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-rw-r--r--audio/filter/af_volume.c226
1 files changed, 226 insertions, 0 deletions
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
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+++ b/audio/filter/af_volume.c
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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 audio filter changes the volume of the sound, and can be used
+ when the mixer doesn't support the PCM channel. It can handle
+ between 1 and AF_NCH channels. The volume can be adjusted between -60dB
+ to +20dB and is set on a per channels basis. The is accessed through
+ AF_CONTROL_VOLUME_LEVEL.
+
+ The filter has support for soft-clipping, it is enabled by
+ AF_CONTROL_VOLUME_SOFTCLIPP. It has also a probing feature which
+ can be used to measure the power in the audio stream, both an
+ instantaneous value and the maximum value can be probed. The
+ probing is enable by AF_CONTROL_VOLUME_PROBE_ON_OFF and is done on a
+ per channel basis. The result from the probing is obtained using
+ AF_CONTROL_VOLUME_PROBE_GET and AF_CONTROL_VOLUME_PROBE_GET_MAX. The
+ probed values are calculated in dB.
+*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <inttypes.h>
+#include <math.h>
+#include <limits.h>
+
+#include "af.h"
+
+// Data for specific instances of this filter
+typedef struct af_volume_s
+{
+ int enable[AF_NCH]; // Enable/disable / channel
+ float pow[AF_NCH]; // Estimated power level [dB]
+ float max[AF_NCH]; // Max Power level [dB]
+ float level[AF_NCH]; // Gain level for each channel
+ float time; // Forgetting factor for power estimate
+ int soft; // Enable/disable soft clipping
+ int fast; // Use fix-point volume control
+}af_volume_t;
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+ af_volume_t* s = (af_volume_t*)af->setup;
+
+ switch(cmd){
+ case AF_CONTROL_REINIT:
+ // Sanity check
+ if(!arg) return AF_ERROR;
+
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
+
+ if(s->fast && (((struct mp_audio*)arg)->format != (AF_FORMAT_FLOAT_NE))){
+ af->data->format = AF_FORMAT_S16_NE;
+ af->data->bps = 2;
+ }
+ else{
+ // Cutoff set to 10Hz for forgetting factor
+ float x = 2.0*M_PI*15.0/(float)af->data->rate;
+ float t = 2.0-cos(x);
+ s->time = 1.0 - (t - sqrt(t*t - 1));
+ mp_msg(MSGT_AFILTER, MSGL_DBG2, "[volume] Forgetting factor = %0.5f\n",s->time);
+ af->data->format = AF_FORMAT_FLOAT_NE;
+ af->data->bps = 4;
+ }
+ return af_test_output(af,(struct mp_audio*)arg);
+ case AF_CONTROL_COMMAND_LINE:{
+ float v=0.0;
+ float vol[AF_NCH];
+ int i;
+ sscanf((char*)arg,"%f:%i", &v, &s->soft);
+ for(i=0;i<AF_NCH;i++) vol[i]=v;
+ return control(af,AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET, vol);
+ }
+ case AF_CONTROL_POST_CREATE:
+ s->fast = ((((struct af_cfg*)arg)->force & AF_INIT_FORMAT_MASK) ==
+ AF_INIT_FLOAT) ? 0 : 1;
+ return AF_OK;
+ case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_SET:
+ memcpy(s->enable,(int*)arg,AF_NCH*sizeof(int));
+ return AF_OK;
+ case AF_CONTROL_VOLUME_ON_OFF | AF_CONTROL_GET:
+ memcpy((int*)arg,s->enable,AF_NCH*sizeof(int));
+ return AF_OK;
+ case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_SET:
+ s->soft = *(int*)arg;
+ return AF_OK;
+ case AF_CONTROL_VOLUME_SOFTCLIP | AF_CONTROL_GET:
+ *(int*)arg = s->soft;
+ return AF_OK;
+ case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_SET:
+ return af_from_dB(AF_NCH,(float*)arg,s->level,20.0,-200.0,60.0);
+ case AF_CONTROL_VOLUME_LEVEL | AF_CONTROL_GET:
+ return af_to_dB(AF_NCH,s->level,(float*)arg,20.0);
+ case AF_CONTROL_VOLUME_PROBE | AF_CONTROL_GET:
+ return af_to_dB(AF_NCH,s->pow,(float*)arg,10.0);
+ case AF_CONTROL_VOLUME_PROBE_MAX | AF_CONTROL_GET:
+ return af_to_dB(AF_NCH,s->max,(float*)arg,10.0);
+ case AF_CONTROL_PRE_DESTROY:{
+ float m = 0.0;
+ int i;
+ if(!s->fast){
+ for(i=0;i<AF_NCH;i++)
+ m=max(m,s->max[i]);
+ af_to_dB(1, &m, &m, 10.0);
+ mp_msg(MSGT_AFILTER, MSGL_INFO, "[volume] The maximum volume was %0.2fdB \n", m);
+ }
+ 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_volume_t* s = (af_volume_t*)af->setup; // Setup for this instance
+ register int nch = c->nch; // Number of channels
+ register int i = 0;
+
+ // Basic operation volume control only (used on slow machines)
+ if(af->data->format == (AF_FORMAT_S16_NE)){
+ int16_t* a = (int16_t*)c->audio; // Audio data
+ int len = c->len/2; // Number of samples
+ for (int ch = 0; ch < nch; ch++) {
+ int vol = 256.0 * s->level[ch];
+ if (s->enable[ch] && vol != 256) {
+ for(i=ch;i<len;i+=nch){
+ register int x = (a[i] * vol) >> 8;
+ a[i]=clamp(x,SHRT_MIN,SHRT_MAX);
+ }
+ }
+ }
+ }
+ // Machine is fast and data is floating point
+ else if(af->data->format == (AF_FORMAT_FLOAT_NE)){
+ float* a = (float*)c->audio; // Audio data
+ int len = c->len/4; // Number of samples
+ for (int ch = 0; ch < nch; ch++) {
+ // Volume control (fader)
+ if(s->enable[ch]){
+ float t = 1.0 - s->time;
+ for(i=ch;i<len;i+=nch){
+ register float x = a[i];
+ register float pow = x*x;
+ // Check maximum power value
+ if(pow > s->max[ch])
+ s->max[ch] = pow;
+ // Set volume
+ x *= s->level[ch];
+ // Peak meter
+ pow = x*x;
+ if(pow > s->pow[ch])
+ s->pow[ch] = pow;
+ else
+ s->pow[ch] = t*s->pow[ch] + pow*s->time; // LP filter
+ /* Soft clipping, the sound of a dream, thanks to Jon Wattes
+ post to Musicdsp.org */
+ if(s->soft)
+ x=af_softclip(x);
+ // Hard clipping
+ else
+ x=clamp(x,-1.0,1.0);
+ a[i] = x;
+ }
+ }
+ }
+ }
+ return c;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+ int i = 0;
+ 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_volume_t));
+ if(af->data == NULL || af->setup == NULL)
+ return AF_ERROR;
+ // Enable volume control and set initial volume to 0dB.
+ for(i=0;i<AF_NCH;i++){
+ ((af_volume_t*)af->setup)->enable[i] = 1;
+ ((af_volume_t*)af->setup)->level[i] = 1.0;
+ }
+ return AF_OK;
+}
+
+// Description of this filter
+struct af_info af_info_volume = {
+ "Volume control audio filter",
+ "volume",
+ "Anders",
+ "",
+ AF_FLAGS_NOT_REENTRANT,
+ af_open
+};
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