libavfilter/af_apulsator.c - manifest_repos/ffmpeg - Git at Google

 /*
  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg 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
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 #include "libavutil/avassert.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "internal.h"
 #include "audio.h"

 enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
 enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };

 typedef struct SimpleLFO {
     double phase;
     double freq;
     double offset;
     double amount;
     double pwidth;
     int mode;
     int srate;
 } SimpleLFO;

 typedef struct AudioPulsatorContext {
     const AVClass *class;
     int mode;
     double level_in;
     double level_out;
     double amount;
     double offset_l;
     double offset_r;
     double pwidth;
     double bpm;
     double hertz;
     int ms;
     int timing;

     SimpleLFO lfoL, lfoR;
 } AudioPulsatorContext;

 #define OFFSET(x) offsetof(AudioPulsatorContext, x)
 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption apulsator_options[] = {
     { "level_in",   "set input gain", OFFSET(level_in),  AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
     { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
     { "mode",             "set mode", OFFSET(mode),      AV_OPT_TYPE_INT,    {.i64=SINE}, SINE,   NB_MODES-1, FLAGS, "mode" },
     {   "sine",                 NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SINE},    0,            0, FLAGS, "mode" },
     {   "triangle",             NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=TRIANGLE},0,            0, FLAGS, "mode" },
     {   "square",               NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SQUARE},  0,            0, FLAGS, "mode" },
     {   "sawup",                NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWUP},   0,            0, FLAGS, "mode" },
     {   "sawdown",              NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=SAWDOWN}, 0,            0, FLAGS, "mode" },
     { "amount",     "set modulation", OFFSET(amount),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            1, FLAGS },
     { "offset_l",     "set offset L", OFFSET(offset_l),  AV_OPT_TYPE_DOUBLE, {.dbl=0},       0,            1, FLAGS },
     { "offset_r",     "set offset R", OFFSET(offset_r),  AV_OPT_TYPE_DOUBLE, {.dbl=.5},      0,            1, FLAGS },
     { "width",     "set pulse width", OFFSET(pwidth),    AV_OPT_TYPE_DOUBLE, {.dbl=1},       0,            2, FLAGS },
     { "timing",         "set timing", OFFSET(timing),    AV_OPT_TYPE_INT,    {.i64=2},       0, NB_TIMINGS-1, FLAGS, "timing" },
     {   "bpm",                  NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_BPM},  0,          0, FLAGS, "timing" },
     {   "ms",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_MS},   0,          0, FLAGS, "timing" },
     {   "hz",                   NULL, 0,                 AV_OPT_TYPE_CONST,  {.i64=UNIT_HZ},   0,          0, FLAGS, "timing" },
     { "bpm",               "set BPM", OFFSET(bpm),       AV_OPT_TYPE_DOUBLE, {.dbl=120},    30,          300, FLAGS },
     { "ms",                 "set ms", OFFSET(ms),        AV_OPT_TYPE_INT,    {.i64=500},    10,         2000, FLAGS },
     { "hz",          "set frequency", OFFSET(hertz),     AV_OPT_TYPE_DOUBLE, {.dbl=2},    0.01,          100, FLAGS },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(apulsator);

 static void lfo_advance(SimpleLFO *lfo, unsigned count)
 {
     lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
     if (lfo->phase >= 1)
         lfo->phase = fmod(lfo->phase, 1);
 }

 static double lfo_get_value(SimpleLFO *lfo)
 {
     double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
     double val;

     if (phs > 1)
         phs = fmod(phs, 1.);

     switch (lfo->mode) {
     case SINE:
         val = sin(phs * 2 * M_PI);
         break;
     case TRIANGLE:
         if (phs > 0.75)
             val = (phs - 0.75) * 4 - 1;
         else if (phs > 0.25)
             val = -4 * phs + 2;
         else
             val = phs * 4;
         break;
     case SQUARE:
         val = phs < 0.5 ? -1 : +1;
         break;
     case SAWUP:
         val = phs * 2 - 1;
         break;
     case SAWDOWN:
         val = 1 - phs * 2;
         break;
     default: av_assert0(0);
     }

     return val * lfo->amount;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     AudioPulsatorContext *s = ctx->priv;
     const double *src = (const double *)in->data[0];
     const int nb_samples = in->nb_samples;
     const double level_out = s->level_out;
     const double level_in = s->level_in;
     const double amount = s->amount;
     AVFrame *out;
     double *dst;
     int n;

     if (av_frame_is_writable(in)) {
         out = in;
     } else {
         out = ff_get_audio_buffer(inlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);
     }
     dst = (double *)out->data[0];

     for (n = 0; n < nb_samples; n++) {
         double outL;
         double outR;
         double inL = src[0] * level_in;
         double inR = src[1] * level_in;
         double procL = inL;
         double procR = inR;

         procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
         procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;

         outL = procL + inL * (1 - amount);
         outR = procR + inR * (1 - amount);

         outL *= level_out;
         outR *= level_out;

         dst[0] = outL;
         dst[1] = outR;

         lfo_advance(&s->lfoL, 1);
         lfo_advance(&s->lfoR, 1);

         dst += 2;
         src += 2;
     }

     if (in != out)
         av_frame_free(&in);

     return ff_filter_frame(outlink, out);
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layout = NULL;
     AVFilterFormats *formats = NULL;
     int ret;

     if ((ret = ff_add_format                 (&formats, AV_SAMPLE_FMT_DBL  )) < 0 ||
         (ret = ff_set_common_formats         (ctx     , formats            )) < 0 ||
         (ret = ff_add_channel_layout         (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
         (ret = ff_set_common_channel_layouts (ctx     , layout             )) < 0)
         return ret;

     formats = ff_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }

 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioPulsatorContext *s = ctx->priv;
     double freq;

     switch (s->timing) {
     case UNIT_BPM:  freq = s->bpm / 60;         break;
     case UNIT_MS:   freq = 1 / (s->ms / 1000.); break;
     case UNIT_HZ:   freq = s->hertz;            break;
     default: av_assert0(0);
     }

     s->lfoL.freq   = freq;
     s->lfoR.freq   = freq;
     s->lfoL.mode   = s->mode;
     s->lfoR.mode   = s->mode;
     s->lfoL.offset = s->offset_l;
     s->lfoR.offset = s->offset_r;
     s->lfoL.srate  = inlink->sample_rate;
     s->lfoR.srate  = inlink->sample_rate;
     s->lfoL.amount = s->amount;
     s->lfoR.amount = s->amount;
     s->lfoL.pwidth = s->pwidth;
     s->lfoR.pwidth = s->pwidth;

     return 0;
 }

 static const AVFilterPad inputs[] = {
     {
         .name         = "default",
         .type         = AVMEDIA_TYPE_AUDIO,
         .config_props = config_input,
         .filter_frame = filter_frame,
     },
     { NULL }
 };

 static const AVFilterPad outputs[] = {
     {
         .name = "default",
         .type = AVMEDIA_TYPE_AUDIO,
     },
     { NULL }
 };

 AVFilter ff_af_apulsator = {
     .name          = "apulsator",
     .description   = NULL_IF_CONFIG_SMALL("Audio pulsator."),
     .priv_size     = sizeof(AudioPulsatorContext),
     .priv_class    = &apulsator_class,
     .query_formats = query_formats,
     .inputs        = inputs,
     .outputs       = outputs,
 };
	/*
	* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg 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
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	#include "libavutil/avassert.h"
	#include "libavutil/opt.h"
	#include "avfilter.h"
	#include "internal.h"
	#include "audio.h"

	enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
	enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };

	typedef struct SimpleLFO {
	double phase;
	double freq;
	double offset;
	double amount;
	double pwidth;
	int mode;
	int srate;
	} SimpleLFO;

	typedef struct AudioPulsatorContext {
	const AVClass *class;
	int mode;
	double level_in;
	double level_out;
	double amount;
	double offset_l;
	double offset_r;
	double pwidth;
	double bpm;
	double hertz;
	int ms;
	int timing;

	SimpleLFO lfoL, lfoR;
	} AudioPulsatorContext;

	#define OFFSET(x) offsetof(AudioPulsatorContext, x)
	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption apulsator_options[] = {
	{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
	{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
	{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, "mode" },
	{ "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, "mode" },
	{ "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, "mode" },
	{ "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, "mode" },
	{ "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, "mode" },
	{ "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, "mode" },
	{ "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
	{ "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
	{ "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
	{ "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
	{ "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, "timing" },
	{ "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, "timing" },
	{ "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, "timing" },
	{ "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, "timing" },
	{ "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
	{ "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
	{ "hz", "set frequency", OFFSET(hertz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(apulsator);

	static void lfo_advance(SimpleLFO *lfo, unsigned count)
	{
	lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
	if (lfo->phase >= 1)
	lfo->phase = fmod(lfo->phase, 1);
	}

	static double lfo_get_value(SimpleLFO *lfo)
	{
	double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
	double val;

	if (phs > 1)
	phs = fmod(phs, 1.);

	switch (lfo->mode) {
	case SINE:
	val = sin(phs * 2 * M_PI);
	break;
	case TRIANGLE:
	if (phs > 0.75)
	val = (phs - 0.75) * 4 - 1;
	else if (phs > 0.25)
	val = -4 * phs + 2;
	else
	val = phs * 4;
	break;
	case SQUARE:
	val = phs < 0.5 ? -1 : +1;
	break;
	case SAWUP:
	val = phs * 2 - 1;
	break;
	case SAWDOWN:
	val = 1 - phs * 2;
	break;
	default: av_assert0(0);
	}

	return val * lfo->amount;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AVFilterLink *outlink = ctx->outputs[0];
	AudioPulsatorContext *s = ctx->priv;
	const double src = (const double )in->data[0];
	const int nb_samples = in->nb_samples;
	const double level_out = s->level_out;
	const double level_in = s->level_in;
	const double amount = s->amount;
	AVFrame *out;
	double *dst;
	int n;

	if (av_frame_is_writable(in)) {
	out = in;
	} else {
	out = ff_get_audio_buffer(inlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out, in);
	}
	dst = (double *)out->data[0];

	for (n = 0; n < nb_samples; n++) {
	double outL;
	double outR;
	double inL = src[0] * level_in;
	double inR = src[1] * level_in;
	double procL = inL;
	double procR = inR;

	procL = lfo_get_value(&s->lfoL) 0.5 + amount / 2;
	procR = lfo_get_value(&s->lfoR) 0.5 + amount / 2;

	outL = procL + inL * (1 - amount);
	outR = procR + inR * (1 - amount);

	outL *= level_out;
	outR *= level_out;

	dst[0] = outL;
	dst[1] = outR;

	lfo_advance(&s->lfoL, 1);
	lfo_advance(&s->lfoR, 1);

	dst += 2;
	src += 2;
	}

	if (in != out)
	av_frame_free(&in);

	return ff_filter_frame(outlink, out);
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterChannelLayouts *layout = NULL;
	AVFilterFormats *formats = NULL;
	int ret;

	if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 \|\|
	(ret = ff_set_common_formats (ctx , formats )) < 0 \|\|
	(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 \|\|
	(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
	return ret;

	formats = ff_all_samplerates();
	return ff_set_common_samplerates(ctx, formats);
	}

	static int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	AudioPulsatorContext *s = ctx->priv;
	double freq;

	switch (s->timing) {
	case UNIT_BPM: freq = s->bpm / 60; break;
	case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
	case UNIT_HZ: freq = s->hertz; break;
	default: av_assert0(0);
	}

	s->lfoL.freq = freq;
	s->lfoR.freq = freq;
	s->lfoL.mode = s->mode;
	s->lfoR.mode = s->mode;
	s->lfoL.offset = s->offset_l;
	s->lfoR.offset = s->offset_r;
	s->lfoL.srate = inlink->sample_rate;
	s->lfoR.srate = inlink->sample_rate;
	s->lfoL.amount = s->amount;
	s->lfoR.amount = s->amount;
	s->lfoL.pwidth = s->pwidth;
	s->lfoR.pwidth = s->pwidth;

	return 0;
	}

	static const AVFilterPad inputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	.config_props = config_input,
	.filter_frame = filter_frame,
	},
	{ NULL }
	};

	static const AVFilterPad outputs[] = {
	{
	.name = "default",
	.type = AVMEDIA_TYPE_AUDIO,
	},
	{ NULL }
	};

	AVFilter ff_af_apulsator = {
	.name = "apulsator",
	.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
	.priv_size = sizeof(AudioPulsatorContext),
	.priv_class = &apulsator_class,
	.query_formats = query_formats,
	.inputs = inputs,
	.outputs = outputs,
	};