Mercurial > mplayer.hg
view libaf/af_resample_template.c @ 28992:947ef23ba798
Test if create_vdp_decoder() might succeed by calling it from config()
with a small value for max_reference_frames.
This does not make automatic recovery by using software decoder possible,
but lets MPlayer fail more graciously on - actually existing - buggy
hardware that does not support certain H264 widths when using
hardware accelerated decoding (784, 864, 944, 1024, 1808, 1888 pixels on
NVIDIA G98) and if the user tries to hardware-decode more samples at
the same time than supported.
Might break playback of H264 Intra-Only samples on hardware with very
little video memory.
author | cehoyos |
---|---|
date | Sat, 21 Mar 2009 20:11:05 +0000 |
parents | 1e8262fa78c3 |
children | 0f1b5b68af32 |
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/* * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au * * This file is part of MPlayer. * * MPlayer is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * MPlayer is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License along * with MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ /* This file contains the resampling engine, the sample format is controlled by the FORMAT parameter, the filter length by the L parameter and the resampling type by UP and DN. This file should only be included by af_resample.c */ #undef L #undef SHIFT #undef FORMAT #undef FIR #undef ADDQUE /* The length Lxx definition selects the length of each poly phase component. Valid definitions are L8 and L16 where the number defines the nuber of taps. This definition affects the computational complexity, the performance and the memory usage. */ /* The FORMAT_x parameter selects the sample format type currently float and int16 are supported. Thes two formats are selected by defining eiter FORMAT_F or FORMAT_I. The advantage of using float is that the amplitude and therefore the SNR isn't affected by the filtering, the disadvantage is that it is a lot slower. */ #if defined(FORMAT_I) #define SHIFT >>16 #define FORMAT int16_t #else #define SHIFT #define FORMAT float #endif // Short filter #if defined(L8) #define L 8 // Filter length // Unrolled loop to speed up execution #define FIR(x,w,y) \ (y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \ + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT #else /* L8/L16 */ #define L 16 // Unrolled loop to speed up execution #define FIR(x,w,y) \ y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \ + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \ + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \ + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT #endif /* L8/L16 */ // Macro to add data to circular que #define ADDQUE(xi,xq,in)\ xq[xi]=xq[(xi)+L]=*(in);\ xi=((xi)-1)&(L-1); #if defined(UP) uint32_t ci = l->nch; // Index for channels uint32_t nch = l->nch; // Number of channels uint32_t inc = s->up/s->dn; uint32_t level = s->up%s->dn; uint32_t up = s->up; uint32_t dn = s->dn; uint32_t ns = c->len/l->bps; register FORMAT* w = s->w; register uint32_t wi = 0; register uint32_t xi = 0; // Index current channel while(ci--){ // Temporary pointers register FORMAT* x = s->xq[ci]; register FORMAT* in = ((FORMAT*)c->audio)+ci; register FORMAT* out = ((FORMAT*)l->audio)+ci; FORMAT* end = in+ns; // Block loop end wi = s->wi; xi = s->xi; while(in < end){ register uint32_t i = inc; if(wi<level) i++; ADDQUE(xi,x,in); in+=nch; while(i--){ // Run the FIR filter FIR((&x[xi]),(&w[wi*L]),out); len++; out+=nch; // Update wi to point at the correct polyphase component wi=(wi+dn)%up; } } } // Save values that needs to be kept for next time s->wi = wi; s->xi = xi; #endif /* UP */ #if defined(DN) /* DN */ uint32_t ci = l->nch; // Index for channels uint32_t nch = l->nch; // Number of channels uint32_t inc = s->dn/s->up; uint32_t level = s->dn%s->up; uint32_t up = s->up; uint32_t dn = s->dn; uint32_t ns = c->len/l->bps; FORMAT* w = s->w; register int32_t i = 0; register uint32_t wi = 0; register uint32_t xi = 0; // Index current channel while(ci--){ // Temporary pointers register FORMAT* x = s->xq[ci]; register FORMAT* in = ((FORMAT*)c->audio)+ci; register FORMAT* out = ((FORMAT*)l->audio)+ci; register FORMAT* end = in+ns; // Block loop end i = s->i; wi = s->wi; xi = s->xi; while(in < end){ ADDQUE(xi,x,in); in+=nch; if((--i)<=0){ // Run the FIR filter FIR((&x[xi]),(&w[wi*L]),out); len++; out+=nch; // Update wi to point at the correct polyphase component wi=(wi+dn)%up; // Insert i number of new samples in queue i = inc; if(wi<level) i++; } } } // Save values that needs to be kept for next time s->wi = wi; s->xi = xi; s->i = i; #endif /* DN */