Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
WebRtcSpl_ComplexBitReverse, WebRtcSpl_ComplexFFT, WebRtcSpl_ComplexIFFT, WebRtcSpl_DownsampleFast and WebRtcSpl_FilterARFastQ12.
Also, moved the common table used in complex_fft functions to a separate header file (webrtc/common_audio/signal_processing/include/complex_fft_tables.h).
R=andrew@webrtc.org, kma@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1126004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4141 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL solves the wraparound distortion in Opus.
In the Opus decoder-wrapper we are downsampling the signal from 48 kHz to 32 kHz. This is done in two steps, using the following functions from the signal processing library:
WebRtcSpl_Resample48khzTo32khz() and WebRtcSpl_VectorBitShiftW32ToW16
The latter does not have a saturation check, and the signal can suffer from wraparound. I've added saturation control to the function.
BUG=issue1089
Review URL: https://webrtc-codereview.appspot.com/967004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3103 4adac7df-926f-26a2-2b94-8c16560cd09d
