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| -rw-r--r-- | NEWS | 44 |
1 files changed, 44 insertions, 0 deletions
@@ -1,3 +1,47 @@ +Release 0.2 +----------- + +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 + + Release 0.1 ----------- |