56 lines
1.8 KiB
Plaintext
56 lines
1.8 KiB
Plaintext
Release 0.3
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-----------
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Minor build fixes.
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Release 0.2
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-----------
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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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-----------
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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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