coding/codecs to common_video/interface, and some general re-write to JPEG, especially with regard to memory handling. Required VCM/ViE changes are also included.
Review URL: http://webrtc-codereview.appspot.com/55002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@139 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-01 01:17:49 +00:00
mikhal@google.com
21a4405d01 VPLIB/Interpolation - Delete decode buffer only if too small, this required an API change. In addition, done some clean up and updated test and related code in VCM.
Review URL: http://webrtc-codereview.appspot.com/46003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@131 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-29 17:00:03 +00:00
mikhal@google.com
2360c09549 VPLIB: Fixing a bug in ConvertYUY2TOI420 + some code clean-up
Review URL: http://webrtc-codereview.appspot.com/46001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@106 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-20 18:03:34 +00:00
leozwang@google.com
0b0c28c495 add android makefile, some modification in vpx makefile to build encoder from c source for now
Review URL: http://webrtc-codereview.appspot.com/29012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@50 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-07 17:24:39 +00:00
niklase@google.com
01813fe945 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:21:31 +00:00