Rename directories, move files (step 1 of 2) (does not compile)

Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.

1 files changed, 0 insertions, 248 deletions

diff --git a/libaf/af_equalizer.c b/libaf/af_equalizer.c
deleted file mode 100644
index c488ffaeaf..0000000000
--- a/libaf/af_equalizer.c
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Equalizer filter, implementation of a 10 band time domain graphic
- * equalizer using IIR filters. The IIR filters are implemented using a
- * Direct Form II approach, but has been modified (b1 == 0 always) to
- * save computation.
- *
- * Copyright (C) 2001 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.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <inttypes.h>
-#include <math.h>
-
-#include "af.h"
-
-#define L   	2      // Storage for filter taps
-#define KM  	10     // Max number of bands
-
-#define Q   1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
-			 gives 4dB suppression @ Fc*2 and Fc/2 */
-
-/* Center frequencies for band-pass filters
-   The different frequency bands are:
-   nr.    	center frequency
-   0  	31.25 Hz
-   1 	62.50 Hz
-   2	125.0 Hz
-   3	250.0 Hz
-   4	500.0 Hz
-   5	1.000 kHz
-   6	2.000 kHz
-   7	4.000 kHz
-   8	8.000 kHz
-   9 	16.00 kHz
-*/
-#define CF  	{31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
-
-// Maximum and minimum gain for the bands
-#define G_MAX	+12.0
-#define G_MIN	-12.0
-
-// Data for specific instances of this filter
-typedef struct af_equalizer_s
-{
-  float   a[KM][L];        	// A weights
-  float   b[KM][L];	     	// B weights
-  float   wq[AF_NCH][KM][L];  	// Circular buffer for W data
-  float   g[AF_NCH][KM];      	// Gain factor for each channel and band
-  int     K; 		   	// Number of used eq bands
-  int     channels;        	// Number of channels
-  float   gain_factor;     // applied at output to avoid clipping
-} af_equalizer_t;
-
-// 2nd order Band-pass Filter design
-static void bp2(float* a, float* b, float fc, float q){
-  double th= 2.0 * M_PI * fc;
-  double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
-
-  a[0] = (1.0 + C) * cos(th);
-  a[1] = -1 * C;
-
-  b[0] = (1.0 - C)/2.0;
-  b[1] = -1.0050;
-}
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_equalizer_t* s   = (af_equalizer_t*)af->setup;
-
-  switch(cmd){
-  case AF_CONTROL_REINIT:{
-    int k =0, i =0;
-    float F[KM] = CF;
-
-    s->gain_factor=0.0;
-
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
-
-    // Calculate number of active filters
-    s->K=KM;
-    while(F[s->K-1] > (float)af->data->rate/2.2)
-      s->K--;
-
-    if(s->K != KM)
-      mp_msg(MSGT_AFILTER, MSGL_INFO, "[equalizer] Limiting the number of filters to"
-	     " %i due to low sample rate.\n",s->K);
-
-    // Generate filter taps
-    for(k=0;k<s->K;k++)
-      bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
-
-    // Calculate how much this plugin adds to the overall time delay
-    af->delay = 2 * af->data->nch * af->data->bps;
-
-    // Calculate gain factor to prevent clipping at output
-    for(k=0;k<AF_NCH;k++)
-    {
-        for(i=0;i<KM;i++)
-        {
-            if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
-        }
-    }
-
-    s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
-
-    if(s->gain_factor > 0.0)
-    {
-        s->gain_factor=0.1+(s->gain_factor/12.0);
-    }else{
-        s->gain_factor=1;
-    }
-
-    return af_test_output(af,arg);
-  }
-  case AF_CONTROL_COMMAND_LINE:{
-    float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
-    int i,j;
-    sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
-	   &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
-    for(i=0;i<AF_NCH;i++){
-      for(j=0;j<KM;j++){
-	((af_equalizer_t*)af->setup)->g[i][j] =
-	  pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
-      }
-    }
-    return AF_OK;
-  }
-  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
-    float* gain = ((af_control_ext_t*)arg)->arg;
-    int    ch   = ((af_control_ext_t*)arg)->ch;
-    int    k;
-    if(ch >= AF_NCH || ch < 0)
-      return AF_ERROR;
-
-    for(k = 0 ; k<KM ; k++)
-      s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
-
-    return AF_OK;
-  }
-  case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
-    float* gain = ((af_control_ext_t*)arg)->arg;
-    int    ch   = ((af_control_ext_t*)arg)->ch;
-    int    k;
-    if(ch >= AF_NCH || ch < 0)
-      return AF_ERROR;
-
-    for(k = 0 ; k<KM ; k++)
-      gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
-
-    return AF_OK;
-  }
-  }
-  return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance* af)
-{
-    free(af->data);
-    free(af->setup);
-}
-
-// Filter data through filter
-static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
-{
-  struct mp_audio*       c 	= data;			    	// Current working data
-  af_equalizer_t*  s 	= (af_equalizer_t*)af->setup; 	// Setup
-  uint32_t  	   ci  	= af->data->nch; 	    	// Index for channels
-  uint32_t	   nch 	= af->data->nch;   	    	// Number of channels
-
-  while(ci--){
-    float*	g   = s->g[ci];      // Gain factor
-    float*	in  = ((float*)c->audio)+ci;
-    float*	out = ((float*)c->audio)+ci;
-    float* 	end = in + c->len/4; // Block loop end
-
-    while(in < end){
-      register int	k  = 0;		// Frequency band index
-      register float 	yt = *in; 	// Current input sample
-      in+=nch;
-
-      // Run the filters
-      for(;k<s->K;k++){
- 	// Pointer to circular buffer wq
- 	register float* wq = s->wq[ci][k];
- 	// Calculate output from AR part of current filter
- 	register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
- 	// Calculate output form MA part of current filter
- 	yt+=(w + wq[1]*s->b[k][1])*g[k];
- 	// Update circular buffer
- 	wq[1] = wq[0];
-	wq[0] = w;
-      }
-      // Calculate output
-      *out=yt*s->gain_factor;
-      out+=nch;
-    }
-  }
-  return c;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  af->control=control;
-  af->uninit=uninit;
-  af->play=play;
-  af->mul=1;
-  af->data=calloc(1,sizeof(struct mp_audio));
-  af->setup=calloc(1,sizeof(af_equalizer_t));
-  if(af->data == NULL || af->setup == NULL)
-    return AF_ERROR;
-  return AF_OK;
-}
-
-// Description of this filter
-struct af_info af_info_equalizer = {
-  "Equalizer audio filter",
-  "equalizer",
-  "Anders",
-  "",
-  AF_FLAGS_NOT_REENTRANT,
-  af_open
-};