Mercurial > mplayer.hg
view libaf/af_volume.c @ 9021:070b91093606
* Remove unneeded includes & macros
* Get layers by name rather that fixed ID. Only works for DirectFB >=
0.9.16 so fall back to ID based stuff for older releases.
* New option "noinput" (useful in X)
* Add some NULL pointer checks to uninit()
* Improve error messages and use mp_msg()
* Direct rendering fix (at least IPB was broken)
* Use 0xff for alpha component in all colors
* Blit from temp surface to screen was only done in draw_osd() which
doesn't always get called. Do the blit in flip_page() in this case.
* Add GUI support to control()
* Change the driver name to include G450/G550 ;)
patch by Ville Syrj¸«£l¸«£ <syrjala@sci.fi>
author | arpi |
---|---|
date | Sun, 19 Jan 2003 15:41:26 +0000 |
parents | 398e3fb7c103 |
children | 1d75a7ecf3b8 |
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/*============================================================================= // // This software has been released under the terms of the GNU Public // license. See http://www.gnu.org/copyleft/gpl.html for details. // // Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au // //============================================================================= */ /* 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 6 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 <unistd.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_s* 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 = ((af_data_t*)arg)->rate; af->data->nch = ((af_data_t*)arg)->nch; if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_F | AF_FORMAT_NE))){ af->data->format = AF_FORMAT_SI | AF_FORMAT_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)); af_msg(AF_MSG_DEBUG0,"[volume] Forgetting factor = %0.5f\n",s->time); af->data->format = AF_FORMAT_F | AF_FORMAT_NE; af->data->bps = 4; } return af_test_output(af,(af_data_t*)arg); case AF_CONTROL_COMMAND_LINE:{ float v=-10.0; 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 = ((((af_cfg_t*)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_msg(AF_MSG_INFO,"[volume] The maximum volume was %0.2fdB \n", 10*log10(m)); } return AF_OK; } } return AF_UNKNOWN; } // Deallocate memory static void uninit(struct af_instance_s* af) { if(af->data) free(af->data); if(af->setup) free(af->setup); } // Filter data through filter static af_data_t* play(struct af_instance_s* af, af_data_t* data) { af_data_t* c = data; // Current working data af_volume_t* s = (af_volume_t*)af->setup; // Setup for this instance int ch = 0; // Channel counter 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_SI | AF_FORMAT_NE)){ int16_t* a = (int16_t*)c->audio; // Audio data int len = c->len/2; // Number of samples for(ch = 0; ch < nch ; ch++){ if(s->enable[ch]){ register int vol = (int)(255.0 * s->level[ch]); 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_F | AF_FORMAT_NE)){ float* a = (float*)c->audio; // Audio data int len = c->len/4; // Number of samples for(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){ if (x >= M_PI/2) x = 1.0; else if(x <= -M_PI/2) x = -1.0; else x = sin(x); } // Hard clipping else x=clamp(x,-1.0,1.0); a[i] = x; } } } } return c; } // Allocate memory and set function pointers static int open(af_instance_t* af){ int i = 0; af->control=control; af->uninit=uninit; af->play=play; af->mul.n=1; af->mul.d=1; af->data=calloc(1,sizeof(af_data_t)); 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 0.1 this is a safety measure to ensure that the user doesn't blow his speakers. If the user isn't happy with this he can use the command-line parameters to set the initial volume */ for(i=0;i<AF_NCH;i++){ ((af_volume_t*)af->setup)->enable[i] = 1; ((af_volume_t*)af->setup)->level[i] = 0.1; } return AF_OK; } // Description of this filter af_info_t af_info_volume = { "Volume control audio filter", "volume", "Anders", "", AF_FLAGS_NOT_REENTRANT, open };