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

/*
 * Copyright (c) 2011 Stefano Sabatini
 * Copyright (c) 2011 Mina Nagy Zaki
 *
 * 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
 * resampling audio filter
 */

#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "libswresample/swresample.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"

typedef struct {
    const AVClass *class;
    int sample_rate_arg;
    double ratio;
    struct SwrContext *swr;
    int64_t next_pts;
    int req_fullfilled;
} AResampleContext;

static av_cold int init_dict(AVFilterContext *ctx, AVDictionary **opts)
{
    AResampleContext *aresample = ctx->priv;
    int ret = 0;

    aresample->next_pts = AV_NOPTS_VALUE;
    aresample->swr = swr_alloc();
    if (!aresample->swr) {
        ret = AVERROR(ENOMEM);
        goto end;
    }

    if (opts) {
        AVDictionaryEntry *e = NULL;

        while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
            if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
                goto end;
        }
        av_dict_free(opts);
    }
    if (aresample->sample_rate_arg > 0)
        av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
end:
    return ret;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    AResampleContext *aresample = ctx->priv;
    swr_free(&aresample->swr);
}

static int query_formats(AVFilterContext *ctx)
{
    AResampleContext *aresample = ctx->priv;
    int out_rate                   = av_get_int(aresample->swr, "osr", NULL);
    uint64_t out_layout            = av_get_int(aresample->swr, "ocl", NULL);
    enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);

    AVFilterLink *inlink  = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];

    AVFilterFormats        *in_formats      = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    AVFilterFormats        *out_formats;
    AVFilterFormats        *in_samplerates  = ff_all_samplerates();
    AVFilterFormats        *out_samplerates;
    AVFilterChannelLayouts *in_layouts      = ff_all_channel_counts();
    AVFilterChannelLayouts *out_layouts;

    ff_formats_ref  (in_formats,      &inlink->out_formats);
    ff_formats_ref  (in_samplerates,  &inlink->out_samplerates);
    ff_channel_layouts_ref(in_layouts,      &inlink->out_channel_layouts);

    if(out_rate > 0) {
        int ratelist[] = { out_rate, -1 };
        out_samplerates = ff_make_format_list(ratelist);
    } else {
        out_samplerates = ff_all_samplerates();
    }
    ff_formats_ref(out_samplerates, &outlink->in_samplerates);

    if(out_format != AV_SAMPLE_FMT_NONE) {
        int formatlist[] = { out_format, -1 };
        out_formats = ff_make_format_list(formatlist);
    } else
        out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
    ff_formats_ref(out_formats, &outlink->in_formats);

    if(out_layout) {
        int64_t layout_list[] = { out_layout, -1 };
        out_layouts = avfilter_make_format64_list(layout_list);
    } else
        out_layouts = ff_all_channel_counts();
    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

    return 0;
}


static int config_output(AVFilterLink *outlink)
{
    int ret;
    AVFilterContext *ctx = outlink->src;
    AVFilterLink *inlink = ctx->inputs[0];
    AResampleContext *aresample = ctx->priv;
    int out_rate;
    uint64_t out_layout;
    enum AVSampleFormat out_format;
    char inchl_buf[128], outchl_buf[128];

    aresample->swr = swr_alloc_set_opts(aresample->swr,
                                        outlink->channel_layout, outlink->format, outlink->sample_rate,
                                        inlink->channel_layout, inlink->format, inlink->sample_rate,
                                        0, ctx);
    if (!aresample->swr)
        return AVERROR(ENOMEM);
    if (!inlink->channel_layout)
        av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
    if (!outlink->channel_layout)
        av_opt_set_int(aresample->swr, "och", outlink->channels, 0);

    ret = swr_init(aresample->swr);
    if (ret < 0)
        return ret;

    out_rate   = av_get_int(aresample->swr, "osr", NULL);
    out_layout = av_get_int(aresample->swr, "ocl", NULL);
    out_format = av_get_int(aresample->swr, "osf", NULL);
    outlink->time_base = (AVRational) {1, out_rate};

    av_assert0(outlink->sample_rate == out_rate);
    av_assert0(outlink->channel_layout == out_layout || !outlink->channel_layout);
    av_assert0(outlink->format == out_format);

    aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;

    av_get_channel_layout_string(inchl_buf,  sizeof(inchl_buf),  inlink ->channels, inlink ->channel_layout);
    av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);

    av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
           inlink ->channels, inchl_buf,  av_get_sample_fmt_name(inlink->format),  inlink->sample_rate,
           outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *insamplesref)
{
    AResampleContext *aresample = inlink->dst->priv;
    const int n_in  = insamplesref->nb_samples;
    int64_t delay;
    int n_out       = n_in * aresample->ratio + 32;
    AVFilterLink *const outlink = inlink->dst->outputs[0];
    AVFrame *outsamplesref;
    int ret;

    delay = swr_get_delay(aresample->swr, outlink->sample_rate);
    if (delay > 0)
        n_out += delay;

    outsamplesref = ff_get_audio_buffer(outlink, n_out);

    if(!outsamplesref)
        return AVERROR(ENOMEM);

    av_frame_copy_props(outsamplesref, insamplesref);
    outsamplesref->format                = outlink->format;
    av_frame_set_channels(outsamplesref, outlink->channels);
    outsamplesref->channel_layout        = outlink->channel_layout;
    outsamplesref->sample_rate           = outlink->sample_rate;

    if(insamplesref->pts != AV_NOPTS_VALUE) {
        int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
        int64_t outpts= swr_next_pts(aresample->swr, inpts);
        aresample->next_pts =
        outsamplesref->pts  = ROUNDED_DIV(outpts, inlink->sample_rate);
    } else {
        outsamplesref->pts  = AV_NOPTS_VALUE;
    }
    n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
                                 (void *)insamplesref->extended_data, n_in);
    if (n_out <= 0) {
        av_frame_free(&outsamplesref);
        av_frame_free(&insamplesref);
        return 0;
    }

    outsamplesref->nb_samples  = n_out;

    ret = ff_filter_frame(outlink, outsamplesref);
    aresample->req_fullfilled= 1;
    av_frame_free(&insamplesref);
    return ret;
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AResampleContext *aresample = ctx->priv;
    AVFilterLink *const inlink = outlink->src->inputs[0];
    int ret;

    aresample->req_fullfilled = 0;
    do{
        ret = ff_request_frame(ctx->inputs[0]);
    }while(!aresample->req_fullfilled && ret>=0);

    if (ret == AVERROR_EOF) {
        AVFrame *outsamplesref;
        int n_out = 4096;
        int64_t pts;

        outsamplesref = ff_get_audio_buffer(outlink, n_out);
        if (!outsamplesref)
            return AVERROR(ENOMEM);

        pts = swr_next_pts(aresample->swr, INT64_MIN);
        pts = ROUNDED_DIV(pts, inlink->sample_rate);

        n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
        if (n_out <= 0) {
            av_frame_free(&outsamplesref);
            return (n_out == 0) ? AVERROR_EOF : n_out;
        }

        outsamplesref->sample_rate = outlink->sample_rate;
        outsamplesref->nb_samples  = n_out;

        outsamplesref->pts = pts;

        return ff_filter_frame(outlink, outsamplesref);
    }
    return ret;
}

static const AVClass *resample_child_class_next(const AVClass *prev)
{
    return prev ? NULL : swr_get_class();
}

static void *resample_child_next(void *obj, void *prev)
{
    AResampleContext *s = obj;
    return prev ? NULL : s->swr;
}

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

static const AVOption options[] = {
    {"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0},  0,        INT_MAX, FLAGS },
    {NULL}
};

static const AVClass aresample_class = {
    .class_name       = "aresample",
    .item_name        = av_default_item_name,
    .option           = options,
    .version          = LIBAVUTIL_VERSION_INT,
    .child_class_next = resample_child_class_next,
    .child_next       = resample_child_next,
};

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

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

AVFilter ff_af_aresample = {
    .name          = "aresample",
    .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
    .init_dict     = init_dict,
    .uninit        = uninit,
    .query_formats = query_formats,
    .priv_size     = sizeof(AResampleContext),
    .priv_class    = &aresample_class,
    .inputs        = aresample_inputs,
    .outputs       = aresample_outputs,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* Copyright (c) 2011 Stefano Sabatini
	3	* Copyright (c) 2011 Mina Nagy Zaki
	4	*
	5	* This file is part of FFmpeg.
	6	*
	7	* FFmpeg is free software; you can redistribute it and/or
	8	* modify it under the terms of the GNU Lesser General Public
	9	* License as published by the Free Software Foundation; either
	10	* version 2.1 of the License, or (at your option) any later version.
	11	*
	12	* FFmpeg is distributed in the hope that it will be useful,
	13	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	14	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	15	* Lesser General Public License for more details.
	16	*
	17	* You should have received a copy of the GNU Lesser General Public
	18	* License along with FFmpeg; if not, write to the Free Software
	19	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	20	*/
	21
	22	/**
	23	* @file
	24	* resampling audio filter
	25	*/
	26
	27	#include "libavutil/avstring.h"
	28	#include "libavutil/channel_layout.h"
	29	#include "libavutil/opt.h"
	30	#include "libavutil/samplefmt.h"
	31	#include "libavutil/avassert.h"
	32	#include "libswresample/swresample.h"
	33	#include "avfilter.h"
	34	#include "audio.h"
	35	#include "internal.h"
	36
	37	typedef struct {
	38	const AVClass *class;
	39	int sample_rate_arg;
	40	double ratio;
	41	struct SwrContext *swr;
	42	int64_t next_pts;
	43	int req_fullfilled;
	44	} AResampleContext;
	45
	46	static av_cold int init_dict(AVFilterContext ctx, AVDictionary *opts)
	47	{
	48	AResampleContext *aresample = ctx->priv;
	49	int ret = 0;
	50
	51	aresample->next_pts = AV_NOPTS_VALUE;
	52	aresample->swr = swr_alloc();
	53	if (!aresample->swr) {
	54	ret = AVERROR(ENOMEM);
	55	goto end;
	56	}
	57
	58	if (opts) {
	59	AVDictionaryEntry *e = NULL;
	60
	61	while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
	62	if ((ret = av_opt_set(aresample->swr, e->key, e->value, 0)) < 0)
	63	goto end;
	64	}
65	av_dict_free(opts);
66	}
67	if (aresample->sample_rate_arg > 0)
68	av_opt_set_int(aresample->swr, "osr", aresample->sample_rate_arg, 0);
69	end:
70	return ret;
71	}
72
73	static av_cold void uninit(AVFilterContext *ctx)
74	{
75	AResampleContext *aresample = ctx->priv;
76	swr_free(&aresample->swr);
77	}
78
79	static int query_formats(AVFilterContext *ctx)
80	{
81	AResampleContext *aresample = ctx->priv;
82	int out_rate = av_get_int(aresample->swr, "osr", NULL);
83	uint64_t out_layout = av_get_int(aresample->swr, "ocl", NULL);
84	enum AVSampleFormat out_format = av_get_int(aresample->swr, "osf", NULL);
85
86	AVFilterLink *inlink = ctx->inputs[0];
87	AVFilterLink *outlink = ctx->outputs[0];
88
89	AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
90	AVFilterFormats *out_formats;
91	AVFilterFormats *in_samplerates = ff_all_samplerates();
92	AVFilterFormats *out_samplerates;
93	AVFilterChannelLayouts *in_layouts = ff_all_channel_counts();
94	AVFilterChannelLayouts *out_layouts;
95
96	ff_formats_ref (in_formats, &inlink->out_formats);
97	ff_formats_ref (in_samplerates, &inlink->out_samplerates);
98	ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
99
100	if(out_rate > 0) {
101	int ratelist[] = { out_rate, -1 };
102	out_samplerates = ff_make_format_list(ratelist);
103	} else {
104	out_samplerates = ff_all_samplerates();
105	}
106	ff_formats_ref(out_samplerates, &outlink->in_samplerates);
107
108	if(out_format != AV_SAMPLE_FMT_NONE) {
109	int formatlist[] = { out_format, -1 };
110	out_formats = ff_make_format_list(formatlist);
111	} else
112	out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
113	ff_formats_ref(out_formats, &outlink->in_formats);
114
115	if(out_layout) {
116	int64_t layout_list[] = { out_layout, -1 };
117	out_layouts = avfilter_make_format64_list(layout_list);
118	} else
119	out_layouts = ff_all_channel_counts();
120	ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
121
122	return 0;
123	}
124
125
126	static int config_output(AVFilterLink *outlink)
127	{
128	int ret;
129	AVFilterContext *ctx = outlink->src;
130	AVFilterLink *inlink = ctx->inputs[0];
131	AResampleContext *aresample = ctx->priv;
132	int out_rate;
133	uint64_t out_layout;
134	enum AVSampleFormat out_format;
135	char inchl_buf[128], outchl_buf[128];
136
137	aresample->swr = swr_alloc_set_opts(aresample->swr,
138	outlink->channel_layout, outlink->format, outlink->sample_rate,
139	inlink->channel_layout, inlink->format, inlink->sample_rate,
140	0, ctx);
141	if (!aresample->swr)
142	return AVERROR(ENOMEM);
143	if (!inlink->channel_layout)
144	av_opt_set_int(aresample->swr, "ich", inlink->channels, 0);
145	if (!outlink->channel_layout)
146	av_opt_set_int(aresample->swr, "och", outlink->channels, 0);
147
148	ret = swr_init(aresample->swr);
149	if (ret < 0)
150	return ret;
151
152	out_rate = av_get_int(aresample->swr, "osr", NULL);
153	out_layout = av_get_int(aresample->swr, "ocl", NULL);
154	out_format = av_get_int(aresample->swr, "osf", NULL);
155	outlink->time_base = (AVRational) {1, out_rate};
156
157	av_assert0(outlink->sample_rate == out_rate);
158	av_assert0(outlink->channel_layout == out_layout \|\| !outlink->channel_layout);
159	av_assert0(outlink->format == out_format);
160
161	aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
162
163	av_get_channel_layout_string(inchl_buf, sizeof(inchl_buf), inlink ->channels, inlink ->channel_layout);
164	av_get_channel_layout_string(outchl_buf, sizeof(outchl_buf), outlink->channels, outlink->channel_layout);
165
166	av_log(ctx, AV_LOG_VERBOSE, "ch:%d chl:%s fmt:%s r:%dHz -> ch:%d chl:%s fmt:%s r:%dHz\n",
167	inlink ->channels, inchl_buf, av_get_sample_fmt_name(inlink->format), inlink->sample_rate,
168	outlink->channels, outchl_buf, av_get_sample_fmt_name(outlink->format), outlink->sample_rate);
169	return 0;
170	}
171
172	static int filter_frame(AVFilterLink inlink, AVFrame insamplesref)
173	{
174	AResampleContext *aresample = inlink->dst->priv;
175	const int n_in = insamplesref->nb_samples;
176	int64_t delay;
177	int n_out = n_in * aresample->ratio + 32;
178	AVFilterLink *const outlink = inlink->dst->outputs[0];
179	AVFrame *outsamplesref;
180	int ret;
181
182	delay = swr_get_delay(aresample->swr, outlink->sample_rate);
183	if (delay > 0)
184	n_out += delay;
185
186	outsamplesref = ff_get_audio_buffer(outlink, n_out);
187
188	if(!outsamplesref)
189	return AVERROR(ENOMEM);
190
191	av_frame_copy_props(outsamplesref, insamplesref);
192	outsamplesref->format = outlink->format;
193	av_frame_set_channels(outsamplesref, outlink->channels);
194	outsamplesref->channel_layout = outlink->channel_layout;
195	outsamplesref->sample_rate = outlink->sample_rate;
196
197	if(insamplesref->pts != AV_NOPTS_VALUE) {
198	int64_t inpts = av_rescale(insamplesref->pts, inlink->time_base.num * (int64_t)outlink->sample_rate * inlink->sample_rate, inlink->time_base.den);
199	int64_t outpts= swr_next_pts(aresample->swr, inpts);
200	aresample->next_pts =
201	outsamplesref->pts = ROUNDED_DIV(outpts, inlink->sample_rate);
202	} else {
203	outsamplesref->pts = AV_NOPTS_VALUE;
204	}
205	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out,
206	(void *)insamplesref->extended_data, n_in);
207	if (n_out <= 0) {
208	av_frame_free(&outsamplesref);
209	av_frame_free(&insamplesref);
210	return 0;
211	}
212
213	outsamplesref->nb_samples = n_out;
214
215	ret = ff_filter_frame(outlink, outsamplesref);
216	aresample->req_fullfilled= 1;
217	av_frame_free(&insamplesref);
218	return ret;
219	}
220
221	static int request_frame(AVFilterLink *outlink)
222	{
223	AVFilterContext *ctx = outlink->src;
224	AResampleContext *aresample = ctx->priv;
225	AVFilterLink *const inlink = outlink->src->inputs[0];
226	int ret;
227
228	aresample->req_fullfilled = 0;
229	do{
230	ret = ff_request_frame(ctx->inputs[0]);
231	}while(!aresample->req_fullfilled && ret>=0);
232
233	if (ret == AVERROR_EOF) {
234	AVFrame *outsamplesref;
235	int n_out = 4096;
236	int64_t pts;
237
238	outsamplesref = ff_get_audio_buffer(outlink, n_out);
239	if (!outsamplesref)
240	return AVERROR(ENOMEM);
241
242	pts = swr_next_pts(aresample->swr, INT64_MIN);
243	pts = ROUNDED_DIV(pts, inlink->sample_rate);
244
245	n_out = swr_convert(aresample->swr, outsamplesref->extended_data, n_out, 0, 0);
246	if (n_out <= 0) {
247	av_frame_free(&outsamplesref);
248	return (n_out == 0) ? AVERROR_EOF : n_out;
249	}
250
251	outsamplesref->sample_rate = outlink->sample_rate;
252	outsamplesref->nb_samples = n_out;
253
254	outsamplesref->pts = pts;
255
256	return ff_filter_frame(outlink, outsamplesref);
257	}
258	return ret;
259	}
260
261	static const AVClass resample_child_class_next(const AVClass prev)
262	{
263	return prev ? NULL : swr_get_class();
264	}
265
266	static void resample_child_next(void obj, void *prev)
267	{
268	AResampleContext *s = obj;
269	return prev ? NULL : s->swr;
270	}
271
272	#define OFFSET(x) offsetof(AResampleContext, x)
273	#define FLAGS AV_OPT_FLAG_AUDIO_PARAM\|AV_OPT_FLAG_FILTERING_PARAM
274
275	static const AVOption options[] = {
276	{"sample_rate", NULL, OFFSET(sample_rate_arg), AV_OPT_TYPE_INT, {.i64=0}, 0, INT_MAX, FLAGS },
277	{NULL}
278	};
279
280	static const AVClass aresample_class = {
281	.class_name = "aresample",
282	.item_name = av_default_item_name,
283	.option = options,
284	.version = LIBAVUTIL_VERSION_INT,
285	.child_class_next = resample_child_class_next,
286	.child_next = resample_child_next,
287	};
288
289	static const AVFilterPad aresample_inputs[] = {
290	{
291	.name = "default",
292	.type = AVMEDIA_TYPE_AUDIO,
293	.filter_frame = filter_frame,
294	},
295	{ NULL }
296	};
297
298	static const AVFilterPad aresample_outputs[] = {
299	{
300	.name = "default",
301	.config_props = config_output,
302	.request_frame = request_frame,
303	.type = AVMEDIA_TYPE_AUDIO,
304	},
305	{ NULL }
306	};
307
308	AVFilter ff_af_aresample = {
309	.name = "aresample",
310	.description = NULL_IF_CONFIG_SMALL("Resample audio data."),
311	.init_dict = init_dict,
312	.uninit = uninit,
313	.query_formats = query_formats,
314	.priv_size = sizeof(AResampleContext),
315	.priv_class = &aresample_class,
316	.inputs = aresample_inputs,
317	.outputs = aresample_outputs,
318	};