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

 /*
  * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
  *
  * 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/avstring.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 #include "avfilter.h"
 #include "audio.h"
 #include "internal.h"
 #include "generate_wave_table.h"

 #define INTERPOLATION_LINEAR    0
 #define INTERPOLATION_QUADRATIC 1

 typedef struct FlangerContext {
     const AVClass *class;
     double delay_min;
     double delay_depth;
     double feedback_gain;
     double delay_gain;
     double speed;
     int wave_shape;
     double channel_phase;
     int interpolation;
     double in_gain;
     int max_samples;
     uint8_t **delay_buffer;
     int delay_buf_pos;
     double *delay_last;
     float *lfo;
     int lfo_length;
     int lfo_pos;
 } FlangerContext;

 #define OFFSET(x) offsetof(FlangerContext, x)
 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

 static const AVOption flanger_options[] = {
     { "delay", "base delay in milliseconds",        OFFSET(delay_min),   AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
     { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
     { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
     { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
     { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
     { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
     { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
     { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, A, "type" },
     { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
     { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, A, "type" },
     { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
     { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
     { "linear",     NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_LINEAR},    0, 0, A, "itype" },
     { "quadratic",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
     { NULL }
 };

 AVFILTER_DEFINE_CLASS(flanger);

 static av_cold int init(AVFilterContext *ctx)
 {
     FlangerContext *s = ctx->priv;

     s->feedback_gain /= 100;
     s->delay_gain    /= 100;
     s->channel_phase /= 100;
     s->delay_min     /= 1000;
     s->delay_depth   /= 1000;
     s->in_gain        = 1 / (1 + s->delay_gain);
     s->delay_gain    /= 1 + s->delay_gain;
     s->delay_gain    *= 1 - fabs(s->feedback_gain);

     return 0;
 }

 static int query_formats(AVFilterContext *ctx)
 {
     AVFilterChannelLayouts *layouts;
     AVFilterFormats *formats;
     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 int config_input(AVFilterLink *inlink)
 {
     AVFilterContext *ctx = inlink->dst;
     FlangerContext *s = ctx->priv;

     s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
     s->lfo_length  = inlink->sample_rate / s->speed;
     s->delay_last  = av_calloc(inlink->channels, sizeof(*s->delay_last));
     s->lfo         = av_calloc(s->lfo_length, sizeof(*s->lfo));
     if (!s->lfo || !s->delay_last)
         return AVERROR(ENOMEM);

     ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
                            rint(s->delay_min * inlink->sample_rate),
                            s->max_samples - 2., 3 * M_PI_2);

     return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
                                               inlink->channels, s->max_samples,
                                               inlink->format, 0);
 }

 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
 {
     AVFilterContext *ctx = inlink->dst;
     FlangerContext *s = ctx->priv;
     AVFrame *out_frame;
     int chan, i;

     if (av_frame_is_writable(frame)) {
         out_frame = frame;
     } else {
         out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
         if (!out_frame) {
             av_frame_free(&frame);
             return AVERROR(ENOMEM);
         }
         av_frame_copy_props(out_frame, frame);
     }

     for (i = 0; i < frame->nb_samples; i++) {

         s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;

         for (chan = 0; chan < inlink->channels; chan++) {
             double *src = (double *)frame->extended_data[chan];
             double *dst = (double *)out_frame->extended_data[chan];
             double delayed_0, delayed_1;
             double delayed;
             double in, out;
             int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
             double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
             int int_delay = (int)delay;
             double frac_delay = modf(delay, &delay);
             double *delay_buffer = (double *)s->delay_buffer[chan];

             in = src[i];
             delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
                                                            s->feedback_gain;
             delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
             delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];

             if (s->interpolation == INTERPOLATION_LINEAR) {
                 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
             } else {
                 double a, b;
                 double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
                 delayed_2 -= delayed_0;
                 delayed_1 -= delayed_0;
                 a = delayed_2 * .5 - delayed_1;
                 b = delayed_1 *  2 - delayed_2 *.5;
                 delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
             }

             s->delay_last[chan] = delayed;
             out = in * s->in_gain + delayed * s->delay_gain;
             dst[i] = out;
         }
         s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
     }

     if (frame != out_frame)
         av_frame_free(&frame);

     return ff_filter_frame(ctx->outputs[0], out_frame);
 }

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

     av_freep(&s->lfo);
     av_freep(&s->delay_last);

     if (s->delay_buffer)
         av_freep(&s->delay_buffer[0]);
     av_freep(&s->delay_buffer);
 }

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

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

 AVFilter ff_af_flanger = {
     .name          = "flanger",
     .description   = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
     .query_formats = query_formats,
     .priv_size     = sizeof(FlangerContext),
     .priv_class    = &flanger_class,
     .init          = init,
     .uninit        = uninit,
     .inputs        = flanger_inputs,
     .outputs       = flanger_outputs,
 };
	/*
	* Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
	*
	* 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/avstring.h"
	#include "libavutil/opt.h"
	#include "libavutil/samplefmt.h"
	#include "avfilter.h"
	#include "audio.h"
	#include "internal.h"
	#include "generate_wave_table.h"

	#define INTERPOLATION_LINEAR 0
	#define INTERPOLATION_QUADRATIC 1

	typedef struct FlangerContext {
	const AVClass *class;
	double delay_min;
	double delay_depth;
	double feedback_gain;
	double delay_gain;
	double speed;
	int wave_shape;
	double channel_phase;
	int interpolation;
	double in_gain;
	int max_samples;
	uint8_t **delay_buffer;
	int delay_buf_pos;
	double *delay_last;
	float *lfo;
	int lfo_length;
	int lfo_pos;
	} FlangerContext;

	#define OFFSET(x) offsetof(FlangerContext, x)
	#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM

	static const AVOption flanger_options[] = {
	{ "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
	{ "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
	{ "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
	{ "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
	{ "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
	{ "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
	{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
	{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
	{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
	{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
	{ "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
	{ "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
	{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
	{ "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
	{ NULL }
	};

	AVFILTER_DEFINE_CLASS(flanger);

	static av_cold int init(AVFilterContext *ctx)
	{
	FlangerContext *s = ctx->priv;

	s->feedback_gain /= 100;
	s->delay_gain /= 100;
	s->channel_phase /= 100;
	s->delay_min /= 1000;
	s->delay_depth /= 1000;
	s->in_gain = 1 / (1 + s->delay_gain);
	s->delay_gain /= 1 + s->delay_gain;
	s->delay_gain *= 1 - fabs(s->feedback_gain);

	return 0;
	}

	static int query_formats(AVFilterContext *ctx)
	{
	AVFilterChannelLayouts *layouts;
	AVFilterFormats *formats;
	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 int config_input(AVFilterLink *inlink)
	{
	AVFilterContext *ctx = inlink->dst;
	FlangerContext *s = ctx->priv;

	s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
	s->lfo_length = inlink->sample_rate / s->speed;
	s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
	s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
	if (!s->lfo \|\| !s->delay_last)
	return AVERROR(ENOMEM);

	ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
	rint(s->delay_min * inlink->sample_rate),
	s->max_samples - 2., 3 * M_PI_2);

	return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
	inlink->channels, s->max_samples,
	inlink->format, 0);
	}

	static int filter_frame(AVFilterLink inlink, AVFrame frame)
	{
	AVFilterContext *ctx = inlink->dst;
	FlangerContext *s = ctx->priv;
	AVFrame *out_frame;
	int chan, i;

	if (av_frame_is_writable(frame)) {
	out_frame = frame;
	} else {
	out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
	if (!out_frame) {
	av_frame_free(&frame);
	return AVERROR(ENOMEM);
	}
	av_frame_copy_props(out_frame, frame);
	}

	for (i = 0; i < frame->nb_samples; i++) {

	s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;

	for (chan = 0; chan < inlink->channels; chan++) {
	double src = (double )frame->extended_data[chan];
	double dst = (double )out_frame->extended_data[chan];
	double delayed_0, delayed_1;
	double delayed;
	double in, out;
	int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
	double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
	int int_delay = (int)delay;
	double frac_delay = modf(delay, &delay);
	double delay_buffer = (double )s->delay_buffer[chan];

	in = src[i];
	delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
	s->feedback_gain;
	delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
	delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];

	if (s->interpolation == INTERPOLATION_LINEAR) {
	delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
	} else {
	double a, b;
	double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
	delayed_2 -= delayed_0;
	delayed_1 -= delayed_0;
	a = delayed_2 * .5 - delayed_1;
	b = delayed_1 * 2 - delayed_2 *.5;
	delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
	}

	s->delay_last[chan] = delayed;
	out = in * s->in_gain + delayed * s->delay_gain;
	dst[i] = out;
	}
	s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
	}

	if (frame != out_frame)
	av_frame_free(&frame);

	return ff_filter_frame(ctx->outputs[0], out_frame);
	}

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

	av_freep(&s->lfo);
	av_freep(&s->delay_last);

	if (s->delay_buffer)
	av_freep(&s->delay_buffer[0]);
	av_freep(&s->delay_buffer);
	}

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

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

	AVFilter ff_af_flanger = {
	.name = "flanger",
	.description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
	.query_formats = query_formats,
	.priv_size = sizeof(FlangerContext),
	.priv_class = &flanger_class,
	.init = init,
	.uninit = uninit,
	.inputs = flanger_inputs,
	.outputs = flanger_outputs,
	};