Age | Commit message (Collapse) | Author |
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Change-Id: I61f4e0917954d1557fe4cd1605aecaded68d4a97
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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
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are broken
This reverts commit 62f88b75e4f2cf47433a0a8948d679c0334f2204
Change-Id: I3a2c531c28d47867eb189d45c04632b54e9acca6
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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
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C pitch filter implementation was missing.
Change-Id: I7ab37627c1ddb79bca326c677ecc5f2cb5617647
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- Added fixed-point iSAC audio codec implementation (from stable r2699).
- Updated system wrappers (from stable r2699).
- Some cosmetic makefile changes.
Change-Id: If75d503698c11a4e4ceb851529127aadfe52f255
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- 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
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Change-Id: Ic341e9fb830cb364b9a3f51a8f804c6d6d1b6b3e
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Merged optimized sqrt() function from webRTC svn rev r1627
to improve performance ARM.
Change-Id: Ie9c57e2a3b8a0786c0169028c0940184b7c9db5e
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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
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Change-Id: Ib76086577a8b1a6567d91751e137f550ca64a288
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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
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This reverts commit 4e51691e58d8d32590b03c1951cb13de4d1c4758
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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
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