libavfilter/af_asubboost.c - manifest_repos/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
  */

 #include "libavutil/channel_layout.h"
 #include "libavutil/ffmath.h"
 #include "libavutil/opt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "formats.h"

 typedef struct ASubBoostContext {
     const AVClass *class;

     double dry_gain;
     double wet_gain;
     double feedback;
     double decay;
     double delay;
     double cutoff;
     double slope;

     double a0, a1, a2;
     double b0, b1, b2;

     int *write_pos;
     int buffer_samples;

     AVFrame *w;
     AVFrame *buffer;
 } ASubBoostContext;

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

     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_all_samplerates();
     return ff_set_common_samplerates(ctx, formats);
 }

 static int get_coeffs(AVFilterContext *ctx)
 {
     ASubBoostContext *s = ctx->priv;
     AVFilterLink *inlink = ctx->inputs[0];
     double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
     double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);

     s->a0 =  1 + alpha;
     s->a1 = -2 * cos(w0);
     s->a2 =  1 - alpha;
     s->b0 = (1 - cos(w0)) / 2;
     s->b1 =  1 - cos(w0);
     s->b2 = (1 - cos(w0)) / 2;

     s->a1 /= s->a0;
     s->a2 /= s->a0;
     s->b0 /= s->a0;
     s->b1 /= s->a0;
     s->b2 /= s->a0;

     s->buffer_samples = inlink->sample_rate * s->delay / 1000;

     return 0;
 }

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

     s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
     s->w = ff_get_audio_buffer(inlink, 2);
     s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
     if (!s->buffer || !s->w || !s->write_pos)
         return AVERROR(ENOMEM);

     return get_coeffs(ctx);
 }

 typedef struct ThreadData {
     AVFrame *in, *out;
 } ThreadData;

 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
 {
     ASubBoostContext *s = ctx->priv;
     ThreadData *td = arg;
     AVFrame *out = td->out;
     AVFrame *in = td->in;
     const double mix = ctx->is_disabled ? 0. : 1.;
     const double wet = ctx->is_disabled ? 1. : s->wet_gain;
     const double dry = ctx->is_disabled ? 1. : s->dry_gain;
     const double feedback = s->feedback, decay = s->decay;
     const double b0 = s->b0;
     const double b1 = s->b1;
     const double b2 = s->b2;
     const double a1 = -s->a1;
     const double a2 = -s->a2;
     const int start = (in->channels * jobnr) / nb_jobs;
     const int end = (in->channels * (jobnr+1)) / nb_jobs;
     const int buffer_samples = s->buffer_samples;

     for (int ch = start; ch < end; ch++) {
         const double *src = (const double *)in->extended_data[ch];
         double *dst = (double *)out->extended_data[ch];
         double *buffer = (double *)s->buffer->extended_data[ch];
         double *w = (double *)s->w->extended_data[ch];
         int write_pos = s->write_pos[ch];

         for (int n = 0; n < in->nb_samples; n++) {
             double out_sample;

             out_sample = src[n] * b0 + w[0];
             w[0] = b1 * src[n] + w[1] + a1 * out_sample;
             w[1] = b2 * src[n] + a2 * out_sample;

             buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
             dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;

             if (++write_pos >= buffer_samples)
                 write_pos = 0;
         }

         s->write_pos[ch] = write_pos;
     }

     return 0;
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
     ThreadData td;
     AVFrame *out;

     if (av_frame_is_writable(in)) {
         out = in;
     } else {
         out = ff_get_audio_buffer(outlink, in->nb_samples);
         if (!out) {
             av_frame_free(&in);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out, in);
     }

     td.in = in; td.out = out;
     ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
                                                             ff_filter_get_nb_threads(ctx)));

     if (out != in)
         av_frame_free(&in);
     return ff_filter_frame(outlink, out);
 }

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

     av_frame_free(&s->buffer);
     av_frame_free(&s->w);
     av_freep(&s->write_pos);
 }

 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                            char *res, int res_len, int flags)
 {
     int ret;

     ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
     if (ret < 0)
         return ret;

     return get_coeffs(ctx);
 }

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

 static const AVOption asubboost_options[] = {
     { "dry",      "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
     { "wet",      "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
     { "decay",    "set decay",    OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=0.7},      0,   1, FLAGS },
     { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9},      0,   1, FLAGS },
     { "cutoff",   "set cutoff",   OFFSET(cutoff),   AV_OPT_TYPE_DOUBLE, {.dbl=100},     50, 900, FLAGS },
     { "slope",    "set slope",    OFFSET(slope),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001,   1, FLAGS },
     { "delay",    "set delay",    OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=20},       1, 100, FLAGS },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(asubboost);

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

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

 AVFilter ff_af_asubboost = {
     .name           = "asubboost",
     .description    = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
     .query_formats  = query_formats,
     .priv_size      = sizeof(ASubBoostContext),
     .priv_class     = &asubboost_class,
     .uninit         = uninit,
     .inputs         = inputs,
     .outputs        = outputs,
     .process_command = process_command,
     .flags           = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
                        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
	*/

	#include "libavutil/channel_layout.h"
	#include "libavutil/ffmath.h"
	#include "libavutil/opt.h"
	#include "avfilter.h"
	#include "audio.h"
	#include "formats.h"

	typedef struct ASubBoostContext {
	const AVClass *class;

	double dry_gain;
	double wet_gain;
	double feedback;
	double decay;
	double delay;
	double cutoff;
	double slope;

	double a0, a1, a2;
	double b0, b1, b2;

	int *write_pos;
	int buffer_samples;

	AVFrame *w;
	AVFrame *buffer;
	} ASubBoostContext;

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

	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_all_samplerates();
	return ff_set_common_samplerates(ctx, formats);
	}

	static int get_coeffs(AVFilterContext *ctx)
	{
	ASubBoostContext *s = ctx->priv;
	AVFilterLink *inlink = ctx->inputs[0];
	double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
	double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);

	s->a0 = 1 + alpha;
	s->a1 = -2 * cos(w0);
	s->a2 = 1 - alpha;
	s->b0 = (1 - cos(w0)) / 2;
	s->b1 = 1 - cos(w0);
	s->b2 = (1 - cos(w0)) / 2;

	s->a1 /= s->a0;
	s->a2 /= s->a0;
	s->b0 /= s->a0;
	s->b1 /= s->a0;
	s->b2 /= s->a0;

	s->buffer_samples = inlink->sample_rate * s->delay / 1000;

	return 0;
	}

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

	s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
	s->w = ff_get_audio_buffer(inlink, 2);
	s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
	if (!s->buffer \|\| !s->w \|\| !s->write_pos)
	return AVERROR(ENOMEM);

	return get_coeffs(ctx);
	}

	typedef struct ThreadData {
	AVFrame in, out;
	} ThreadData;

	static int filter_channels(AVFilterContext ctx, void arg, int jobnr, int nb_jobs)
	{
	ASubBoostContext *s = ctx->priv;
	ThreadData *td = arg;
	AVFrame *out = td->out;
	AVFrame *in = td->in;
	const double mix = ctx->is_disabled ? 0. : 1.;
	const double wet = ctx->is_disabled ? 1. : s->wet_gain;
	const double dry = ctx->is_disabled ? 1. : s->dry_gain;
	const double feedback = s->feedback, decay = s->decay;
	const double b0 = s->b0;
	const double b1 = s->b1;
	const double b2 = s->b2;
	const double a1 = -s->a1;
	const double a2 = -s->a2;
	const int start = (in->channels * jobnr) / nb_jobs;
	const int end = (in->channels * (jobnr+1)) / nb_jobs;
	const int buffer_samples = s->buffer_samples;

	for (int ch = start; ch < end; ch++) {
	const double src = (const double )in->extended_data[ch];
	double dst = (double )out->extended_data[ch];
	double buffer = (double )s->buffer->extended_data[ch];
	double w = (double )s->w->extended_data[ch];
	int write_pos = s->write_pos[ch];

	for (int n = 0; n < in->nb_samples; n++) {
	double out_sample;

	out_sample = src[n] * b0 + w[0];
	w[0] = b1 * src[n] + w[1] + a1 * out_sample;
	w[1] = b2 * src[n] + a2 * out_sample;

	buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
	dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;

	if (++write_pos >= buffer_samples)
	write_pos = 0;
	}

	s->write_pos[ch] = write_pos;
	}

	return 0;
	}

	static int filter_frame(AVFilterLink inlink, AVFrame in)
	{
	AVFilterContext *ctx = inlink->dst;
	AVFilterLink *outlink = ctx->outputs[0];
	ThreadData td;
	AVFrame *out;

	if (av_frame_is_writable(in)) {
	out = in;
	} else {
	out = ff_get_audio_buffer(outlink, in->nb_samples);
	if (!out) {
	av_frame_free(&in);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out, in);
	}

	td.in = in; td.out = out;
	ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
	ff_filter_get_nb_threads(ctx)));

	if (out != in)
	av_frame_free(&in);
	return ff_filter_frame(outlink, out);
	}

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

	av_frame_free(&s->buffer);
	av_frame_free(&s->w);
	av_freep(&s->write_pos);
	}

	static int process_command(AVFilterContext ctx, const char cmd, const char *args,
	char *res, int res_len, int flags)
	{
	int ret;

	ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
	if (ret < 0)
	return ret;

	return get_coeffs(ctx);
	}

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

	static const AVOption asubboost_options[] = {
	{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
	{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
	{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
	{ "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9}, 0, 1, FLAGS },
	{ "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
	{ "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
	{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(asubboost);

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

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

	AVFilter ff_af_asubboost = {
	.name = "asubboost",
	.description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
	.query_formats = query_formats,
	.priv_size = sizeof(ASubBoostContext),
	.priv_class = &asubboost_class,
	.uninit = uninit,
	.inputs = inputs,
	.outputs = outputs,
	.process_command = process_command,
	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL \|
	AVFILTER_FLAG_SLICE_THREADS,
	};