diff options
author | wm4 <wm4@nowhere> | 2012-11-05 17:02:04 +0100 |
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committer | wm4 <wm4@nowhere> | 2012-11-12 20:06:14 +0100 |
commit | d4bdd0473d6f43132257c9fb3848d829755167a3 (patch) | |
tree | 8021c2f7da1841393c8c832105e20cd527826d6c /audio/filter/af_volnorm.c | |
parent | bd48deba77bd5582c5829d6fe73a7d2571088aba (diff) | |
download | mpv-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 'audio/filter/af_volnorm.c')
-rw-r--r-- | audio/filter/af_volnorm.c | 353 |
1 files changed, 353 insertions, 0 deletions
diff --git a/audio/filter/af_volnorm.c b/audio/filter/af_volnorm.c new file mode 100644 index 0000000000..b4c204d305 --- /dev/null +++ b/audio/filter/af_volnorm.c @@ -0,0 +1,353 @@ +/* + * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard + * + * 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 <string.h> + +#include <inttypes.h> +#include <math.h> +#include <limits.h> + +#include "af.h" + +// Methods: +// 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1) +// 2: uses several samples to smooth the variations (standard weighted mean +// on past samples) + +// Size of the memory array +// FIXME: should depend on the frequency of the data (should be a few seconds) +#define NSAMPLES 128 + +// If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we +// choose to ignore the computed value as it's not significant enough +// FIXME: should depend on the frequency of the data (0.5s maybe) +#define MIN_SAMPLE_SIZE 32000 + +// mul is the value by which the samples are scaled +// and has to be in [MUL_MIN, MUL_MAX] +#define MUL_INIT 1.0 +#define MUL_MIN 0.1 +#define MUL_MAX 5.0 + +// Silence level +// FIXME: should be relative to the level of the samples +#define SIL_S16 (SHRT_MAX * 0.01) +#define SIL_FLOAT (INT_MAX * 0.01) // FIXME + +// smooth must be in ]0.0, 1.0[ +#define SMOOTH_MUL 0.06 +#define SMOOTH_LASTAVG 0.06 + +#define DEFAULT_TARGET 0.25 + +// Data for specific instances of this filter +typedef struct af_volume_s +{ + int method; // method used + float mul; + // method 1 + float lastavg; // history value of the filter + // method 2 + int idx; + struct { + float avg; // average level of the sample + int len; // sample size (weight) + } mem[NSAMPLES]; + // "Ideal" level + float mid_s16; + float mid_float; +}af_volnorm_t; + +// Initialization and runtime control +static int control(struct af_instance* af, int cmd, void* arg) +{ + af_volnorm_t* s = (af_volnorm_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(((struct mp_audio*)arg)->format == (AF_FORMAT_S16_NE)){ + af->data->format = AF_FORMAT_S16_NE; + af->data->bps = 2; + }else{ + 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:{ + int i = 0; + float target = DEFAULT_TARGET; + sscanf((char*)arg,"%d:%f", &i, &target); + if (i != 1 && i != 2) + return AF_ERROR; + s->method = i-1; + s->mid_s16 = ((float)SHRT_MAX) * target; + s->mid_float = ((float)INT_MAX) * target; + return AF_OK; + } + } + return AF_UNKNOWN; +} + +// Deallocate memory +static void uninit(struct af_instance* af) +{ + free(af->data); + free(af->setup); +} + +static void method1_int16(af_volnorm_t *s, struct mp_audio *c) +{ + register int i = 0; + int16_t *data = (int16_t*)c->audio; // Audio data + int len = c->len/2; // Number of samples + float curavg = 0.0, newavg, neededmul; + int tmp; + + for (i = 0; i < len; i++) + { + tmp = data[i]; + curavg += tmp * tmp; + } + curavg = sqrt(curavg / (float) len); + + // Evaluate an adequate 'mul' coefficient based on previous state, current + // samples level, etc + + if (curavg > SIL_S16) + { + neededmul = s->mid_s16 / (curavg * s->mul); + s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; + + // clamp the mul coefficient + s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); + } + + // Scale & clamp the samples + for (i = 0; i < len; i++) + { + tmp = s->mul * data[i]; + tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); + data[i] = tmp; + } + + // Evaulation of newavg (not 100% accurate because of values clamping) + newavg = s->mul * curavg; + + // Stores computed values for future smoothing + s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; +} + +static void method1_float(af_volnorm_t *s, struct mp_audio *c) +{ + register int i = 0; + float *data = (float*)c->audio; // Audio data + int len = c->len/4; // Number of samples + float curavg = 0.0, newavg, neededmul, tmp; + + for (i = 0; i < len; i++) + { + tmp = data[i]; + curavg += tmp * tmp; + } + curavg = sqrt(curavg / (float) len); + + // Evaluate an adequate 'mul' coefficient based on previous state, current + // samples level, etc + + if (curavg > SIL_FLOAT) // FIXME + { + neededmul = s->mid_float / (curavg * s->mul); + s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul; + + // clamp the mul coefficient + s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); + } + + // Scale & clamp the samples + for (i = 0; i < len; i++) + data[i] *= s->mul; + + // Evaulation of newavg (not 100% accurate because of values clamping) + newavg = s->mul * curavg; + + // Stores computed values for future smoothing + s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg; +} + +static void method2_int16(af_volnorm_t *s, struct mp_audio *c) +{ + register int i = 0; + int16_t *data = (int16_t*)c->audio; // Audio data + int len = c->len/2; // Number of samples + float curavg = 0.0, newavg, avg = 0.0; + int tmp, totallen = 0; + + for (i = 0; i < len; i++) + { + tmp = data[i]; + curavg += tmp * tmp; + } + curavg = sqrt(curavg / (float) len); + + // Evaluate an adequate 'mul' coefficient based on previous state, current + // samples level, etc + for (i = 0; i < NSAMPLES; i++) + { + avg += s->mem[i].avg * (float)s->mem[i].len; + totallen += s->mem[i].len; + } + + if (totallen > MIN_SAMPLE_SIZE) + { + avg /= (float)totallen; + if (avg >= SIL_S16) + { + s->mul = s->mid_s16 / avg; + s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); + } + } + + // Scale & clamp the samples + for (i = 0; i < len; i++) + { + tmp = s->mul * data[i]; + tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); + data[i] = tmp; + } + + // Evaulation of newavg (not 100% accurate because of values clamping) + newavg = s->mul * curavg; + + // Stores computed values for future smoothing + s->mem[s->idx].len = len; + s->mem[s->idx].avg = newavg; + s->idx = (s->idx + 1) % NSAMPLES; +} + +static void method2_float(af_volnorm_t *s, struct mp_audio *c) +{ + register int i = 0; + float *data = (float*)c->audio; // Audio data + int len = c->len/4; // Number of samples + float curavg = 0.0, newavg, avg = 0.0, tmp; + int totallen = 0; + + for (i = 0; i < len; i++) + { + tmp = data[i]; + curavg += tmp * tmp; + } + curavg = sqrt(curavg / (float) len); + + // Evaluate an adequate 'mul' coefficient based on previous state, current + // samples level, etc + for (i = 0; i < NSAMPLES; i++) + { + avg += s->mem[i].avg * (float)s->mem[i].len; + totallen += s->mem[i].len; + } + + if (totallen > MIN_SAMPLE_SIZE) + { + avg /= (float)totallen; + if (avg >= SIL_FLOAT) + { + s->mul = s->mid_float / avg; + s->mul = clamp(s->mul, MUL_MIN, MUL_MAX); + } + } + + // Scale & clamp the samples + for (i = 0; i < len; i++) + data[i] *= s->mul; + + // Evaulation of newavg (not 100% accurate because of values clamping) + newavg = s->mul * curavg; + + // Stores computed values for future smoothing + s->mem[s->idx].len = len; + s->mem[s->idx].avg = newavg; + s->idx = (s->idx + 1) % NSAMPLES; +} + +// Filter data through filter +static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) +{ + af_volnorm_t *s = af->setup; + + if(af->data->format == (AF_FORMAT_S16_NE)) + { + if (s->method) + method2_int16(s, data); + else + method1_int16(s, data); + } + else if(af->data->format == (AF_FORMAT_FLOAT_NE)) + { + if (s->method) + method2_float(s, data); + else + method1_float(s, data); + } + return data; +} + +// 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_volnorm_t)); + if(af->data == NULL || af->setup == NULL) + return AF_ERROR; + + ((af_volnorm_t*)af->setup)->mul = MUL_INIT; + ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET; + ((af_volnorm_t*)af->setup)->idx = 0; + ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET; + ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET; + for (i = 0; i < NSAMPLES; i++) + { + ((af_volnorm_t*)af->setup)->mem[i].len = 0; + ((af_volnorm_t*)af->setup)->mem[i].avg = 0; + } + return AF_OK; +} + +// Description of this filter +struct af_info af_info_volnorm = { + "Volume normalizer filter", + "volnorm", + "Alex Beregszaszi & Pierre Lombard", + "", + AF_FLAGS_NOT_REENTRANT, + af_open +}; |