webrtc-audio-processing-0.3-150000.3.2.1

- Add baselibs.conf for gstreamer-plugins-bad-32bit

- 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

- Add patch webrtc-aarch64.patch from algraf to add aarch64 support

- add s390 and s390x to known platforms
  by adding webrtc-s390x.patch

- add ppc64 to known platforms

- initial version (0.1)