99 lines
3.1 KiB
Plaintext
99 lines
3.1 KiB
Plaintext
Release 1.3
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Fix for generate pkg-config file.
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Release 1.2
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Improvements for building with abseil-cpp as a subproject, and pkg-config
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improvements for abseil dependency detection.
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Release 1.1
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Build fixes for various platforms.
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Release 1.0
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This is an API breaking release (as a reminder, the AudioProcessing module does
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not provide a stable public API, so we expose whatever API exists in the
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upstream project).
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In order to make package management easier with these inevitable breakages, the
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package is now suffixed with a version (currently it is
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webrtc-audio-processing-1). When the next API break happens, we will bump the
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major version, allowing incompatible versions to coexist. This also means that
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the previous version can also coexist with this one. Non-breaking changes will
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see a minor version update only.
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Changes:
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* The code base is now updated to correspond to the version shipping with the
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Chromium 88.0.4290.1 tag
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* There are a very large number changes to the underlying AEC implementation
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since the last update was a while ago. Most visibly the use of the AEC3
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canceller by default, the deletion of the beamformer code
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* The autotools build system is replaced by meson
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* The pkg-config name is changed as described above
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Release 0.3
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Minor build fixes.
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Release 0.2
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Updated AudioProcessing code to be more current.
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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
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* Intelligibility enhancer
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* Extended AEC filter
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* Beamformer
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* Transient suppressor
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* 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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* Use config API for newer audio processing options
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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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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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* The rewritten AGC appears to be less sensitive, and it might make sense to
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initialise the capture volume to something reasonable (33% or 50%, for
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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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Release 0.1
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Initial release, consisting of the WebRTC AudioProcessing module with a
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distributor-friendly build system.
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