45 Commits

Author SHA1 Message Date
kjellander@webrtc.org
2db1dbb2ca Remove references to build_with_libjingle and libjingle_java GYP variables.
These were removed a while back in https://codereview.webrtc.org/1457053003

TBR=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1790753004 .

Cr-Commit-Position: refs/heads/master@{#11967}
2016-03-12 05:34:35 +00:00
kjellander
e26e78784b Roll chromium_revision ee31124..508edd3 (378158:379249)
This includes renaming back libvpx_new to libvpx in
https://codereview.chromium.org/1765703002

Add symlink to src/mojo as workaround while figuring out how to fix
this upstream in Chromium. See webrtc:5629.

Change log: ee31124..508edd3
Full diff: ee31124..508edd3

Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6d49157..708db16
* src/third_party/libvpx_new/source/libvpx: https://chromium.googlesource.com/webm/libvpx.git/+log/89cc682..None
* src/tools/swarming_client: https://chromium.googlesource.com/external/swarming.client.git/+log/a72f46e..df6e95e
DEPS diff: https://chromium.googlesource.com/chromium/src/+/ee31124..508edd3/DEPS

No update to Clang.

BUG=webrtc:5629
TBR=marpan@webrtc.org, stefan@webrtc.org,
NOTRY=True

Review URL: https://codereview.webrtc.org/1766643002

Cr-Commit-Position: refs/heads/master@{#11879}
2016-03-04 22:39:32 +00:00
pkotwicz
a75339c40d Set rtc_prefer_fixed_point to true on arm64 to match GYP
BUG=None
NOTRY=True

Review URL: https://codereview.webrtc.org/1681183002

Cr-Commit-Position: refs/heads/master@{#11555}
2016-02-10 18:21:11 +00:00
hbos
62756ee411 Default build flag |rtc_use_h264| to |proprietary_codecs|
if not on Android/iOS.

This is a re-land of https://codereview.webrtc.org/1674103002/.
The reason Chromium FYI turned red was due to deps not
being relative. See kjellander's CL:
https://codereview.webrtc.org/1681493002/.

This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.

This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).

Third time's the charm?

TBR=kjellander@webrtc.org
BUG=chromium:500605, chromium:468365

Review URL: https://codereview.webrtc.org/1675143003

Cr-Commit-Position: refs/heads/master@{#11523}
2016-02-08 10:57:06 +00:00
hbos
a81f6a3fc0 Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1674103002/ )
Reason for revert:
Chromium FYI turns red.

Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> Note: This is a re-land of
> https://codereview.webrtc.org/1660403004/. Reverting it
> was not necessary.
>
> TBR=kjellander@webrtc.org
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/10b9dd7ab1a8c3f80b2d2924be658e43131a4fbe
> Cr-Commit-Position: refs/heads/master@{#11517}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365

Review URL: https://codereview.webrtc.org/1675113002

Cr-Commit-Position: refs/heads/master@{#11518}
2016-02-07 23:05:26 +00:00
hbos
10b9dd7ab1 Default build flag |rtc_use_h264| to |proprietary_codecs|
if not on Android/iOS.

This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.

This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).

Note: This is a re-land of
https://codereview.webrtc.org/1660403004/. Reverting it
was not necessary.

TBR=kjellander@webrtc.org
BUG=chromium:500605, chromium:468365

Review URL: https://codereview.webrtc.org/1674103002

Cr-Commit-Position: refs/heads/master@{#11517}
2016-02-07 22:40:46 +00:00
hbos
c09525a547 Revert of Default build flag |rtc_use_h264| to |proprietary_codecs| if not on Android/iOS. (patchset #1 id:1 of https://codereview.webrtc.org/1660403004/ )
Reason for revert:
Trybots red? Don't have time to intvestigate

Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/7cd94f66ebfe5bf808d7dcb8da069d35f4a2b31a
> Cr-Commit-Position: refs/heads/master@{#11506}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365

Review URL: https://codereview.webrtc.org/1677623002

Cr-Commit-Position: refs/heads/master@{#11508}
2016-02-05 19:02:47 +00:00
hbos
7cd94f66eb Default build flag |rtc_use_h264| to |proprietary_codecs|
if not on Android/iOS.

This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.

This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).

BUG=chromium:500605, chromium:468365

Review URL: https://codereview.webrtc.org/1660403004

Cr-Commit-Position: refs/heads/master@{#11506}
2016-02-05 18:31:26 +00:00
hbos
c5a39c2591 H264: Thread-safe InitializeFFmpeg. Flag to control if InitializeFFmpeg should be called.
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.

In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.

BUG=chromium:500605, chromium:468365, webrtc:5427

Review URL: https://codereview.webrtc.org/1639273002

Cr-Commit-Position: refs/heads/master@{#11456}
2016-02-02 10:30:57 +00:00
hbos
bab934bffe H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
It works on all platforms except Android and iOS (FFmpeg limitation).

Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.

Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)

Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)

NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424

Review URL: https://codereview.webrtc.org/1306813009

Cr-Commit-Position: refs/heads/master@{#11390}
2016-01-27 09:36:07 +00:00
hbos
902c03e724 rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni.
BUG=468365
NOTRY=True

Review URL: https://codereview.webrtc.org/1601813005

Cr-Commit-Position: refs/heads/master@{#11333}
2016-01-21 11:34:47 +00:00
kwiberg
f8c2baca4e Add a gyp/gn variable for whether to use iLBC or not
BUG=webrtc:5415

Review URL: https://codereview.webrtc.org/1578953003

Cr-Commit-Position: refs/heads/master@{#11291}
2016-01-18 14:38:40 +00:00
hbos
a9a1d2acaf H.264: Default flags and pulling in openh264 and ffmpeg.
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.

BUG=468365

Review URL: https://codereview.webrtc.org/1575913003

Cr-Commit-Position: refs/heads/master@{#11204}
2016-01-11 18:19:06 +00:00
Peter Boström
bd05f0ba52 Unconditionally build VP9 support.
Broken for PeerConnection either way (since VP9 support is announced)
and would fail on a CHECK apart from generating incorrect
offers/answers. This isn't a flag that we want to support, so it's
better to remove the foot-shooting gun.

BUG=
R=asapersson@webrtc.org, kjellander@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1451663002 .

Cr-Commit-Position: refs/heads/master@{#10676}
2015-11-17 14:27:41 +00:00
noahric
23725e09c6 Remove ICU usage from jni_helpers.cc.
JNI already has jstring<->UTF8 string conversion, so using that should
save ~1mb off android binaries (ICU is *large*), probably around
300-400k after compression.

BUG=

Review URL: https://codereview.webrtc.org/1430023005

Cr-Commit-Position: refs/heads/master@{#10545}
2015-11-06 21:56:11 +00:00
Henrik Kjellander
b79472a4fb Roll chromium_revision c089d37..159828f (353662:353696)
Due to https://codereview.chromium.org/1397493004 we're now adding
a build_overrides directory in WebRTC. Thanks to this, we no longer
need to pass --args="build_with_chromium=false" when running GN in
standalone WebRTC.

Change log: c089d37..159828f
Full diff: c089d37..159828f

No dependencies changed.
No update to Clang.

BUG=webrtc:5070,chromium:541791
TBR=tommi@webrtc.org
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal

Review URL: https://codereview.webrtc.org/1403453003 .

Cr-Commit-Position: refs/heads/master@{#10270}
2015-10-14 06:14:10 +00:00
kjellander
d6024e3c34 Roll chromium_revision 310ea93..8cf53d6 (349094:351112)
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).

Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).

Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/

Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).

Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS

Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh

BUG=481034, 535973
TBR=marpan@webrtc.org

Review URL: https://codereview.webrtc.org/1355083002

Cr-Commit-Position: refs/heads/master@{#10101}
2015-09-29 04:16:53 +00:00
petermayo
50cf10d82a Make .gni less sensitive to type of arm_use_neon flags
Comparing with 1 is less clear than using the input flags as
booleans.

BUG=5008276

Review URL: https://codereview.webrtc.org/1231663002

Cr-Commit-Position: refs/heads/master@{#9561}
2015-07-09 16:45:09 +00:00
Zeke Chin
71f6f4405c iOS HW H264 support.
First step towards supporting H264 on iOS. More tuning/experimentation
required in future CLs. Tested using AppRTCDemo on iPhone6 + iPad Mini.
Future work to get it working on OS/X, simulator (renders black screen
currently) and with the Android AppRTCDemo. Currently protected with a
compile time guard.

BUG=4081
R=andrew@webrtc.org, haysc@webrtc.org, holmer@google.com, jiayl@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1187573004.

Cr-Commit-Position: refs/heads/master@{#9515}
2015-06-29 21:35:08 +00:00
Andrew MacDonald
ac4234ccfc Add a [rtc_]build_with_neon variable to unify conditions.
Also consolidate ARM options for gn in an arm_neon_config.

R=jridges@masque.com, kjellander@webrtc.org, zhongwei.yao@chromium.org

Review URL: https://codereview.webrtc.org/1181373004.

Cr-Commit-Position: refs/heads/master@{#9501}
2015-06-25 01:25:59 +00:00
Henrik Kjellander
57e5fd2e60 PRESUBMIT: Improve PyLint check and add GN format check.
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).

Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.

Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py

TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.

R=henrika@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50069004

Cr-Commit-Position: refs/heads/master@{#9274}
2015-05-25 10:55:50 +00:00
Henrik Kjellander
e6cefb60f8 GYP variables for building expat, icu, libsrtp, usrsctp
This makes the build more flexible when linking against
prebuilt external libraries.

Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).

Also make it possible to avoid building the GTK parts of the Linux build.

BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44179005

Cr-Commit-Position: refs/heads/master@{#9087}
2015-04-27 12:38:37 +00:00
Andrew MacDonald
23dc68e515 Add the rtc_build_openmax_dl variable to the GN build.
For symmetry with the gyp build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49109005

Cr-Commit-Position: refs/heads/master@{#9083}
2015-04-24 15:46:31 +00:00
minyue@webrtc.org
7c112f3e5a Adding build_opus as a switch in GYP.
This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds.

BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43739004

Cr-Commit-Position: refs/heads/master@{#8754}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:05:18 +00:00
kjellander@webrtc.org
6dab6d700d Let Chromium declare the mips_dsp_rev build variable.
In https://codereview.chromium.org/883253003, the mips_dsp_rev
build variable is added to Chromium's GYP and GN build files. Remove
the declarations of mips_dsp_rev from WebRTC's GYP and GN build files.

Replace mips_fpu with mips_float_abi and remove the compiler flags that
are already set by Chromium.

The main review of this was done in https://webrtc-codereview.appspot.com/39779004
but since that CL wasn't created with the right base URL, I made
this in order to be able to run WebRTC trybots properly.

BUG=446234
TBR=wtc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/44549004

Cr-Commit-Position: refs/heads/master@{#8590}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8590 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 09:51:17 +00:00
andresp@webrtc.org
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
pbos@webrtc.org
4b3618c7f3 Remove TraceImpl logging thread.
Simplifies TraceImpl significantly. Chromium uses trace callbacks and
logs directly through the trace callback interface. Added slowdowns when
logging to file are expected to only affect test targets.

BUG=
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34329004

Cr-Commit-Position: refs/heads/master@{#8529}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8529 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:05:50 +00:00
kjellander@webrtc.org
722739108a Roll chromium_revision b0c3ed3..2c3ffb2 (316737:317530)
Includes GN changes from
https://webrtc-codereview.appspot.com/39249004/

Android changes for JNI were required due to
https://codereview.chromium.org/843103003

Other relevant changes:
* src/buildtools: 5c5e924..93b3d0a
* src/third_party/boringssl/src: d306f16..b180ee9
* src/third_party/icu: 4e3266f..2081ee6
* src/third_party/libvpx: 5cdd302..33bbffe
* src/third_party/usrsctp/usrsctplib: 190c8cb..13718c7
* src/tools/gyp: 4d7c139..3464008
* src/tools/swarming_client: bdad118..1b7bfec
Details: b0c3ed3..2c3ffb2/DEPS

Clang version was not updated in this roll.

R=dpranke@chromium.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40079004

Cr-Commit-Position: refs/heads/master@{#8466}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8466 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-23 19:09:22 +00:00
kjellander@webrtc.org
ce22f13f0e GN: Changes for vp9, opus and direct trace
Corresponding GN changes for
https://webrtc-codereview.appspot.com/34099004/

BUG=4185
R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/40669004

Cr-Commit-Position: refs/heads/master@{#8377}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8377 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 12:47:45 +00:00
kjellander@webrtc.org
acb8085678 Roll chromium_revision c086b4e..4664fe0 (312108:312733)
Mainly to pick up the MIPS changes in
https://codereview.chromium.org/843563002/
for which the changes in
https://webrtc-codereview.appspot.com/41399004/
are included in this CL.

Relevant changes:
* src/third_party/android_tools: 56b3d3e..aaeda3d
* src/third_party/boringssl/src: ca9a538..5fa3eba
* src/third_party/libvpx: 4f9bd1b..8dc6ea9
* src/third_party/openmax_dl: 1a4171c..8f7bf0b
* src/tools/gyp: 194ec65..adb7d24
* src/tools/swarming_client: 0a795bd..e98dde9
Details: c086b4e..4664fe0/DEPS

Clang version was not updated in this roll.

BUG=4214, 4222
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8145 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-25 19:17:56 +00:00
henrika@webrtc.org
45db7eefa2 Use Java based audio as default for WebRTC.
The work landed in 4034 (use of HW AEC in AppRTC) is currently not
active by default since we build for Open SL. I missed that when I
did my initial change (since I always disabled OpenSL by GYP_DEFINES).

This CL ensures that Java based audio is used as default in WebRTC.
It would be great if we could shift over to Open SL (to cut latency)
but that would (today) mean that we can't support the HW AEC.
Hence, we are not ready to do so yet.

BUG=4034
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8040 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 14:27:23 +00:00
andrew@webrtc.org
34ac956706 Do not use openmax_dl for MIPS64 platform.
This fix is intended for MIPS64 Chromium Android builds, which has no openmax_dl
support at this moment.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31339004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7983 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:19:56 +00:00
andrew@webrtc.org
1090a6eccf Remove obsolete target_arch == armv7.
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 21:36:18 +00:00
andrew@webrtc.org
0001adcfef Use openmax_dl on all ARM (v7 or higher) platforms.
openmax_dl now works on non-Android ARM, but it still requires
arm_version >= 7, and doesn't work on iOS at all.

TEST=Chromium build for a ChromeOS ARM device passes.
BUG=chromium:415393
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 04:13:02 +00:00
andrew@webrtc.org
4165f7aa22 Add a variable for deciding when to use openmax_dl.
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 18:01:27 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 08:58:15 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture)
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:50:19 +00:00
kjellander@webrtc.org
6d08ca6379 GN: Prefix WebRTC specific variables with "rtc_"
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/27379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
e281f7fba3 GN: Update webrtc/base to recent GYP changes.
Update the webrtc/base/BUILD.gn file to reflect
webrtc/base/base.gyp changes between r6438 and r7011.

BUG=3441
TESTED= Trybots + compilation with a standalone WebRTC checkout:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default

Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""

R=brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7022 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 11:22:06 +00:00
kjellander@webrtc.org
524b8f7304 GN: Implement voice engine, common audio, audio coding and audio processing
NOTICE: Assembly offsets generation for audio processing will
not be ported to GN and the process of removing them is tracked
in https://code.google.com/p/webrtc/issues/detail?id=3580.

The GN files are based upon the GYP files as of r7009.

BUG=3441
TESTED=Passing builds with:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false aec_debug_dump=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false aec_untrusted_delay_for_testing=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false prefer_fixed_point=true" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" is_clang=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7 is_clang=false" && ninja -C out/Default

I don't know how to setup the mips toolchain to test the following, but it's out of scope for the GN effort for now:
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=0" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=1" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false cpu_arch=\"mipsel\" mips_dsp_rev=2" && ninja -C out/Default

Compilation of Chromium's 'all' target with src/third_party/webrtc
symlinked to the WebRTC checkout with this CL applied, both
with the default GN settings and using
--args="is_debug=false os=\"android\" cpu_arch=\"arm\""

R=andrew@webrtc.org, brettw@chromium.org

Review URL: https://webrtc-codereview.appspot.com/15999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7012 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-31 20:32:53 +00:00
kjellander@webrtc.org
788f0581c7 GN: Implement video_engine, video_capture and video_render.
Also add more from common.gypi to webrtc.gni.

These GN configs are based on GYP files in r6997.

BUG=3441
TEST=Trybots and local compile using:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_debug=true" && ninja -C out/Default

Passed compile from a Chromium checkout with src/third_party/webrtc linked to the webrtc/ dir of a checkout with this patch applied.

R=brettw@chromium.org, glaznev@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6999 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 13:51:08 +00:00
stefan@webrtc.org
73823cafa4 Add initial gn build files for video_coding and video_processing.
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6611 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 11:46:43 +00:00
kjellander@webrtc.org
11bea8977e GN: Implement BUILD.gn for common_video.
This adds copying of Chromium's third_party/BUILD.gn
to acommondate libyuv's BUILD.gn that imports the 'jpeg'
config from that file.

BUG=3441
TEST=trybots + local compile passing with:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false build_libyuv=false" && ninja -C out/Default

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6595 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 17:04:12 +00:00
kjellander@webrtc.org
78f440c5e7 GN: BUILD.gn for system_wrappers
Also cleaned up some unneeded stuff from webrtc/base/BUILD.gn

BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default

I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6515 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-21 14:25:16 +00:00
kjellander@webrtc.org
851a09e71a Initial GN work for WebRTC
This CL makes it possible to build the 'webrtc_base'
target using GN.
The majority of our GYP stuff in webrtc/build/common.gypi has been
translated into the configs of webrtc/BUILD.gn.
The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn.

This CL depends on https://codereview.chromium.org/322373002/ for the
jsoncpp BUILD.gn file and the ssl config.
To build inside Chromium, https://codereview.chromium.org/321313006/
needs to be landed first.

BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default
I also ran:
gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true"
but it fails to compile: something is probably wrong with with pkg-config for that.

For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the
WebRTC checkout and applied the following patches:
https://codereview.chromium.org/322373002 (for jsoncpp and ssl config)
https://codereview.chromium.org/321313006 (enable building WebRTC)
Then I built successfully using:
gn gen out/Default && ninja -C out/Default webrtc_base

R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:54:03 +00:00