2017-01-19 14:38:47 +01:00
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Thu Jan 12 08:32:04 UTC 2017 - olaf@aepfle.de
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- Add baselibs.conf for gstreamer-plugins-bad-32bit
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Accepting request 404777 from home:oholecek:branches:multimedia:libs
- Remove webrtc-aarch64.patch, no longer needed
- Adapt the rest of webrtc- patches to new arch naming
- Remove unneeded explicit version dependency for automake
- Update to 0.3
* build: enforce linking with --no-undefined, add explicit -lpthread
* build: Make sure files with SSE2 code are compiled with -msse2
- Remove no-undefined.patch
- Remove webrtc-audio-processing-0.2-x86_msse2.patch
- Add no-undefined.patch patch
https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
- Adapt big_endian_support.patch to new version
- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
- Add big_endian_support.patch
https://bugs.freedesktop.org/show_bug.cgi?id=95738
- New automake version dependency >= 1.5
- Update to 0.2:
Contains API breaking changes.
Upstream changes include:
* Rewritten AGC and voice activity detection
* Intelligibility enhancer
* Extended AEC filter
* Beamformer
* Transient suppressor
* ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
API changes:
* We no longer include a top-level audio_processing.h. The webrtc tree format
is used, so use webrtc/modules/audio_processing/include/audio_processing.h
* The top-level module_common_types.h has also been moved to
webrtc/modules/interface/module_common_types.h
* C++11 support is now required while compiling client code
* AudioProcessing::Create() does not take any arguments any more
* AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
* Stream parameters are now configured via StreamConfig and ProcessingConfig
rather than set_sample_rate(), set_num_channels(), etc.
* AudioFrame field names have changed
* Use config API for newer audio processing options
* Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
when using the intelligibility enhancer
* GainControl::set_analog_level_limits() is broken. The AGC implementation
hard codes 0-255 as the volume range
Other notes:
* The new audio processing parameters are not all tested, and a few are not
enabled upstream (in Chromium) either
* The rewritten AGC appears to be less sensitive, and it might make sense to
initialise the capture volume to something reasonable (33% or 50%, for
example) to make sure there is sufficient energy in the stream to trigger
the AGC mechanism
- Adapted all 3 arch patches
OBS-URL: https://build.opensuse.org/request/show/404777
OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/webrtc-audio-processing?expand=0&rev=11
2016-06-25 18:50:49 +02:00
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-------------------------------------------------------------------
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Sat Jun 25 10:39:08 UTC 2016 - oholecek@suse.com
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- Remove webrtc-aarch64.patch, no longer needed
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- Adapt the rest of webrtc- patches to new arch naming
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-------------------------------------------------------------------
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Thu Jun 23 13:31:14 UTC 2016 - oholecek@suse.com
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- Remove unneeded explicit version dependency for automake
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-------------------------------------------------------------------
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Wed Jun 22 11:55:11 UTC 2016 - oholecek@suse.com
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- Update to 0.3
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* build: enforce linking with --no-undefined, add explicit -lpthread
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* build: Make sure files with SSE2 code are compiled with -msse2
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- Remove no-undefined.patch
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- Remove webrtc-audio-processing-0.2-x86_msse2.patch
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-------------------------------------------------------------------
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Mon Jun 20 13:02:06 UTC 2016 - oholecek@suse.com
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- Add no-undefined.patch patch
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https://cgit.freedesktop.org/pulseaudio/webrtc-audio-processing/patch/?id=d58164e4d87854233564b59e76259b72e21507f6
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- Add big_endian_support_2.patch https://bugs.freedesktop.org/show_bug.cgi?id=95738
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- Adapt webrtc-audio-processing-0.2-x86_msse2.patch to new version
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- Adapt big_endian_support.patch to new version
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-------------------------------------------------------------------
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Mon May 30 09:00:51 UTC 2016 - oholecek@suse.com
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- Add webrtc-audio-processing-0.2-x86_msse2.patch patch fixing 386 build
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https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-May/026294.html
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- Add big_endian_support.patch
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https://bugs.freedesktop.org/show_bug.cgi?id=95738
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- New automake version dependency >= 1.5
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-------------------------------------------------------------------
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Thu May 26 21:19:28 UTC 2016 - oholecek@suse.com
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- Update to 0.2:
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Contains API breaking changes.
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Upstream changes include:
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* Rewritten AGC and voice activity detection
|
|
|
|
|
* Intelligibility enhancer
|
|
|
|
|
* Extended AEC filter
|
|
|
|
|
* Beamformer
|
|
|
|
|
* Transient suppressor
|
|
|
|
|
* ARM, NEON and MIPS optimisations (MIPS optimisations are not hooked up)
|
|
|
|
|
|
|
|
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API changes:
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* We no longer include a top-level audio_processing.h. The webrtc tree format
|
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|
is used, so use webrtc/modules/audio_processing/include/audio_processing.h
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* The top-level module_common_types.h has also been moved to
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webrtc/modules/interface/module_common_types.h
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* C++11 support is now required while compiling client code
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* AudioProcessing::Create() does not take any arguments any more
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* AudioProcessing::Destroy() is gone, use standard C++ "delete" instead
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* Stream parameters are now configured via StreamConfig and ProcessingConfig
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rather than set_sample_rate(), set_num_channels(), etc.
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* AudioFrame field names have changed
|
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|
|
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* Use config API for newer audio processing options
|
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|
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* Use ProcessReverseStream() instead of AnalyzeReverseStream(), particularly
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|
when using the intelligibility enhancer
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* GainControl::set_analog_level_limits() is broken. The AGC implementation
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hard codes 0-255 as the volume range
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|
|
|
|
|
|
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Other notes:
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* The new audio processing parameters are not all tested, and a few are not
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enabled upstream (in Chromium) either
|
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|
|
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* The rewritten AGC appears to be less sensitive, and it might make sense to
|
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|
|
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initialise the capture volume to something reasonable (33% or 50%, for
|
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|
|
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example) to make sure there is sufficient energy in the stream to trigger
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the AGC mechanism
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- Adapted all 3 arch patches
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2013-03-07 14:52:11 +01:00
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-------------------------------------------------------------------
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Thu Mar 7 13:51:31 UTC 2013 - idonmez@suse.com
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- Add patch webrtc-aarch64.patch from algraf to add aarch64 support
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2012-12-31 16:46:00 +01:00
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-------------------------------------------------------------------
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Wed Dec 19 10:39:23 CET 2012 - ro@suse.de
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- add s390 and s390x to known platforms
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by adding webrtc-s390x.patch
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2012-07-12 13:30:28 +02:00
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-------------------------------------------------------------------
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Tue Jul 3 15:00:06 UTC 2012 - dvaleev@suse.com
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- add ppc64 to known platforms
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2012-05-15 12:42:12 +02:00
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-------------------------------------------------------------------
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Tue May 15 10:40:38 CET 2012 - pascal.bleser@opensuse.org
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- initial version (0.1)
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