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

/*
 * 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
 * sample format and channel layout conversion audio filter
 */

#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/common.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"

#include "libavresample/avresample.h"

#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"

typedef struct ResampleContext {
    const AVClass *class;
    AVAudioResampleContext *avr;
    AVDictionary *options;

    int64_t next_pts;
    int64_t next_in_pts;

    /* set by filter_frame() to signal an output frame to request_frame() */
    int got_output;
} ResampleContext;

static av_cold int init(AVFilterContext *ctx, AVDictionary **opts)
{
    ResampleContext *s = ctx->priv;
    const AVClass *avr_class = avresample_get_class();
    AVDictionaryEntry *e = NULL;

    while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
        if (av_opt_find(&avr_class, e->key, NULL, 0,
                        AV_OPT_SEARCH_FAKE_OBJ | AV_OPT_SEARCH_CHILDREN))
            av_dict_set(&s->options, e->key, e->value, 0);
    }

    e = NULL;
    while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
        av_dict_set(opts, e->key, NULL, 0);

    /* do not allow the user to override basic format options */
    av_dict_set(&s->options,  "in_channel_layout", NULL, 0);
    av_dict_set(&s->options, "out_channel_layout", NULL, 0);
    av_dict_set(&s->options,  "in_sample_fmt",     NULL, 0);
    av_dict_set(&s->options, "out_sample_fmt",     NULL, 0);
    av_dict_set(&s->options,  "in_sample_rate",    NULL, 0);
    av_dict_set(&s->options, "out_sample_rate",    NULL, 0);

    return 0;
}

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

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }
    av_dict_free(&s->options);
}

static int query_formats(AVFilterContext *ctx)
{
    AVFilterLink *inlink  = ctx->inputs[0];
    AVFilterLink *outlink = ctx->outputs[0];

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

    ff_formats_ref(in_formats,  &inlink->out_formats);
    ff_formats_ref(out_formats, &outlink->in_formats);

    ff_formats_ref(in_samplerates,  &inlink->out_samplerates);
    ff_formats_ref(out_samplerates, &outlink->in_samplerates);

    ff_channel_layouts_ref(in_layouts,  &inlink->out_channel_layouts);
    ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);

    return 0;
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    AVFilterLink *inlink = ctx->inputs[0];
    ResampleContext   *s = ctx->priv;
    char buf1[64], buf2[64];
    int ret;

    if (s->avr) {
        avresample_close(s->avr);
        avresample_free(&s->avr);
    }

    if (inlink->channel_layout == outlink->channel_layout &&
        inlink->sample_rate    == outlink->sample_rate    &&
        (inlink->format        == outlink->format ||
        (av_get_channel_layout_nb_channels(inlink->channel_layout)  == 1 &&
         av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
         av_get_planar_sample_fmt(inlink->format) ==
         av_get_planar_sample_fmt(outlink->format))))
        return 0;

    if (!(s->avr = avresample_alloc_context()))
        return AVERROR(ENOMEM);

    if (s->options) {
        AVDictionaryEntry *e = NULL;
        while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
            av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);

        av_opt_set_dict(s->avr, &s->options);
    }

    av_opt_set_int(s->avr,  "in_channel_layout", inlink ->channel_layout, 0);
    av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
    av_opt_set_int(s->avr,  "in_sample_fmt",     inlink ->format,         0);
    av_opt_set_int(s->avr, "out_sample_fmt",     outlink->format,         0);
    av_opt_set_int(s->avr,  "in_sample_rate",    inlink ->sample_rate,    0);
    av_opt_set_int(s->avr, "out_sample_rate",    outlink->sample_rate,    0);

    if ((ret = avresample_open(s->avr)) < 0)
        return ret;

    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
    s->next_pts        = AV_NOPTS_VALUE;
    s->next_in_pts     = AV_NOPTS_VALUE;

    av_get_channel_layout_string(buf1, sizeof(buf1),
                                 -1, inlink ->channel_layout);
    av_get_channel_layout_string(buf2, sizeof(buf2),
                                 -1, outlink->channel_layout);
    av_log(ctx, AV_LOG_VERBOSE,
           "fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
           av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);

    return 0;
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    ResampleContext   *s = ctx->priv;
    int ret = 0;

    s->got_output = 0;
    while (ret >= 0 && !s->got_output)
        ret = ff_request_frame(ctx->inputs[0]);

    /* flush the lavr delay buffer */
    if (ret == AVERROR_EOF && s->avr) {
        AVFrame *frame;
        int nb_samples = avresample_get_out_samples(s->avr, 0);

        if (!nb_samples)
            return ret;

        frame = ff_get_audio_buffer(outlink, nb_samples);
        if (!frame)
            return AVERROR(ENOMEM);

        ret = avresample_convert(s->avr, frame->extended_data,
                                 frame->linesize[0], nb_samples,
                                 NULL, 0, 0);
        if (ret <= 0) {
            av_frame_free(&frame);
            return (ret == 0) ? AVERROR_EOF : ret;
        }

        frame->pts = s->next_pts;
        return ff_filter_frame(outlink, frame);
    }
    return ret;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext  *ctx = inlink->dst;
    ResampleContext    *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];
    int ret;

    if (s->avr) {
        AVFrame *out;
        int delay, nb_samples;

        /* maximum possible samples lavr can output */
        delay      = avresample_get_delay(s->avr);
        nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);

        out = ff_get_audio_buffer(outlink, nb_samples);
        if (!out) {
            ret = AVERROR(ENOMEM);
            goto fail;
        }

        ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
                                 nb_samples, in->extended_data, in->linesize[0],
                                 in->nb_samples);
        if (ret <= 0) {
            av_frame_free(&out);
            if (ret < 0)
                goto fail;
        }

        av_assert0(!avresample_available(s->avr));

        if (s->next_pts == AV_NOPTS_VALUE) {
            if (in->pts == AV_NOPTS_VALUE) {
                av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                       "assuming 0.\n");
                s->next_pts = 0;
            } else
                s->next_pts = av_rescale_q(in->pts, inlink->time_base,
                                           outlink->time_base);
        }

        if (ret > 0) {
            out->nb_samples = ret;

            ret = av_frame_copy_props(out, in);
            if (ret < 0) {
                av_frame_free(&out);
                goto fail;
            }

            out->sample_rate = outlink->sample_rate;
            /* Only convert in->pts if there is a discontinuous jump.
               This ensures that out->pts tracks the number of samples actually
               output by the resampler in the absence of such a jump.
               Otherwise, the rounding in av_rescale_q() and av_rescale()
               causes off-by-1 errors. */
            if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
                out->pts = av_rescale_q(in->pts, inlink->time_base,
                                            outlink->time_base) -
                               av_rescale(delay, outlink->sample_rate,
                                          inlink->sample_rate);
            } else
                out->pts = s->next_pts;

            s->next_pts = out->pts + out->nb_samples;
            s->next_in_pts = in->pts + in->nb_samples;

            ret = ff_filter_frame(outlink, out);
            s->got_output = 1;
        }

fail:
        av_frame_free(&in);
    } else {
        in->format = outlink->format;
        ret = ff_filter_frame(outlink, in);
        s->got_output = 1;
    }

    return ret;
}

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

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

static const AVClass resample_class = {
    .class_name       = "resample",
    .item_name        = av_default_item_name,
    .version          = LIBAVUTIL_VERSION_INT,
    .child_class_next = resample_child_class_next,
    .child_next       = resample_child_next,
};

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

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

AVFilter ff_af_resample = {
    .name          = "resample",
    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
    .priv_size     = sizeof(ResampleContext),
    .priv_class    = &resample_class,
    .init_dict     = init,
    .uninit        = uninit,
    .query_formats = query_formats,
    .inputs        = avfilter_af_resample_inputs,
    .outputs       = avfilter_af_resample_outputs,
};
Commit	Line	Data
2ba45a60 DM	1	/*
	2	* This file is part of FFmpeg.
	3	*
	4	* FFmpeg is free software; you can redistribute it and/or
	5	* modify it under the terms of the GNU Lesser General Public
	6	* License as published by the Free Software Foundation; either
	7	* version 2.1 of the License, or (at your option) any later version.
	8	*
	9	* FFmpeg is distributed in the hope that it will be useful,
	10	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	11	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	12	* Lesser General Public License for more details.
	13	*
	14	* You should have received a copy of the GNU Lesser General Public
	15	* License along with FFmpeg; if not, write to the Free Software
	16	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	17	*/
	18
	19	/**
	20	* @file
	21	* sample format and channel layout conversion audio filter
	22	*/
	23
	24	#include "libavutil/avassert.h"
	25	#include "libavutil/avstring.h"
	26	#include "libavutil/common.h"
	27	#include "libavutil/dict.h"
	28	#include "libavutil/mathematics.h"
	29	#include "libavutil/opt.h"
	30
	31	#include "libavresample/avresample.h"
	32
	33	#include "audio.h"
	34	#include "avfilter.h"
	35	#include "formats.h"
	36	#include "internal.h"
	37
	38	typedef struct ResampleContext {
	39	const AVClass *class;
	40	AVAudioResampleContext *avr;
	41	AVDictionary *options;
	42
	43	int64_t next_pts;
	44	int64_t next_in_pts;
	45
	46	/* set by filter_frame() to signal an output frame to request_frame() */
	47	int got_output;
	48	} ResampleContext;
	49
	50	static av_cold int init(AVFilterContext ctx, AVDictionary *opts)
	51	{
	52	ResampleContext *s = ctx->priv;
	53	const AVClass *avr_class = avresample_get_class();
	54	AVDictionaryEntry *e = NULL;
	55
	56	while ((e = av_dict_get(*opts, "", e, AV_DICT_IGNORE_SUFFIX))) {
	57	if (av_opt_find(&avr_class, e->key, NULL, 0,
	58	AV_OPT_SEARCH_FAKE_OBJ \| AV_OPT_SEARCH_CHILDREN))
	59	av_dict_set(&s->options, e->key, e->value, 0);
	60	}
	61
	62	e = NULL;
	63	while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
	64	av_dict_set(opts, e->key, NULL, 0);
65
66	/* do not allow the user to override basic format options */
67	av_dict_set(&s->options, "in_channel_layout", NULL, 0);
68	av_dict_set(&s->options, "out_channel_layout", NULL, 0);
69	av_dict_set(&s->options, "in_sample_fmt", NULL, 0);
70	av_dict_set(&s->options, "out_sample_fmt", NULL, 0);
71	av_dict_set(&s->options, "in_sample_rate", NULL, 0);
72	av_dict_set(&s->options, "out_sample_rate", NULL, 0);
73
74	return 0;
75	}
76
77	static av_cold void uninit(AVFilterContext *ctx)
78	{
79	ResampleContext *s = ctx->priv;
80
81	if (s->avr) {
82	avresample_close(s->avr);
83	avresample_free(&s->avr);
84	}
85	av_dict_free(&s->options);
86	}
87
88	static int query_formats(AVFilterContext *ctx)
89	{
90	AVFilterLink *inlink = ctx->inputs[0];
91	AVFilterLink *outlink = ctx->outputs[0];
92
93	AVFilterFormats *in_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
94	AVFilterFormats *out_formats = ff_all_formats(AVMEDIA_TYPE_AUDIO);
95	AVFilterFormats *in_samplerates = ff_all_samplerates();
96	AVFilterFormats *out_samplerates = ff_all_samplerates();
97	AVFilterChannelLayouts *in_layouts = ff_all_channel_layouts();
98	AVFilterChannelLayouts *out_layouts = ff_all_channel_layouts();
99
100	ff_formats_ref(in_formats, &inlink->out_formats);
101	ff_formats_ref(out_formats, &outlink->in_formats);
102
103	ff_formats_ref(in_samplerates, &inlink->out_samplerates);
104	ff_formats_ref(out_samplerates, &outlink->in_samplerates);
105
106	ff_channel_layouts_ref(in_layouts, &inlink->out_channel_layouts);
107	ff_channel_layouts_ref(out_layouts, &outlink->in_channel_layouts);
108
109	return 0;
110	}
111
112	static int config_output(AVFilterLink *outlink)
113	{
114	AVFilterContext *ctx = outlink->src;
115	AVFilterLink *inlink = ctx->inputs[0];
116	ResampleContext *s = ctx->priv;
117	char buf1[64], buf2[64];
118	int ret;
119
120	if (s->avr) {
121	avresample_close(s->avr);
122	avresample_free(&s->avr);
123	}
124
125	if (inlink->channel_layout == outlink->channel_layout &&
126	inlink->sample_rate == outlink->sample_rate &&
127	(inlink->format == outlink->format \|\|
128	(av_get_channel_layout_nb_channels(inlink->channel_layout) == 1 &&
129	av_get_channel_layout_nb_channels(outlink->channel_layout) == 1 &&
130	av_get_planar_sample_fmt(inlink->format) ==
131	av_get_planar_sample_fmt(outlink->format))))
132	return 0;
133
134	if (!(s->avr = avresample_alloc_context()))
135	return AVERROR(ENOMEM);
136
137	if (s->options) {
138	AVDictionaryEntry *e = NULL;
139	while ((e = av_dict_get(s->options, "", e, AV_DICT_IGNORE_SUFFIX)))
140	av_log(ctx, AV_LOG_VERBOSE, "lavr option: %s=%s\n", e->key, e->value);
141
142	av_opt_set_dict(s->avr, &s->options);
143	}
144
145	av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
146	av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
147	av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
148	av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
149	av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
150	av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
151
152	if ((ret = avresample_open(s->avr)) < 0)
153	return ret;
154
155	outlink->time_base = (AVRational){ 1, outlink->sample_rate };
156	s->next_pts = AV_NOPTS_VALUE;
157	s->next_in_pts = AV_NOPTS_VALUE;
158
159	av_get_channel_layout_string(buf1, sizeof(buf1),
160	-1, inlink ->channel_layout);
161	av_get_channel_layout_string(buf2, sizeof(buf2),
162	-1, outlink->channel_layout);
163	av_log(ctx, AV_LOG_VERBOSE,
164	"fmt:%s srate:%d cl:%s -> fmt:%s srate:%d cl:%s\n",
165	av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
166	av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
167
168	return 0;
169	}
170
171	static int request_frame(AVFilterLink *outlink)
172	{
173	AVFilterContext *ctx = outlink->src;
174	ResampleContext *s = ctx->priv;
175	int ret = 0;
176
177	s->got_output = 0;
178	while (ret >= 0 && !s->got_output)
179	ret = ff_request_frame(ctx->inputs[0]);
180
181	/* flush the lavr delay buffer */
182	if (ret == AVERROR_EOF && s->avr) {
183	AVFrame *frame;
184	int nb_samples = avresample_get_out_samples(s->avr, 0);
185
186	if (!nb_samples)
187	return ret;
188
189	frame = ff_get_audio_buffer(outlink, nb_samples);
190	if (!frame)
191	return AVERROR(ENOMEM);
192
193	ret = avresample_convert(s->avr, frame->extended_data,
194	frame->linesize[0], nb_samples,
195	NULL, 0, 0);
196	if (ret <= 0) {
197	av_frame_free(&frame);
198	return (ret == 0) ? AVERROR_EOF : ret;
199	}
200
201	frame->pts = s->next_pts;
202	return ff_filter_frame(outlink, frame);
203	}
204	return ret;
205	}
206
207	static int filter_frame(AVFilterLink inlink, AVFrame in)
208	{
209	AVFilterContext *ctx = inlink->dst;
210	ResampleContext *s = ctx->priv;
211	AVFilterLink *outlink = ctx->outputs[0];
212	int ret;
213
214	if (s->avr) {
215	AVFrame *out;
216	int delay, nb_samples;
217
218	/* maximum possible samples lavr can output */
219	delay = avresample_get_delay(s->avr);
220	nb_samples = avresample_get_out_samples(s->avr, in->nb_samples);
221
222	out = ff_get_audio_buffer(outlink, nb_samples);
223	if (!out) {
224	ret = AVERROR(ENOMEM);
225	goto fail;
226	}
227
228	ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
229	nb_samples, in->extended_data, in->linesize[0],
230	in->nb_samples);
231	if (ret <= 0) {
232	av_frame_free(&out);
233	if (ret < 0)
234	goto fail;
235	}
236
237	av_assert0(!avresample_available(s->avr));
238
239	if (s->next_pts == AV_NOPTS_VALUE) {
240	if (in->pts == AV_NOPTS_VALUE) {
241	av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
242	"assuming 0.\n");
243	s->next_pts = 0;
244	} else
245	s->next_pts = av_rescale_q(in->pts, inlink->time_base,
246	outlink->time_base);
247	}
248
249	if (ret > 0) {
250	out->nb_samples = ret;
251
252	ret = av_frame_copy_props(out, in);
253	if (ret < 0) {
254	av_frame_free(&out);
255	goto fail;
256	}
257
258	out->sample_rate = outlink->sample_rate;
259	/* Only convert in->pts if there is a discontinuous jump.
260	This ensures that out->pts tracks the number of samples actually
261	output by the resampler in the absence of such a jump.
262	Otherwise, the rounding in av_rescale_q() and av_rescale()
263	causes off-by-1 errors. */
264	if (in->pts != AV_NOPTS_VALUE && in->pts != s->next_in_pts) {
265	out->pts = av_rescale_q(in->pts, inlink->time_base,
266	outlink->time_base) -
267	av_rescale(delay, outlink->sample_rate,
268	inlink->sample_rate);
269	} else
270	out->pts = s->next_pts;
271
272	s->next_pts = out->pts + out->nb_samples;
273	s->next_in_pts = in->pts + in->nb_samples;
274
275	ret = ff_filter_frame(outlink, out);
276	s->got_output = 1;
277	}
278
279	fail:
280	av_frame_free(&in);
281	} else {
282	in->format = outlink->format;
283	ret = ff_filter_frame(outlink, in);
284	s->got_output = 1;
285	}
286
287	return ret;
288	}
289
290	static const AVClass resample_child_class_next(const AVClass prev)
291	{
292	return prev ? NULL : avresample_get_class();
293	}
294
295	static void resample_child_next(void obj, void *prev)
296	{
297	ResampleContext *s = obj;
298	return prev ? NULL : s->avr;
299	}
300
301	static const AVClass resample_class = {
302	.class_name = "resample",
303	.item_name = av_default_item_name,
304	.version = LIBAVUTIL_VERSION_INT,
305	.child_class_next = resample_child_class_next,
306	.child_next = resample_child_next,
307	};
308
309	static const AVFilterPad avfilter_af_resample_inputs[] = {
310	{
311	.name = "default",
312	.type = AVMEDIA_TYPE_AUDIO,
313	.filter_frame = filter_frame,
314	},
315	{ NULL }
316	};
317
318	static const AVFilterPad avfilter_af_resample_outputs[] = {
319	{
320	.name = "default",
321	.type = AVMEDIA_TYPE_AUDIO,
322	.config_props = config_output,
323	.request_frame = request_frame
324	},
325	{ NULL }
326	};
327
328	AVFilter ff_af_resample = {
329	.name = "resample",
330	.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
331	.priv_size = sizeof(ResampleContext),
332	.priv_class = &resample_class,
333	.init_dict = init,
334	.uninit = uninit,
335	.query_formats = query_formats,
336	.inputs = avfilter_af_resample_inputs,
337	.outputs = avfilter_af_resample_outputs,
338	};