Package has now a different build system and different focus. It can statically build for a combination of different platforms.
The build system was changed to CMake. CMake presets are used heavily to build for the supported platforms.
Minor build fixes.
Updated AudioProcessing code to be more current.
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
Initial release, consisting of the WebRTC AudioProcessing module with a distributor-friendly build system.