Specifically:
external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/../test/kenny.c:366: error: undefined reference to 'atof'
The real problem here is that this code is compiling against the platform
bionic headers but linking against the NDK C library. That's a terrible
idea, but I can't understand the makefile well enough to stop that.
This fixes the build by using a function that's common to both C libraries.
Change-Id: I3d5747014a45269520a4c3585be42f2a78293d86
Moved webrts_isac_test and webrtc_isac_test_gnustl_static targets
to a separate makefile. Build error was caused by "../kenny.c" src
file - build system was building kenny.o outside of intermediates directory
(out/target/product/generic/obj/EXECUTABLES/webrtc_isac_test_gnustl_static_intermediates/../test/kenny.o).
This somehow, sometimes caused webrtc_isac_test_gnustl_static binary being build
BEFORE kenny.o file was built.
Change-Id: Iaa4a31327d8d381c6cc7f676583985259dd14d83
This allows webrtc to build with clang 3.4.
Change-Id: Idc043072db83834875a958b401ce747aa7e21403
Signed-off-by: Bernhard Rosenkränzer <Bernhard.Rosenkranzer@linaro.org>
Attempt number 2. Now with working tests.
This change allows to build fully unbundled GoogleTTS apk that can be deployed
on any >= ICS_MR1 device.
All static libraries under src/* can be build using ndk stl libraries, using
WEBRTC_STL varible. libwebrtc_audio_coding_gnustl_static is static version of
libwebrtc_audio_coding, build using gnustl from ndk.
Change-Id: I41a5163eb434432eab3131f5df23ffd311e6159b
This change allows to build fully unbundled GoogleTTS apk that can be deployed
on any >= ICS_MR1 device.
All static libraries under src/* can be build using ndk stl libraries, using
WEBRTC_STL varible. libwebrtc_audio_coding_gnustl_static is static version of
libwebrtc_audio_coding, build using gnustl from ndk.
Bug: 6397748
Change-Id: Ibf0acb11d3e605a1d4c668bbf98b0a0bb55399bc
Since Chromium has moved to this policy, we should too.
Code is copied from /depot_tools/presubmit_canned_checks.py but modified for our purpose.
BUG=
TEST=Tested git cl presubmit with a modified .cc file with the 2011 header and one with the 2012.
Review URL: https://webrtc-codereview.appspot.com/770005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2691 4adac7df-926f-26a2-2b94-8c16560cd09d
* Remove the peerconnection_server target from peerconnection.gyp since we have it in libjingle.gyp.
* Add enabled_libjingle_device_manager in supplement.gypi to add devicemanger to stand alone build.
* Add link settings to base.gyp which is needed by the new changes in peerconnection_client.
Note: Resolving hostname function has some problem on Windows in this revision.
So with this revision the peerconnection client can only take ip address directly as
the server address on Windows.
Review URL: https://webrtc-codereview.appspot.com/753008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2689 4adac7df-926f-26a2-2b94-8c16560cd09d
There are Chrome crashes which appear to be occurring during some kind
of teardown. We might be able to avoid them by locking in the destructor.
On the other hand, this might have no impact, but at least isn't a bad
thing to do.
BUG=chromium:145341
Review URL: https://webrtc-codereview.appspot.com/768005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2687 4adac7df-926f-26a2-2b94-8c16560cd09d
removing vie_codec from cl
Moving debug call from Codec to File impl.
Updating cl following review
Updating file name
Updating cl following review.
Updating CL following review.
Adding an API that allows recording of video data
updating cl
Adding debug options
BUG=
Review URL: https://webrtc-codereview.appspot.com/751006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2678 4adac7df-926f-26a2-2b94-8c16560cd09d
