libavfilter/af_acrossover.c - third_party/ffmpeg - Git at Google

 /*
  * 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
  */

 /**
  * @file
  * Crossover filter
  *
  * Split an audio stream into several bands.
  */

 #include "libavutil/attributes.h"
 #include "libavutil/avstring.h"
 #include "libavutil/channel_layout.h"
 #include "libavutil/eval.h"
 #include "libavutil/internal.h"
 #include "libavutil/opt.h"

 #include "audio.h"
 #include "avfilter.h"
 #include "formats.h"
 #include "internal.h"

 #define MAX_SPLITS 16
 #define MAX_BANDS MAX_SPLITS + 1

 typedef struct BiquadContext {
     double a0, a1, a2;
     double b1, b2;
     double i1, i2;
     double o1, o2;
 } BiquadContext;

 typedef struct CrossoverChannel {
     BiquadContext lp[MAX_BANDS][4];
     BiquadContext hp[MAX_BANDS][4];
 } CrossoverChannel;

 typedef struct AudioCrossoverContext {
     const AVClass *class;

     char *splits_str;
     int order;

     int filter_count;
     int nb_splits;
     float *splits;

     CrossoverChannel *xover;

     AVFrame *input_frame;
     AVFrame *frames[MAX_BANDS];
 } AudioCrossoverContext;

 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption acrossover_options[] = {
     { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
     { "order", "set order",             OFFSET(order),      AV_OPT_TYPE_INT,    {.i64=1},     0, 2, AF, "m" },
     { "2nd",   "2nd order",             0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0, 0, AF, "m" },
     { "4th",   "4th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0, 0, AF, "m" },
     { "8th",   "8th order",             0,                  AV_OPT_TYPE_CONST,  {.i64=2},     0, 0, AF, "m" },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(acrossover);

 static av_cold int init(AVFilterContext *ctx)
 {
     AudioCrossoverContext *s = ctx->priv;
     char *p, *arg, *saveptr = NULL;
     int i, ret = 0;

     s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
     if (!s->splits)
         return AVERROR(ENOMEM);

     p = s->splits_str;
     for (i = 0; i < MAX_SPLITS; i++) {
         float freq;

         if (!(arg = av_strtok(p, " |", &saveptr)))
             break;

         p = NULL;

         if (av_sscanf(arg, "%f", &freq) != 1) {
             av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
             return AVERROR(EINVAL);
         }
         if (freq <= 0) {
             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
             return AVERROR(EINVAL);
         }

         if (i > 0 && freq <= s->splits[i-1]) {
             av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
             return AVERROR(EINVAL);
         }

         s->splits[i] = freq;
     }

     s->nb_splits = i;

     for (i = 0; i <= s->nb_splits; i++) {
         AVFilterPad pad  = { 0 };
         char *name;

         pad.type = AVMEDIA_TYPE_AUDIO;
         name = av_asprintf("out%d", ctx->nb_outputs);
         if (!name)
             return AVERROR(ENOMEM);
         pad.name = name;

         if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
             av_freep(&pad.name);
             return ret;
         }
     }

     return ret;
 }

 static void set_lp(BiquadContext *b, double fc, double q, double sr)
 {
     double omega = 2.0 * M_PI * fc / sr;
     double sn = sin(omega);
     double cs = cos(omega);
     double alpha = sn / (2. * q);
     double inv = 1.0 / (1.0 + alpha);

     b->a0 = (1. - cs) * 0.5 * inv;
     b->a1 = (1. - cs) * inv;
     b->a2 = b->a0;
     b->b1 = -2. * cs * inv;
     b->b2 = (1. - alpha) * inv;
 }

 static void set_hp(BiquadContext *b, double fc, double q, double sr)
 {
     double omega = 2 * M_PI * fc / sr;
     double sn = sin(omega);
     double cs = cos(omega);
     double alpha = sn / (2 * q);
     double inv = 1.0 / (1.0 + alpha);

     b->a0 = inv * (1. + cs) / 2.;
     b->a1 = -2. * b->a0;
     b->a2 = b->a0;
     b->b1 = -2. * cs * inv;
     b->b2 = (1. - alpha) * inv;
 }

 static int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioCrossoverContext *s = ctx->priv;
     int ch, band, sample_rate = inlink->sample_rate;
     double q;

     s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
     if (!s->xover)
         return AVERROR(ENOMEM);

     switch (s->order) {
     case 0:
         q = 0.5;
         s->filter_count = 1;
         break;
     case 1:
         q = M_SQRT1_2;
         s->filter_count = 2;
         break;
     case 2:
         q = 0.54;
         s->filter_count = 4;
         break;
     }

     for (ch = 0; ch < inlink->channels; ch++) {
         for (band = 0; band <= s->nb_splits; band++) {
             set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
             set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);

             if (s->order > 1) {
                 set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
                 set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
                 set_lp(&s->xover[ch].lp[band][2], s->splits[band],    q, sample_rate);
                 set_hp(&s->xover[ch].hp[band][2], s->splits[band],    q, sample_rate);
                 set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
                 set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
             } else {
                 set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
                 set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
             }
         }
     }

     return 0;
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterFormats *formats;
     AVFilterChannelLayouts *layouts;
     static const enum AVSampleFormat sample_fmts[] = {
         AV_SAMPLE_FMT_DBLP,
         AV_SAMPLE_FMT_NONE
     };
     int ret;

     layouts = ff_all_channel_counts();
     if (!layouts)
         return AVERROR(ENOMEM);
     ret = ff_set_common_channel_layouts(ctx, layouts);
     if (ret < 0)
         return ret;

     formats = ff_make_format_list(sample_fmts);
     if (!formats)
         return AVERROR(ENOMEM);
     ret = ff_set_common_formats(ctx, formats);
     if (ret < 0)
         return ret;

     formats = ff_all_samplerates();
     if (!formats)
         return AVERROR(ENOMEM);
     return ff_set_common_samplerates(ctx, formats);
 }

 static double biquad_process(BiquadContext *b, double in)
 {
     double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;

     b->i2 = b->i1;
     b->o2 = b->o1;
     b->i1 = in;
     b->o1 = out;

     return out;
 }

 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     AudioCrossoverContext *s = ctx->priv;
     AVFrame *in = s->input_frame;
     AVFrame **frames = s->frames;
     const int start = (in->channels * jobnr) / nb_jobs;
     const int end = (in->channels * (jobnr+1)) / nb_jobs;
     int f, band;

     for (int ch = start; ch < end; ch++) {
         const double *src = (const double *)in->extended_data[ch];
         CrossoverChannel *xover = &s->xover[ch];

         for (int i = 0; i < in->nb_samples; i++) {
             double sample = src[i], lo, hi;

             for (band = 0; band < ctx->nb_outputs; band++) {
                 double *dst = (double *)frames[band]->extended_data[ch];

                 lo = sample;
                 hi = sample;
                 for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
                     BiquadContext *lp = &xover->lp[band][f];
                     lo = biquad_process(lp, lo);
                 }

                 for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
                     BiquadContext *hp = &xover->hp[band][f];
                     hi = biquad_process(hp, hi);
                 }

                 dst[i] = lo;

                 sample = hi;
             }
         }
     }

     return 0;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AudioCrossoverContext *s = ctx->priv;
     AVFrame **frames = s->frames;
     int i, ret = 0;

     for (i = 0; i < ctx->nb_outputs; i++) {
         frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);

         if (!frames[i]) {
             ret = AVERROR(ENOMEM);
             break;
         }

         frames[i]->pts = in->pts;
     }

     if (ret < 0)
         goto fail;

     s->input_frame = in;
     ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
                                                                    ff_filter_get_nb_threads(ctx)));

     for (i = 0; i < ctx->nb_outputs; i++) {
         ret = ff_filter_frame(ctx->outputs[i], frames[i]);
         frames[i] = NULL;
         if (ret < 0)
             break;
     }

 fail:
     for (i = 0; i < ctx->nb_outputs; i++)
         av_frame_free(&frames[i]);
     av_frame_free(&in);
     s->input_frame = NULL;

     return ret;
 }

 static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioCrossoverContext *s = ctx->priv;
     int i;

     av_freep(&s->splits);
     av_freep(&s->xover);

     for (i = 0; i < ctx->nb_outputs; i++)
         av_freep(&ctx->output_pads[i].name);
 }

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

 AVFilter ff_af_acrossover = {
     .name           = "acrossover",
     .description    = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
     .priv_size      = sizeof(AudioCrossoverContext),
     .priv_class     = &acrossover_class,
     .init           = init,
     .uninit         = uninit,
     .query_formats  = query_formats,
     .inputs         = inputs,
     .outputs        = NULL,
     .flags          = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
                       AVFILTER_FLAG_SLICE_THREADS,
 };
	/*
	* 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
	*/

	/**
	* @file
	* Crossover filter
	*
	* Split an audio stream into several bands.
	*/

	#include "libavutil/attributes.h"
	#include "libavutil/avstring.h"
	#include "libavutil/channel_layout.h"
	#include "libavutil/eval.h"
	#include "libavutil/internal.h"
	#include "libavutil/opt.h"

	#include "audio.h"
	#include "avfilter.h"
	#include "formats.h"
	#include "internal.h"

	#define MAX_SPLITS 16
	#define MAX_BANDS MAX_SPLITS + 1

	typedef struct BiquadContext {
	double a0, a1, a2;
	double b1, b2;
	double i1, i2;
	double o1, o2;
	} BiquadContext;

	typedef struct CrossoverChannel {
	BiquadContext lp[MAX_BANDS][4];
	BiquadContext hp[MAX_BANDS][4];
	} CrossoverChannel;

	typedef struct AudioCrossoverContext {
	const AVClass *class;

	char *splits_str;
	int order;

	int filter_count;
	int nb_splits;
	float *splits;

	CrossoverChannel *xover;

	AVFrame *input_frame;
	AVFrame *frames[MAX_BANDS];
	} AudioCrossoverContext;

	#define OFFSET(x) offsetof(AudioCrossoverContext, x)
	#define AF AV_OPT_FLAG_AUDIO_PARAM \| AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption acrossover_options[] = {
	{ "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
	{ "order", "set order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
	{ "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
	{ "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
	{ "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(acrossover);

	static av_cold int init(AVFilterContext *ctx)
	{
	AudioCrossoverContext *s = ctx->priv;
	char p, arg, *saveptr = NULL;
	int i, ret = 0;

	s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
	if (!s->splits)
	return AVERROR(ENOMEM);

	p = s->splits_str;
	for (i = 0; i < MAX_SPLITS; i++) {
	float freq;

	if (!(arg = av_strtok(p, " \|", &saveptr)))
	break;

	p = NULL;

	if (av_sscanf(arg, "%f", &freq) != 1) {
	av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
	return AVERROR(EINVAL);
	}
	if (freq <= 0) {
	av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
	return AVERROR(EINVAL);
	}

	if (i > 0 && freq <= s->splits[i-1]) {
	av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
	return AVERROR(EINVAL);
	}

	s->splits[i] = freq;
	}

	s->nb_splits = i;

	for (i = 0; i <= s->nb_splits; i++) {
	AVFilterPad pad = { 0 };
	char *name;

	pad.type = AVMEDIA_TYPE_AUDIO;
	name = av_asprintf("out%d", ctx->nb_outputs);
	if (!name)
	return AVERROR(ENOMEM);
	pad.name = name;

	if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
	av_freep(&pad.name);
	return ret;
	}
	}

	return ret;
	}

	static void set_lp(BiquadContext *b, double fc, double q, double sr)
	{
	double omega = 2.0 * M_PI * fc / sr;
	double sn = sin(omega);
	double cs = cos(omega);
	double alpha = sn / (2. * q);
	double inv = 1.0 / (1.0 + alpha);

	b->a0 = (1. - cs) * 0.5 * inv;
	b->a1 = (1. - cs) * inv;
	b->a2 = b->a0;
	b->b1 = -2. * cs * inv;
	b->b2 = (1. - alpha) * inv;
	}

	static void set_hp(BiquadContext *b, double fc, double q, double sr)
	{
	double omega = 2 * M_PI * fc / sr;
	double sn = sin(omega);
	double cs = cos(omega);
	double alpha = sn / (2 * q);
	double inv = 1.0 / (1.0 + alpha);

	b->a0 = inv * (1. + cs) / 2.;
	b->a1 = -2. * b->a0;
	b->a2 = b->a0;
	b->b1 = -2. * cs * inv;
	b->b2 = (1. - alpha) * inv;
	}

	static int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	AudioCrossoverContext *s = ctx->priv;
	int ch, band, sample_rate = inlink->sample_rate;
	double q;

	s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
	if (!s->xover)
	return AVERROR(ENOMEM);

	switch (s->order) {
	case 0:
	q = 0.5;
	s->filter_count = 1;
	break;
	case 1:
	q = M_SQRT1_2;
	s->filter_count = 2;
	break;
	case 2:
	q = 0.54;
	s->filter_count = 4;
	break;
	}

	for (ch = 0; ch < inlink->channels; ch++) {
	for (band = 0; band <= s->nb_splits; band++) {
	set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
	set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);

	if (s->order > 1) {
	set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
	set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
	set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
	set_hp(&s->xover[ch].hp[band][2], s->splits[band], q, sample_rate);
	set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
	set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
	} else {
	set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
	set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
	}
	}
	}

	return 0;
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterFormats *formats;
	AVFilterChannelLayouts *layouts;
	static const enum AVSampleFormat sample_fmts[] = {
	AV_SAMPLE_FMT_DBLP,
	AV_SAMPLE_FMT_NONE
	};
	int ret;

	layouts = ff_all_channel_counts();
	if (!layouts)
	return AVERROR(ENOMEM);
	ret = ff_set_common_channel_layouts(ctx, layouts);
	if (ret < 0)
	return ret;

	formats = ff_make_format_list(sample_fmts);
	if (!formats)
	return AVERROR(ENOMEM);
	ret = ff_set_common_formats(ctx, formats);
	if (ret < 0)
	return ret;

	formats = ff_all_samplerates();
	if (!formats)
	return AVERROR(ENOMEM);
	return ff_set_common_samplerates(ctx, formats);
	}

	static double biquad_process(BiquadContext *b, double in)
	{
	double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;

	b->i2 = b->i1;
	b->o2 = b->o1;
	b->i1 = in;
	b->o1 = out;

	return out;
	}

	static int filter_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
	{
	AudioCrossoverContext *s = ctx->priv;
	AVFrame *in = s->input_frame;
	AVFrame **frames = s->frames;
	const int start = (in->channels * jobnr) / nb_jobs;
	const int end = (in->channels * (jobnr+1)) / nb_jobs;
	int f, band;

	for (int ch = start; ch < end; ch++) {
	const double src = (const double )in->extended_data[ch];
	CrossoverChannel *xover = &s->xover[ch];

	for (int i = 0; i < in->nb_samples; i++) {
	double sample = src[i], lo, hi;

	for (band = 0; band < ctx->nb_outputs; band++) {
	double dst = (double )frames[band]->extended_data[ch];

	lo = sample;
	hi = sample;
	for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
	BiquadContext *lp = &xover->lp[band][f];
	lo = biquad_process(lp, lo);
	}

	for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
	BiquadContext *hp = &xover->hp[band][f];
	hi = biquad_process(hp, hi);
	}

	dst[i] = lo;

	sample = hi;
	}
	}
	}

	return 0;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AudioCrossoverContext *s = ctx->priv;
	AVFrame **frames = s->frames;
	int i, ret = 0;

	for (i = 0; i < ctx->nb_outputs; i++) {
	frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);

	if (!frames[i]) {
	ret = AVERROR(ENOMEM);
	break;
	}

	frames[i]->pts = in->pts;
	}

	if (ret < 0)
	goto fail;

	s->input_frame = in;
	ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
	ff_filter_get_nb_threads(ctx)));

	for (i = 0; i < ctx->nb_outputs; i++) {
	ret = ff_filter_frame(ctx->outputs[i], frames[i]);
	frames[i] = NULL;
	if (ret < 0)
	break;
	}

	fail:
	for (i = 0; i < ctx->nb_outputs; i++)
	av_frame_free(&frames[i]);
	av_frame_free(&in);
	s->input_frame = NULL;

	return ret;
	}

	static av_cold void uninit(AVFilterContext *ctx)
	{
	AudioCrossoverContext *s = ctx->priv;
	int i;

	av_freep(&s->splits);
	av_freep(&s->xover);

	for (i = 0; i < ctx->nb_outputs; i++)
	av_freep(&ctx->output_pads[i].name);
	}

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

	AVFilter ff_af_acrossover = {
	.name = "acrossover",
	.description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
	.priv_size = sizeof(AudioCrossoverContext),
	.priv_class = &acrossover_class,
	.init = init,
	.uninit = uninit,
	.query_formats = query_formats,
	.inputs = inputs,
	.outputs = NULL,
	.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS \|
	AVFILTER_FLAG_SLICE_THREADS,
	};