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
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
- Imported the floating point implementation of iSAC coder from stable public webRTC release:
http://code.google.com/p/webrtc/source/browse/#svn%2Fstable
- Added a reminder (in a TODO file) to switch from floating point to fixed point implementation.
Please note that this is the minimal amount of functionality we (the speech text-to-speech team) require
for delivering our unit selection algorithms for the Glass project. I'd be happy to address any comments
you have.
Change-Id: I20481680786ec5372258efcea69ea5c9750a321a
Updated audio processing modules from revision 180 to 1349.
Main changes are:
- code clean up and reformating
- source path reorganization
- improved performance
Also imported test code that was not included in initial
drop from webrtc.
Change-Id: Ie4eb0e29990052e5f2d7f0b271b42eead40dbb6a
Only the modules necessary for audio processing have been imported:
src/common_audio/
src/modules/audio_processing/
src/modules/interface/
src/system_wrappers/
src/typedefs.h
src/common_types.h
Android.mk
android-webrtc.mk
Android.mk and android-webrtc.mk have been modified to build only the
audio processing modules.
Files for Windows compatibility have been removed from system_wrappers.
fft_ARM9E directory has been removed from
src/common_audio/signal_processing_library/main/source/
Fixed x86 build.
SVN checkout at working revision 180.
Change-Id: If650f61d96557be8247b17eb4f4d32b7a6ba025d
Only the modules necessary for audio processing have been imported:
src/common_audio/
src/modules/audio_processing/
src/modules/interface/
src/system_wrappers/
src/typedefs.h
src/common_types.h
Android.mk
android-webrtc.mk
Android.mk and android-webrtc.mk have been modified to build only the
audio processing modules.
Files for Windows compatibility have been removed from system_wrappers.
fft_ARM9E directory has been removed from
src/common_audio/signal_processing_library/main/source/
SVN checkout at working revision 180.
Change-Id: I952373750f2c500d37f99aab4557aa84597d11b8
