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authorwm4 <wm4@nowhere>2012-11-05 17:02:04 +0100
committerwm4 <wm4@nowhere>2012-11-12 20:06:14 +0100
commitd4bdd0473d6f43132257c9fb3848d829755167a3 (patch)
tree8021c2f7da1841393c8c832105e20cd527826d6c /libaf/af_surround.c
parentbd48deba77bd5582c5829d6fe73a7d2571088aba (diff)
downloadmpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.bz2
mpv-d4bdd0473d6f43132257c9fb3848d829755167a3.tar.xz
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.
Diffstat (limited to 'libaf/af_surround.c')
-rw-r--r--libaf/af_surround.c273
1 files changed, 0 insertions, 273 deletions
diff --git a/libaf/af_surround.c b/libaf/af_surround.c
deleted file mode 100644
index 57288d6ba2..0000000000
--- a/libaf/af_surround.c
+++ /dev/null
@@ -1,273 +0,0 @@
-/*
- * Filter to do simple decoding of matrixed surround sound.
- * This will provide a (basic) surround-sound effect from
- * audio encoded for Dolby Surround, Pro Logic etc.
- *
- * original author: Steve Davies <steve@daviesfam.org>
- *
- * 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.
- */
-
-/* The principle: Make rear channels by extracting anti-phase data
- from the front channels, delay by 20ms and feed to rear in anti-phase
-*/
-
-
-/* SPLITREAR: Define to decode two distinct rear channels - this
- doesn't work so well in practice because separation in a passive
- matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
- dialogue leaks to the rear. Still - give it a try and send
- feedback. Comment this define for old behavior of a single
- surround sent to rear in anti-phase */
-#define SPLITREAR 1
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "af.h"
-#include "dsp.h"
-
-#define L 32 // Length of fir filter
-#define LD 65536 // Length of delay buffer
-
-// 32 Tap fir filter loop unrolled
-#define FIR(x,w,y) \
- y = ( 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] \
- + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
- + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
- + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
- + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
-
-// Add to circular queue macro + update index
-#ifdef SPLITREAR
-#define ADDQUE(qi,rq,lq,r,l)\
- lq[qi]=lq[qi+L]=(l);\
- rq[qi]=rq[qi+L]=(r);\
- qi=(qi-1)&(L-1);
-#else
-#define ADDQUE(qi,lq,l)\
- lq[qi]=lq[qi+L]=(l);\
- qi=(qi-1)&(L-1);
-#endif
-
-// Macro for updating queue index in delay queues
-#define UPDATEQI(qi) qi=(qi+1)&(LD-1)
-
-// instance data
-typedef struct af_surround_s
-{
- float lq[2*L]; // Circular queue for filtering left rear channel
- float rq[2*L]; // Circular queue for filtering right rear channel
- float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
- float* dr; // Delay queue right rear channel
- float* dl; // Delay queue left rear channel
- float d; // Delay time
- int i; // Position in circular buffer
- int wi; // Write index for delay queue
- int ri; // Read index for delay queue
-}af_surround_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_surround_t *s = af->setup;
- switch(cmd){
- case AF_CONTROL_REINIT:{
- float fc;
- af->data->rate = ((struct mp_audio*)arg)->rate;
- af->data->nch = ((struct mp_audio*)arg)->nch*2;
- af->data->format = AF_FORMAT_FLOAT_NE;
- af->data->bps = 4;
-
- if (af->data->nch != 4){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n");
- return AF_DETACH;
- }
- // Surround filer coefficients
- fc = 2.0 * 7000.0/(float)af->data->rate;
- if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Unable to design low-pass filter.\n");
- return AF_ERROR;
- }
-
- // Free previous delay queues
- free(s->dl);
- free(s->dr);
- // Allocate new delay queues
- s->dl = calloc(LD,af->data->bps);
- s->dr = calloc(LD,af->data->bps);
- if((NULL == s->dl) || (NULL == s->dr))
- mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n");
-
- // Initialize delay queue index
- if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
- return AF_ERROR;
-// printf("%i\n",s->wi);
- s->ri = 0;
-
- if((af->data->format != ((struct mp_audio*)arg)->format) ||
- (af->data->bps != ((struct mp_audio*)arg)->bps)){
- ((struct mp_audio*)arg)->format = af->data->format;
- ((struct mp_audio*)arg)->bps = af->data->bps;
- return AF_FALSE;
- }
- return AF_OK;
- }
- case AF_CONTROL_COMMAND_LINE:{
- float d = 0;
- sscanf((char*)arg,"%f",&d);
- if ((d < 0) || (d > 1000)){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values"
- " are 0ms to 1000ms current value is %0.3f ms\n",d);
- return AF_ERROR;
- }
- s->d = d;
- return AF_OK;
- }
- }
- return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance* af)
-{
- if(af->data)
- free(af->data->audio);
- free(af->data);
- free(af->setup);
-}
-
-// The beginnings of an active matrix...
-static float steering_matrix[][12] = {
-// LL RL LR RR LS RS
-// LLs RLs LRs RRs LC RC
- {.707, .0, .0, .707, .5, -.5,
- .5878, -.3928, .3928, -.5878, .5, .5},
-};
-
-// Experimental moving average dominance
-//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
-
-// Filter data through filter
-static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){
- af_surround_t* s = (af_surround_t*)af->setup;
- float* m = steering_matrix[0];
- float* in = data->audio; // Input audio data
- float* out = NULL; // Output audio data
- float* end = in + data->len / sizeof(float); // Loop end
- int i = s->i; // Filter queue index
- int ri = s->ri; // Read index for delay queue
- int wi = s->wi; // Write index for delay queue
-
- if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
- return NULL;
-
- out = af->data->audio;
-
- while(in < end){
- /* Dominance:
- abs(in[0]) abs(in[1]);
- abs(in[0]+in[1]) abs(in[0]-in[1]);
- 10 * log( abs(in[0]) / (abs(in[1])|1) );
- 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
-
- /* About volume balancing...
- Surround encoding does the following:
- Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
- So S should be extracted as:
- (Lt-Rt)
- But we are splitting the S to two output channels, so we
- must take 3dB off as we split it:
- Ls=Rs=.707*(Lt-Rt)
- Trouble is, Lt could be +1, Rt -1, so possibility that S will
- overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
- 6dB (/2). This keeps the overall balance, but guarantees no
- overflow. */
-
- // Output front left and right
- out[0] = m[0]*in[0] + m[1]*in[1];
- out[1] = m[2]*in[0] + m[3]*in[1];
-
- // Low-pass output @ 7kHz
- FIR((&s->lq[i]), s->w, s->dl[wi]);
-
- // Delay output by d ms
- out[2] = s->dl[ri];
-
-#ifdef SPLITREAR
- // Low-pass output @ 7kHz
- FIR((&s->rq[i]), s->w, s->dr[wi]);
-
- // Delay output by d ms
- out[3] = s->dr[ri];
-#else
- out[3] = -out[2];
-#endif
-
- // Update delay queues indexes
- UPDATEQI(ri);
- UPDATEQI(wi);
-
- // Calculate and save surround in circular queue
-#ifdef SPLITREAR
- ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
-#else
- ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
-#endif
-
- // Next sample...
- in = &in[data->nch];
- out = &out[af->data->nch];
- }
-
- // Save indexes
- s->i = i; s->ri = ri; s->wi = wi;
-
- // Set output data
- data->audio = af->data->audio;
- data->len *= 2;
- data->nch = af->data->nch;
-
- return data;
-}
-
-static int af_open(struct af_instance* af){
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul=2;
- af->data=calloc(1,sizeof(struct mp_audio));
- af->setup=calloc(1,sizeof(af_surround_t));
- if(af->data == NULL || af->setup == NULL)
- return AF_ERROR;
- ((af_surround_t*)af->setup)->d = 20;
- return AF_OK;
-}
-
-struct af_info af_info_surround =
-{
- "Surround decoder filter",
- "surround",
- "Steve Davies <steve@daviesfam.org>",
- "",
- AF_FLAGS_NOT_REENTRANT,
- af_open
-};