Commit Graph

4 Commits

Author SHA1 Message Date
ba6371ec86 Delete unused video capture support for cropping, non-square pixels, and ARGB screencast scaling.
First two are unused, because the instance variables ratio_w_,
ratio_h_, and square_pixel_aspect_ratio_, are never modified after
initialization to 0 and false.

ARGB is believed to be unused, and the scaling logic
is probably not appropriate in any case.

Also delete corresponding helper functions in
videocommon.cc.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1934503002
Cr-Commit-Position: refs/heads/master@{#12659}
2016-05-09 07:47:59 +00:00
a2644c06ee Disable tests failing under UBSan to enable deployment to main waterfall.
modules_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/modules_unittests/logs/stdio
[ RUN      ] ByteIoTest.Test64SBitBigEndian
../../webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc:34:33: runtime error: shift exponent 64 is too large for 64-bit type 'long'

rtc_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_unittests/logs/stdio
[ RUN      ] IPAddressTest.TestCountIPMaskBits
../../webrtc/base/ipaddress.cc:415:20: runtime error: negation of -2147483648 cannot be represented in type 'int32_t' (aka 'int'); cast to an unsigned type to negate this value to itself

[ RUN      ] BandwidthSmootherTest.TestSampleRollover
../../webrtc/base/rollingaccumulator.h:73:22: runtime error: signed integer overflow: 2147483647 * 2147483647 cannot be represented in type 'int'

[ RUN      ] RandomNumberGeneratorTest.UniformSignedInterval
../../webrtc/base/random_unittest.cc:121:50: runtime error: signed integer overflow: 2147483647 - -2147483648 cannot be represented in type 'int'

rtc_media_unittests: https://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20UBSan/builds/1138/steps/rtc_media_unittests/logs/stdio
[ RUN      ] VideoCommonTest.TestComputeScaleWithHighFps
../../webrtc/media/base/videocommon.cc:75:34: runtime error: signed integer overflow: 2621440 - -2147483648 cannot be represented in type 'int'

BUG=webrtc:5487, webrtc:5490, webrtc:5491
NOTRY=True
R=pbos@webrtc.org
TBR=henrika@webrtc.org

Review URL: https://codereview.webrtc.org/1727233005 .

Cr-Commit-Position: refs/heads/master@{#11764}
2016-02-25 13:23:29 +00:00
1afca73055 Change to WebRTC license in webrtc/media
This was decided to be done in a separate CL from the move
that took place in https://codereview.webrtc.org/1587193006/

BUG=webrtc:5420
NOTRY=True
TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1676923002

Cr-Commit-Position: refs/heads/master@{#11520}
2016-02-08 04:46:50 +00:00
a96e2d77cb Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.

The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.

The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL in order to not
break Git history.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
  webrtc/base/testutils.cc
  webrtc/base/testutils.h

The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.

I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/

BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1587193006

Cr-Commit-Position: refs/heads/master@{#11495}
2016-02-05 07:52:35 +00:00