mirror of
https://github.com/mpv-player/mpv
synced 2024-12-12 01:46:16 +00:00
6adaa78ee9
- Improved runtime control system
- 3 New filter panning, compressor/limiter and a noise gate
- The compressor/limiter and the noise gate are not yet finished
- The panning filter does combined mixing and channel routing and
can be used to down-mix from stereo to mono (for example)
- Improvements to volume and channel
- volume now has a very good soft clipping using sin()
- channel can handle generic routing of audio data
- Conversion of all filters to handle floating point data
- Cleanup of message printing
- Fix for the sig 11 bug reported by Denes
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8608 b3059339-0415-0410-9bf9-f77b7e298cf2
162 lines
4.2 KiB
C
162 lines
4.2 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* This file contains the resampling engine, the sample format is
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controlled by the FORMAT parameter, the filter length by the L
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parameter and the resampling type by UP and DN. This file should
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only be included by af_resample.c
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*/
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#undef L
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#undef SHIFT
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#undef FORMAT
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#undef FIR
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#undef ADDQUE
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/* The lenght Lxx definition selects the length of each poly phase
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component. Valid definitions are L8 and L16 where the number
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defines the nuber of taps. This definition affects the
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computational complexity, the performance and the memory usage.
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*/
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/* The FORMAT_x parameter selects the sample format type currently
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float and int16 are supported. Thes two formats are selected by
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defining eiter FORMAT_F or FORMAT_I. The advantage of using float
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is that the amplitude and therefore the SNR isn't affected by the
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filtering, the disadvantage is that it is a lot slower.
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*/
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#if defined(FORMAT_I)
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#define SHIFT >>16
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#define FORMAT int16_t
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#else
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#define SHIFT
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#define FORMAT float
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#endif
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// Short filter
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#if defined(L8)
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#define L 8 // Filter length
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// Unrolled loop to speed up execution
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#define FIR(x,w,y) \
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(y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
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+ w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT
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#else /* L8/L16 */
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#define L 16
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// Unrolled loop to speed up execution
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#define FIR(x,w,y) \
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y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
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+ w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
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+ w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
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+ w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT
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#endif /* L8/L16 */
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// Macro to add data to circular que
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#define ADDQUE(xi,xq,in)\
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xq[xi]=xq[xi+L]=(*in);\
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xi=(xi-1)&(L-1);
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#if defined(UP)
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uint32_t ci = l->nch; // Index for channels
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uint32_t nch = l->nch; // Number of channels
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uint32_t inc = s->up/s->dn;
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uint32_t level = s->up%s->dn;
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uint32_t up = s->up;
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uint32_t dn = s->dn;
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uint32_t ns = c->len/l->bps;
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register FORMAT* w = s->w;
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register uint32_t wi = 0;
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register uint32_t xi = 0;
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// Index current channel
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while(ci--){
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// Temporary pointers
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register FORMAT* x = s->xq[ci];
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register FORMAT* in = ((FORMAT*)c->audio)+ci;
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register FORMAT* out = ((FORMAT*)l->audio)+ci;
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FORMAT* end = in+ns; // Block loop end
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wi = s->wi; xi = s->xi;
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while(in < end){
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register uint32_t i = inc;
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if(wi<level) i++;
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ADDQUE(xi,x,in);
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in+=nch;
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while(i--){
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// Run the FIR filter
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FIR((&x[xi]),(&w[wi*L]),out);
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len++; out+=nch;
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// Update wi to point at the correct polyphase component
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wi=(wi+dn)%up;
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}
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}
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}
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// Save values that needs to be kept for next time
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s->wi = wi;
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s->xi = xi;
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#endif /* UP */
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#if defined(DN) /* DN */
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uint32_t ci = l->nch; // Index for channels
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uint32_t nch = l->nch; // Number of channels
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uint32_t inc = s->dn/s->up;
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uint32_t level = s->dn%s->up;
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uint32_t up = s->up;
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uint32_t dn = s->dn;
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uint32_t ns = c->len/l->bps;
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FORMAT* w = s->w;
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register int32_t i = 0;
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register uint32_t wi = 0;
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register uint32_t xi = 0;
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// Index current channel
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while(ci--){
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// Temporary pointers
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register FORMAT* x = s->xq[ci];
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register FORMAT* in = ((FORMAT*)c->audio)+ci;
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register FORMAT* out = ((FORMAT*)l->audio)+ci;
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register FORMAT* end = in+ns; // Block loop end
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i = s->i; wi = s->wi; xi = s->xi;
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while(in < end){
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ADDQUE(xi,x,in);
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in+=nch;
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if((--i)<=0){
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// Run the FIR filter
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FIR((&x[xi]),(&w[wi*L]),out);
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len++; out+=nch;
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// Update wi to point at the correct polyphase component
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wi=(wi+dn)%up;
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// Insert i number of new samples in queue
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i = inc;
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if(wi<level) i++;
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}
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}
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}
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// Save values that needs to be kept for next time
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s->wi = wi;
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s->xi = xi;
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s->i = i;
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#endif /* DN */
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