[deb_ffmpeg.git] / ffmpeg / libavfilter / af_aphaser.c

/*
 * Copyright (c) 2013 Paul B Mahol
 *
 * 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
 * phaser audio filter
 */

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

typedef struct AudioPhaserContext {
    const AVClass *class;
    double in_gain, out_gain;
    double delay;
    double decay;
    double speed;

    enum WaveType type;

    int delay_buffer_length;
    double *delay_buffer;

    int modulation_buffer_length;
    int32_t *modulation_buffer;

    int delay_pos, modulation_pos;

    void (*phaser)(struct AudioPhaserContext *p,
                   uint8_t * const *src, uint8_t **dst,
                   int nb_samples, int channels);
} AudioPhaserContext;

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

static const AVOption aphaser_options[] = {
    { "in_gain",  "set input gain",            OFFSET(in_gain),  AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0,  1,   FLAGS },
    { "out_gain", "set output gain",           OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0,  1e9, FLAGS },
    { "delay",    "set delay in milliseconds", OFFSET(delay),    AV_OPT_TYPE_DOUBLE, {.dbl=3.},  0,  5,   FLAGS },
    { "decay",    "set decay",                 OFFSET(decay),    AV_OPT_TYPE_DOUBLE, {.dbl=.4},  0, .99,  FLAGS },
    { "speed",    "set modulation speed",      OFFSET(speed),    AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1,  2,   FLAGS },
    { "type",     "set modulation type",       OFFSET(type),     AV_OPT_TYPE_INT,    {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
    { "triangular",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
    { "t",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
    { "sinusoidal",  NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
    { "s",           NULL, 0, AV_OPT_TYPE_CONST,  {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
    { NULL }
};

AVFILTER_DEFINE_CLASS(aphaser);

static av_cold int init(AVFilterContext *ctx)
{
    AudioPhaserContext *p = ctx->priv;

    if (p->in_gain > (1 - p->decay * p->decay))
        av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
    if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
        av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");

    return 0;
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats;
    AVFilterChannelLayouts *layouts;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
        AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
        AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
        AV_SAMPLE_FMT_NONE
    };

    layouts = ff_all_channel_layouts();
    if (!layouts)
        return AVERROR(ENOMEM);
    ff_set_common_channel_layouts(ctx, layouts);

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

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

    return 0;
}

#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))

#define PHASER_PLANAR(name, type)                                      \
static void phaser_## name ##p(AudioPhaserContext *p,                  \
                               uint8_t * const *src, uint8_t **dst,    \
                               int nb_samples, int channels)           \
{                                                                      \
    int i, c, delay_pos, modulation_pos;                               \
                                                                       \
    av_assert0(channels > 0);                                          \
    for (c = 0; c < channels; c++) {                                   \
        type *s = (type *)src[c];                                      \
        type *d = (type *)dst[c];                                      \
        double *buffer = p->delay_buffer +                             \
                         c * p->delay_buffer_length;                   \
                                                                       \
        delay_pos      = p->delay_pos;                                 \
        modulation_pos = p->modulation_pos;                            \
                                                                       \
        for (i = 0; i < nb_samples; i++, s++, d++) {                   \
            double v = *s * p->in_gain + buffer[                       \
                       MOD(delay_pos + p->modulation_buffer[           \
                       modulation_pos],                                \
                       p->delay_buffer_length)] * p->decay;            \
                                                                       \
            modulation_pos = MOD(modulation_pos + 1,                   \
                             p->modulation_buffer_length);             \
            delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);    \
            buffer[delay_pos] = v;                                     \
                                                                       \
            *d = v * p->out_gain;                                      \
        }                                                              \
    }                                                                  \
                                                                       \
    p->delay_pos      = delay_pos;                                     \
    p->modulation_pos = modulation_pos;                                \
}

#define PHASER(name, type)                                              \
static void phaser_## name (AudioPhaserContext *p,                      \
                            uint8_t * const *src, uint8_t **dst,        \
                            int nb_samples, int channels)               \
{                                                                       \
    int i, c, delay_pos, modulation_pos;                                \
    type *s = (type *)src[0];                                           \
    type *d = (type *)dst[0];                                           \
    double *buffer = p->delay_buffer;                                   \
                                                                        \
    delay_pos      = p->delay_pos;                                      \
    modulation_pos = p->modulation_pos;                                 \
                                                                        \
    for (i = 0; i < nb_samples; i++) {                                  \
        int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
                   p->delay_buffer_length) * channels;                  \
        int npos;                                                       \
                                                                        \
        delay_pos = MOD(delay_pos + 1, p->delay_buffer_length);         \
        npos = delay_pos * channels;                                    \
        for (c = 0; c < channels; c++, s++, d++) {                      \
            double v = *s * p->in_gain + buffer[pos + c] * p->decay;    \
                                                                        \
            buffer[npos + c] = v;                                       \
                                                                        \
            *d = v * p->out_gain;                                       \
        }                                                               \
                                                                        \
        modulation_pos = MOD(modulation_pos + 1,                        \
                         p->modulation_buffer_length);                  \
    }                                                                   \
                                                                        \
    p->delay_pos      = delay_pos;                                      \
    p->modulation_pos = modulation_pos;                                 \
}

PHASER_PLANAR(dbl, double)
PHASER_PLANAR(flt, float)
PHASER_PLANAR(s16, int16_t)
PHASER_PLANAR(s32, int32_t)

PHASER(dbl, double)
PHASER(flt, float)
PHASER(s16, int16_t)
PHASER(s32, int32_t)

static int config_output(AVFilterLink *outlink)
{
    AudioPhaserContext *p = outlink->src->priv;
    AVFilterLink *inlink = outlink->src->inputs[0];

    p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
    p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
    p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
    p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));

    if (!p->modulation_buffer || !p->delay_buffer)
        return AVERROR(ENOMEM);

    ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
                           p->modulation_buffer, p->modulation_buffer_length,
                           1., p->delay_buffer_length, M_PI / 2.0);

    p->delay_pos = p->modulation_pos = 0;

    switch (inlink->format) {
    case AV_SAMPLE_FMT_DBL:  p->phaser = phaser_dbl;  break;
    case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
    case AV_SAMPLE_FMT_FLT:  p->phaser = phaser_flt;  break;
    case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
    case AV_SAMPLE_FMT_S16:  p->phaser = phaser_s16;  break;
    case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
    case AV_SAMPLE_FMT_S32:  p->phaser = phaser_s32;  break;
    case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
    default: av_assert0(0);
    }

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
{
    AudioPhaserContext *p = inlink->dst->priv;
    AVFilterLink *outlink = inlink->dst->outputs[0];
    AVFrame *outbuf;

    if (av_frame_is_writable(inbuf)) {
        outbuf = inbuf;
    } else {
        outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
        if (!outbuf)
            return AVERROR(ENOMEM);
        av_frame_copy_props(outbuf, inbuf);
    }

    p->phaser(p, inbuf->extended_data, outbuf->extended_data,
              outbuf->nb_samples, av_frame_get_channels(outbuf));

    if (inbuf != outbuf)
        av_frame_free(&inbuf);

    return ff_filter_frame(outlink, outbuf);
}

static av_cold void uninit(AVFilterContext *ctx)
{
    AudioPhaserContext *p = ctx->priv;

    av_freep(&p->delay_buffer);
    av_freep(&p->modulation_buffer);
}

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

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

AVFilter ff_af_aphaser = {
    .name          = "aphaser",
    .description   = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
    .query_formats = query_formats,
    .priv_size     = sizeof(AudioPhaserContext),
    .init          = init,
    .uninit        = uninit,
    .inputs        = aphaser_inputs,
    .outputs       = aphaser_outputs,
    .priv_class    = &aphaser_class,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* Copyright (c) 2013 Paul B Mahol
	3	*
	4	* This file is part of FFmpeg.
	5	*
	6	* FFmpeg is free software; you can redistribute it and/or
	7	* modify it under the terms of the GNU Lesser General Public
	8	* License as published by the Free Software Foundation; either
	9	* version 2.1 of the License, or (at your option) any later version.
	10	*
	11	* FFmpeg is distributed in the hope that it will be useful,
	12	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	14	* Lesser General Public License for more details.
	15	*
	16	* You should have received a copy of the GNU Lesser General Public
	17	* License along with FFmpeg; if not, write to the Free Software
	18	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	19	*/
	20
	21	/**
	22	* @file
	23	* phaser audio filter
	24	*/
	25
	26	#include "libavutil/avassert.h"
	27	#include "libavutil/opt.h"
	28	#include "audio.h"
	29	#include "avfilter.h"
	30	#include "internal.h"
	31	#include "generate_wave_table.h"
	32
	33	typedef struct AudioPhaserContext {
	34	const AVClass *class;
	35	double in_gain, out_gain;
	36	double delay;
	37	double decay;
	38	double speed;
	39
	40	enum WaveType type;
	41
	42	int delay_buffer_length;
	43	double *delay_buffer;
	44
	45	int modulation_buffer_length;
	46	int32_t *modulation_buffer;
	47
	48	int delay_pos, modulation_pos;
	49
	50	void (phaser)(struct AudioPhaserContext p,
	51	uint8_t * const src, uint8_t *dst,
	52	int nb_samples, int channels);
	53	} AudioPhaserContext;
	54
	55	#define OFFSET(x) offsetof(AudioPhaserContext, x)
	56	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
	57
	58	static const AVOption aphaser_options[] = {
	59	{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
	60	{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
	61	{ "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
	62	{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
	63	{ "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
	64	{ "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
65	{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
66	{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
67	{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
68	{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
69	{ NULL }
70	};
71
72	AVFILTER_DEFINE_CLASS(aphaser);
73
74	static av_cold int init(AVFilterContext *ctx)
75	{
76	AudioPhaserContext *p = ctx->priv;
77
78	if (p->in_gain > (1 - p->decay * p->decay))
79	av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
80	if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
81	av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
82
83	return 0;
84	}
85
86	static int query_formats(AVFilterContext *ctx)
87	{
88	AVFilterFormats *formats;
89	AVFilterChannelLayouts *layouts;
90	static const enum AVSampleFormat sample_fmts[] = {
91	AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
92	AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
93	AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
94	AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
95	AV_SAMPLE_FMT_NONE
96	};
97
98	layouts = ff_all_channel_layouts();
99	if (!layouts)
100	return AVERROR(ENOMEM);
101	ff_set_common_channel_layouts(ctx, layouts);
102
103	formats = ff_make_format_list(sample_fmts);
104	if (!formats)
105	return AVERROR(ENOMEM);
106	ff_set_common_formats(ctx, formats);
107
108	formats = ff_all_samplerates();
109	if (!formats)
110	return AVERROR(ENOMEM);
111	ff_set_common_samplerates(ctx, formats);
112
113	return 0;
114	}
115
116	#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
117
118	#define PHASER_PLANAR(name, type) \
119	static void phaser_## name ##p(AudioPhaserContext *p, \
120	uint8_t * const src, uint8_t *dst, \
121	int nb_samples, int channels) \
122	{ \
123	int i, c, delay_pos, modulation_pos; \
124	\
125	av_assert0(channels > 0); \
126	for (c = 0; c < channels; c++) { \
127	type s = (type )src[c]; \
128	type d = (type )dst[c]; \
129	double *buffer = p->delay_buffer + \
130	c * p->delay_buffer_length; \
131	\
132	delay_pos = p->delay_pos; \
133	modulation_pos = p->modulation_pos; \
134	\
135	for (i = 0; i < nb_samples; i++, s++, d++) { \
136	double v = s p->in_gain + buffer[ \
137	MOD(delay_pos + p->modulation_buffer[ \
138	modulation_pos], \
139	p->delay_buffer_length)] * p->decay; \
140	\
141	modulation_pos = MOD(modulation_pos + 1, \
142	p->modulation_buffer_length); \
143	delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
144	buffer[delay_pos] = v; \
145	\
146	d = v p->out_gain; \
147	} \
148	} \
149	\
150	p->delay_pos = delay_pos; \
151	p->modulation_pos = modulation_pos; \
152	}
153
154	#define PHASER(name, type) \
155	static void phaser_## name (AudioPhaserContext *p, \
156	uint8_t * const src, uint8_t *dst, \
157	int nb_samples, int channels) \
158	{ \
159	int i, c, delay_pos, modulation_pos; \
160	type s = (type )src[0]; \
161	type d = (type )dst[0]; \
162	double *buffer = p->delay_buffer; \
163	\
164	delay_pos = p->delay_pos; \
165	modulation_pos = p->modulation_pos; \
166	\
167	for (i = 0; i < nb_samples; i++) { \
168	int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
169	p->delay_buffer_length) * channels; \
170	int npos; \
171	\
172	delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
173	npos = delay_pos * channels; \
174	for (c = 0; c < channels; c++, s++, d++) { \
175	double v = s p->in_gain + buffer[pos + c] * p->decay; \
176	\
177	buffer[npos + c] = v; \
178	\
179	d = v p->out_gain; \
180	} \
181	\
182	modulation_pos = MOD(modulation_pos + 1, \
183	p->modulation_buffer_length); \
184	} \
185	\
186	p->delay_pos = delay_pos; \
187	p->modulation_pos = modulation_pos; \
188	}
189
190	PHASER_PLANAR(dbl, double)
191	PHASER_PLANAR(flt, float)
192	PHASER_PLANAR(s16, int16_t)
193	PHASER_PLANAR(s32, int32_t)
194
195	PHASER(dbl, double)
196	PHASER(flt, float)
197	PHASER(s16, int16_t)
198	PHASER(s32, int32_t)
199
200	static int config_output(AVFilterLink *outlink)
201	{
202	AudioPhaserContext *p = outlink->src->priv;
203	AVFilterLink *inlink = outlink->src->inputs[0];
204
205	p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
206	p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(p->delay_buffer) inlink->channels);
207	p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
208	p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
209
210	if (!p->modulation_buffer \|\| !p->delay_buffer)
211	return AVERROR(ENOMEM);
212
213	ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
214	p->modulation_buffer, p->modulation_buffer_length,
215	1., p->delay_buffer_length, M_PI / 2.0);
216
217	p->delay_pos = p->modulation_pos = 0;
218
219	switch (inlink->format) {
220	case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
221	case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
222	case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
223	case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
224	case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
225	case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
226	case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
227	case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
228	default: av_assert0(0);
229	}
230
231	return 0;
232	}
233
234	static int filter_frame(AVFilterLink inlink, AVFrame inbuf)
235	{
236	AudioPhaserContext *p = inlink->dst->priv;
237	AVFilterLink *outlink = inlink->dst->outputs[0];
238	AVFrame *outbuf;
239
240	if (av_frame_is_writable(inbuf)) {
241	outbuf = inbuf;
242	} else {
243	outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
244	if (!outbuf)
245	return AVERROR(ENOMEM);
246	av_frame_copy_props(outbuf, inbuf);
247	}
248
249	p->phaser(p, inbuf->extended_data, outbuf->extended_data,
250	outbuf->nb_samples, av_frame_get_channels(outbuf));
251
252	if (inbuf != outbuf)
253	av_frame_free(&inbuf);
254
255	return ff_filter_frame(outlink, outbuf);
256	}
257
258	static av_cold void uninit(AVFilterContext *ctx)
259	{
260	AudioPhaserContext *p = ctx->priv;
261
262	av_freep(&p->delay_buffer);
263	av_freep(&p->modulation_buffer);
264	}
265
266	static const AVFilterPad aphaser_inputs[] = {
267	{
268	.name = "default",
269	.type = AVMEDIA_TYPE_AUDIO,
270	.filter_frame = filter_frame,
271	},
272	{ NULL }
273	};
274
275	static const AVFilterPad aphaser_outputs[] = {
276	{
277	.name = "default",
278	.type = AVMEDIA_TYPE_AUDIO,
279	.config_props = config_output,
280	},
281	{ NULL }
282	};
283
284	AVFilter ff_af_aphaser = {
285	.name = "aphaser",
286	.description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
287	.query_formats = query_formats,
288	.priv_size = sizeof(AudioPhaserContext),
289	.init = init,
290	.uninit = uninit,
291	.inputs = aphaser_inputs,
292	.outputs = aphaser_outputs,
293	.priv_class = &aphaser_class,
294	};