[deb_ffmpeg.git] / ffmpeg / libavfilter / af_adelay.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
 */

#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"

typedef struct ChanDelay {
    int delay;
    unsigned delay_index;
    unsigned index;
    uint8_t *samples;
} ChanDelay;

typedef struct AudioDelayContext {
    const AVClass *class;
    char *delays;
    ChanDelay *chandelay;
    int nb_delays;
    int block_align;
    unsigned max_delay;
    int64_t next_pts;

    void (*delay_channel)(ChanDelay *d, int nb_samples,
                          const uint8_t *src, uint8_t *dst);
} AudioDelayContext;

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

static const AVOption adelay_options[] = {
    { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(adelay);

static int query_formats(AVFilterContext *ctx)
{
    AVFilterChannelLayouts *layouts;
    AVFilterFormats *formats;
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
        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 DELAY(name, type, fill)                                           \
static void delay_channel_## name ##p(ChanDelay *d, int nb_samples,       \
                                      const uint8_t *ssrc, uint8_t *ddst) \
{                                                                         \
    const type *src = (type *)ssrc;                                       \
    type *dst = (type *)ddst;                                             \
    type *samples = (type *)d->samples;                                   \
                                                                          \
    while (nb_samples) {                                                  \
        if (d->delay_index < d->delay) {                                  \
            const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
                                                                          \
            memcpy(&samples[d->delay_index], src, len * sizeof(type));    \
            memset(dst, fill, len * sizeof(type));                        \
            d->delay_index += len;                                        \
            src += len;                                                   \
            dst += len;                                                   \
            nb_samples -= len;                                            \
        } else {                                                          \
            *dst = samples[d->index];                                     \
            samples[d->index] = *src;                                     \
            nb_samples--;                                                 \
            d->index++;                                                   \
            src++, dst++;                                                 \
            d->index = d->index >= d->delay ? 0 : d->index;               \
        }                                                                 \
    }                                                                     \
}

DELAY(u8,  uint8_t, 0x80)
DELAY(s16, int16_t, 0)
DELAY(s32, int32_t, 0)
DELAY(flt, float,   0)
DELAY(dbl, double,  0)

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    AudioDelayContext *s = ctx->priv;
    char *p, *arg, *saveptr = NULL;
    int i;

    s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
    if (!s->chandelay)
        return AVERROR(ENOMEM);
    s->nb_delays = inlink->channels;
    s->block_align = av_get_bytes_per_sample(inlink->format);

    p = s->delays;
    for (i = 0; i < s->nb_delays; i++) {
        ChanDelay *d = &s->chandelay[i];
        float delay;

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

        p = NULL;
        sscanf(arg, "%f", &delay);

        d->delay = delay * inlink->sample_rate / 1000.0;
        if (d->delay < 0) {
            av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
            return AVERROR(EINVAL);
        }
    }

    for (i = 0; i < s->nb_delays; i++) {
        ChanDelay *d = &s->chandelay[i];

        if (!d->delay)
            continue;

        d->samples = av_malloc_array(d->delay, s->block_align);
        if (!d->samples)
            return AVERROR(ENOMEM);

        s->max_delay = FFMAX(s->max_delay, d->delay);
    }

    if (!s->max_delay) {
        av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");
        return AVERROR(EINVAL);
    }

    switch (inlink->format) {
    case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
    case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
    case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
    case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
    case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
    }

    return 0;
}

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

    if (ctx->is_disabled || !s->delays)
        return ff_filter_frame(ctx->outputs[0], frame);

    out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
    if (!out_frame)
        return AVERROR(ENOMEM);
    av_frame_copy_props(out_frame, frame);

    for (i = 0; i < s->nb_delays; i++) {
        ChanDelay *d = &s->chandelay[i];
        const uint8_t *src = frame->extended_data[i];
        uint8_t *dst = out_frame->extended_data[i];

        if (!d->delay)
            memcpy(dst, src, frame->nb_samples * s->block_align);
        else
            s->delay_channel(d, frame->nb_samples, src, dst);
    }

    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
    av_frame_free(&frame);
    return ff_filter_frame(ctx->outputs[0], out_frame);
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AudioDelayContext *s = ctx->priv;
    int ret;

    ret = ff_request_frame(ctx->inputs[0]);
    if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
        int nb_samples = FFMIN(s->max_delay, 2048);
        AVFrame *frame;

        frame = ff_get_audio_buffer(outlink, nb_samples);
        if (!frame)
            return AVERROR(ENOMEM);
        s->max_delay -= nb_samples;

        av_samples_set_silence(frame->extended_data, 0,
                               frame->nb_samples,
                               outlink->channels,
                               frame->format);

        frame->pts = s->next_pts;
        if (s->next_pts != AV_NOPTS_VALUE)
            s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

        ret = filter_frame(ctx->inputs[0], frame);
    }

    return ret;
}

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

    for (i = 0; i < s->nb_delays; i++)
        av_free(s->chandelay[i].samples);
    av_freep(&s->chandelay);
}

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

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

AVFilter ff_af_adelay = {
    .name          = "adelay",
    .description   = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
    .query_formats = query_formats,
    .priv_size     = sizeof(AudioDelayContext),
    .priv_class    = &adelay_class,
    .uninit        = uninit,
    .inputs        = adelay_inputs,
    .outputs       = adelay_outputs,
    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
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	#include "libavutil/avstring.h"
	22	#include "libavutil/opt.h"
	23	#include "libavutil/samplefmt.h"
	24	#include "avfilter.h"
	25	#include "audio.h"
	26	#include "internal.h"
	27
	28	typedef struct ChanDelay {
	29	int delay;
	30	unsigned delay_index;
	31	unsigned index;
	32	uint8_t *samples;
	33	} ChanDelay;
	34
	35	typedef struct AudioDelayContext {
	36	const AVClass *class;
	37	char *delays;
	38	ChanDelay *chandelay;
	39	int nb_delays;
	40	int block_align;
	41	unsigned max_delay;
	42	int64_t next_pts;
	43
	44	void (delay_channel)(ChanDelay d, int nb_samples,
	45	const uint8_t src, uint8_t dst);
	46	} AudioDelayContext;
	47
	48	#define OFFSET(x) offsetof(AudioDelayContext, x)
	49	#define A AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
	50
	51	static const AVOption adelay_options[] = {
	52	{ "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
	53	{ NULL }
	54	};
	55
	56	AVFILTER_DEFINE_CLASS(adelay);
	57
	58	static int query_formats(AVFilterContext *ctx)
	59	{
	60	AVFilterChannelLayouts *layouts;
	61	AVFilterFormats *formats;
	62	static const enum AVSampleFormat sample_fmts[] = {
	63	AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
	64	AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
65	AV_SAMPLE_FMT_NONE
66	};
67
68	layouts = ff_all_channel_layouts();
69	if (!layouts)
70	return AVERROR(ENOMEM);
71	ff_set_common_channel_layouts(ctx, layouts);
72
73	formats = ff_make_format_list(sample_fmts);
74	if (!formats)
75	return AVERROR(ENOMEM);
76	ff_set_common_formats(ctx, formats);
77
78	formats = ff_all_samplerates();
79	if (!formats)
80	return AVERROR(ENOMEM);
81	ff_set_common_samplerates(ctx, formats);
82
83	return 0;
84	}
85
86	#define DELAY(name, type, fill) \
87	static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
88	const uint8_t ssrc, uint8_t ddst) \
89	{ \
90	const type src = (type )ssrc; \
91	type dst = (type )ddst; \
92	type samples = (type )d->samples; \
93	\
94	while (nb_samples) { \
95	if (d->delay_index < d->delay) { \
96	const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
97	\
98	memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
99	memset(dst, fill, len * sizeof(type)); \
100	d->delay_index += len; \
101	src += len; \
102	dst += len; \
103	nb_samples -= len; \
104	} else { \
105	*dst = samples[d->index]; \
106	samples[d->index] = *src; \
107	nb_samples--; \
108	d->index++; \
109	src++, dst++; \
110	d->index = d->index >= d->delay ? 0 : d->index; \
111	} \
112	} \
113	}
114
115	DELAY(u8, uint8_t, 0x80)
116	DELAY(s16, int16_t, 0)
117	DELAY(s32, int32_t, 0)
118	DELAY(flt, float, 0)
119	DELAY(dbl, double, 0)
120
121	static int config_input(AVFilterLink *inlink)
122	{
123	AVFilterContext *ctx = inlink->dst;
124	AudioDelayContext *s = ctx->priv;
125	char p, arg, *saveptr = NULL;
126	int i;
127
128	s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
129	if (!s->chandelay)
130	return AVERROR(ENOMEM);
131	s->nb_delays = inlink->channels;
132	s->block_align = av_get_bytes_per_sample(inlink->format);
133
134	p = s->delays;
135	for (i = 0; i < s->nb_delays; i++) {
136	ChanDelay *d = &s->chandelay[i];
137	float delay;
138
139	if (!(arg = av_strtok(p, "\|", &saveptr)))
140	break;
141
142	p = NULL;
143	sscanf(arg, "%f", &delay);
144
145	d->delay = delay * inlink->sample_rate / 1000.0;
146	if (d->delay < 0) {
147	av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
148	return AVERROR(EINVAL);
149	}
150	}
151
152	for (i = 0; i < s->nb_delays; i++) {
153	ChanDelay *d = &s->chandelay[i];
154
155	if (!d->delay)
156	continue;
157
158	d->samples = av_malloc_array(d->delay, s->block_align);
159	if (!d->samples)
160	return AVERROR(ENOMEM);
161
162	s->max_delay = FFMAX(s->max_delay, d->delay);
163	}
164
165	if (!s->max_delay) {
166	av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");
167	return AVERROR(EINVAL);
168	}
169
170	switch (inlink->format) {
171	case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
172	case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
173	case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
174	case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
175	case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
176	}
177
178	return 0;
179	}
180
181	static int filter_frame(AVFilterLink inlink, AVFrame frame)
182	{
183	AVFilterContext *ctx = inlink->dst;
184	AudioDelayContext *s = ctx->priv;
185	AVFrame *out_frame;
186	int i;
187
188	if (ctx->is_disabled \|\| !s->delays)
189	return ff_filter_frame(ctx->outputs[0], frame);
190
191	out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
192	if (!out_frame)
193	return AVERROR(ENOMEM);
194	av_frame_copy_props(out_frame, frame);
195
196	for (i = 0; i < s->nb_delays; i++) {
197	ChanDelay *d = &s->chandelay[i];
198	const uint8_t *src = frame->extended_data[i];
199	uint8_t *dst = out_frame->extended_data[i];
200
201	if (!d->delay)
202	memcpy(dst, src, frame->nb_samples * s->block_align);
203	else
204	s->delay_channel(d, frame->nb_samples, src, dst);
205	}
206
207	s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
208	av_frame_free(&frame);
209	return ff_filter_frame(ctx->outputs[0], out_frame);
210	}
211
212	static int request_frame(AVFilterLink *outlink)
213	{
214	AVFilterContext *ctx = outlink->src;
215	AudioDelayContext *s = ctx->priv;
216	int ret;
217
218	ret = ff_request_frame(ctx->inputs[0]);
219	if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
220	int nb_samples = FFMIN(s->max_delay, 2048);
221	AVFrame *frame;
222
223	frame = ff_get_audio_buffer(outlink, nb_samples);
224	if (!frame)
225	return AVERROR(ENOMEM);
226	s->max_delay -= nb_samples;
227
228	av_samples_set_silence(frame->extended_data, 0,
229	frame->nb_samples,
230	outlink->channels,
231	frame->format);
232
233	frame->pts = s->next_pts;
234	if (s->next_pts != AV_NOPTS_VALUE)
235	s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
236
237	ret = filter_frame(ctx->inputs[0], frame);
238	}
239
240	return ret;
241	}
242
243	static av_cold void uninit(AVFilterContext *ctx)
244	{
245	AudioDelayContext *s = ctx->priv;
246	int i;
247
248	for (i = 0; i < s->nb_delays; i++)
249	av_free(s->chandelay[i].samples);
250	av_freep(&s->chandelay);
251	}
252
253	static const AVFilterPad adelay_inputs[] = {
254	{
255	.name = "default",
256	.type = AVMEDIA_TYPE_AUDIO,
257	.config_props = config_input,
258	.filter_frame = filter_frame,
259	},
260	{ NULL }
261	};
262
263	static const AVFilterPad adelay_outputs[] = {
264	{
265	.name = "default",
266	.request_frame = request_frame,
267	.type = AVMEDIA_TYPE_AUDIO,
268	},
269	{ NULL }
270	};
271
272	AVFilter ff_af_adelay = {
273	.name = "adelay",
274	.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
275	.query_formats = query_formats,
276	.priv_size = sizeof(AudioDelayContext),
277	.priv_class = &adelay_class,
278	.uninit = uninit,
279	.inputs = adelay_inputs,
280	.outputs = adelay_outputs,
281	.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
282	};