[libav.git] / libavfilter / af_resample.c

/*
 *
 * This file is part of Libav.
 *
 * Libav 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.
 *
 * Libav 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 Libav; 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/mathematics.h"
#include "libavutil/opt.h"

#include "libavresample/avresample.h"

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

typedef struct ResampleContext {
    AVAudioResampleContext *avr;

    int64_t next_pts;
} ResampleContext;

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

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

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

    AVFilterFormats        *in_formats      = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
    AVFilterFormats        *out_formats     = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);

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

    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)
        return 0;

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

    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 both the input and output formats are s16 or u8, use s16 as
       the internal sample format */
    if (av_get_bytes_per_sample(inlink->format)  <= 2 &&
        av_get_bytes_per_sample(outlink->format) <= 2)
        av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);

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

    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
    s->next_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 = avfilter_request_frame(ctx->inputs[0]);

    /* flush the lavr delay buffer */
    if (ret == AVERROR_EOF && s->avr) {
        AVFilterBufferRef *buf;
        int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
                                        outlink->sample_rate,
                                        ctx->inputs[0]->sample_rate,
                                        AV_ROUND_UP);

        if (!nb_samples)
            return ret;

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

        ret = avresample_convert(s->avr, (void**)buf->extended_data,
                                 buf->linesize[0], nb_samples,
                                 NULL, 0, 0);
        if (ret <= 0) {
            avfilter_unref_buffer(buf);
            return (ret == 0) ? AVERROR_EOF : ret;
        }

        buf->pts = s->next_pts;
        ff_filter_samples(outlink, buf);
        return 0;
    }
    return ret;
}

static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
    AVFilterContext  *ctx = inlink->dst;
    ResampleContext    *s = ctx->priv;
    AVFilterLink *outlink = ctx->outputs[0];

    if (s->avr) {
        AVFilterBufferRef *buf_out;
        int delay, nb_samples, ret;

        /* maximum possible samples lavr can output */
        delay      = avresample_get_delay(s->avr);
        nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
                                    outlink->sample_rate, inlink->sample_rate,
                                    AV_ROUND_UP);

        buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
        ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
                                     buf_out->linesize[0], nb_samples,
                                     (void**)buf->extended_data, buf->linesize[0],
                                     buf->audio->nb_samples);

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

        if (s->next_pts == AV_NOPTS_VALUE) {
            if (buf->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(buf->pts, inlink->time_base,
                                           outlink->time_base);
        }

        if (ret > 0) {
            buf_out->audio->nb_samples = ret;
            if (buf->pts != AV_NOPTS_VALUE) {
                buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
                                            outlink->time_base) -
                               av_rescale(delay, outlink->sample_rate,
                                          inlink->sample_rate);
            } else
                buf_out->pts = s->next_pts;

            s->next_pts = buf_out->pts + buf_out->audio->nb_samples;

            ff_filter_samples(outlink, buf_out);
        }
        avfilter_unref_buffer(buf);
    } else
        ff_filter_samples(outlink, buf);
}

AVFilter avfilter_af_resample = {
    .name          = "resample",
    .description   = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
    .priv_size     = sizeof(ResampleContext),

    .uninit         = uninit,
    .query_formats  = query_formats,

    .inputs    = (const AVFilterPad[]) {{ .name            = "default",
                                          .type            = AVMEDIA_TYPE_AUDIO,
                                          .filter_samples  = filter_samples,
                                          .min_perms       = AV_PERM_READ },
                                        { .name = NULL}},
    .outputs   = (const AVFilterPad[]) {{ .name          = "default",
                                          .type          = AVMEDIA_TYPE_AUDIO,
                                          .config_props  = config_output,
                                          .request_frame = request_frame },
                                        { .name = NULL}},
};
Commit	Line	Data
d371e7b9 AK	1	/*
	2	*
	3	* This file is part of Libav.
	4	*
	5	* Libav is free software; you can redistribute it and/or
	6	* modify it under the terms of the GNU Lesser General Public
	7	* License as published by the Free Software Foundation; either
	8	* version 2.1 of the License, or (at your option) any later version.
	9	*
	10	* Libav is distributed in the hope that it will be useful,
	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	13	* Lesser General Public License for more details.
	14	*
	15	* You should have received a copy of the GNU Lesser General Public
	16	* License along with Libav; if not, write to the Free Software
	17	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	18	*/
	19
	20	/**
	21	* @file
	22	* sample format and channel layout conversion audio filter
	23	*/
	24
	25	#include "libavutil/avassert.h"
	26	#include "libavutil/avstring.h"
	27	#include "libavutil/mathematics.h"
	28	#include "libavutil/opt.h"
	29
	30	#include "libavresample/avresample.h"
	31
	32	#include "audio.h"
	33	#include "avfilter.h"
	34	#include "internal.h"
	35
	36	typedef struct ResampleContext {
	37	AVAudioResampleContext *avr;
	38
	39	int64_t next_pts;
	40	} ResampleContext;
	41
	42	static av_cold void uninit(AVFilterContext *ctx)
	43	{
	44	ResampleContext *s = ctx->priv;
	45
	46	if (s->avr) {
	47	avresample_close(s->avr);
	48	avresample_free(&s->avr);
	49	}
	50	}
	51
	52	static int query_formats(AVFilterContext *ctx)
	53	{
	54	AVFilterLink *inlink = ctx->inputs[0];
	55	AVFilterLink *outlink = ctx->outputs[0];
	56
	57	AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
	58	AVFilterFormats *out_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
	59
	60	avfilter_formats_ref(in_formats, &inlink->out_formats);
	61	avfilter_formats_ref(out_formats, &outlink->in_formats);
	62
	63	return 0;
	64	}
65
66	static int config_output(AVFilterLink *outlink)
67	{
68	AVFilterContext *ctx = outlink->src;
69	AVFilterLink *inlink = ctx->inputs[0];
70	ResampleContext *s = ctx->priv;
71	char buf1[64], buf2[64];
72	int ret;
73
74	if (s->avr) {
75	avresample_close(s->avr);
76	avresample_free(&s->avr);
77	}
78
79	if (inlink->channel_layout == outlink->channel_layout &&
80	inlink->sample_rate == outlink->sample_rate &&
81	inlink->format == outlink->format)
82	return 0;
83
84	if (!(s->avr = avresample_alloc_context()))
85	return AVERROR(ENOMEM);
86
87	av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
88	av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
89	av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
90	av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
91	av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
92	av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
93
94	/* if both the input and output formats are s16 or u8, use s16 as
95	the internal sample format */
96	if (av_get_bytes_per_sample(inlink->format) <= 2 &&
97	av_get_bytes_per_sample(outlink->format) <= 2)
98	av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
99
100	if ((ret = avresample_open(s->avr)) < 0)
101	return ret;
102
103	outlink->time_base = (AVRational){ 1, outlink->sample_rate };
104	s->next_pts = AV_NOPTS_VALUE;
105
106	av_get_channel_layout_string(buf1, sizeof(buf1),
107	-1, inlink ->channel_layout);
108	av_get_channel_layout_string(buf2, sizeof(buf2),
109	-1, outlink->channel_layout);
110	av_log(ctx, AV_LOG_VERBOSE,
111	"fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
112	av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
113	av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
114
115	return 0;
116	}
117
118	static int request_frame(AVFilterLink *outlink)
119	{
120	AVFilterContext *ctx = outlink->src;
121	ResampleContext *s = ctx->priv;
122	int ret = avfilter_request_frame(ctx->inputs[0]);
123
124	/* flush the lavr delay buffer */
125	if (ret == AVERROR_EOF && s->avr) {
126	AVFilterBufferRef *buf;
127	int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
128	outlink->sample_rate,
129	ctx->inputs[0]->sample_rate,
130	AV_ROUND_UP);
131
132	if (!nb_samples)
133	return ret;
134
135	buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
136	if (!buf)
137	return AVERROR(ENOMEM);
138
139	ret = avresample_convert(s->avr, (void**)buf->extended_data,
140	buf->linesize[0], nb_samples,
141	NULL, 0, 0);
142	if (ret <= 0) {
143	avfilter_unref_buffer(buf);
144	return (ret == 0) ? AVERROR_EOF : ret;
145	}
146
147	buf->pts = s->next_pts;
148	ff_filter_samples(outlink, buf);
149	return 0;
150	}
151	return ret;
152	}
153
154	static void filter_samples(AVFilterLink inlink, AVFilterBufferRef buf)
155	{
156	AVFilterContext *ctx = inlink->dst;
157	ResampleContext *s = ctx->priv;
158	AVFilterLink *outlink = ctx->outputs[0];
159
160	if (s->avr) {
161	AVFilterBufferRef *buf_out;
162	int delay, nb_samples, ret;
163
164	/* maximum possible samples lavr can output */
165	delay = avresample_get_delay(s->avr);
166	nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
167	outlink->sample_rate, inlink->sample_rate,
168	AV_ROUND_UP);
169
170	buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
171	ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
172	buf_out->linesize[0], nb_samples,
173	(void**)buf->extended_data, buf->linesize[0],
174	buf->audio->nb_samples);
175
176	av_assert0(!avresample_available(s->avr));
177
178	if (s->next_pts == AV_NOPTS_VALUE) {
179	if (buf->pts == AV_NOPTS_VALUE) {
180	av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
181	"assuming 0.\n");
182	s->next_pts = 0;
183	} else
184	s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
185	outlink->time_base);
186	}
187
188	if (ret > 0) {
189	buf_out->audio->nb_samples = ret;
190	if (buf->pts != AV_NOPTS_VALUE) {
191	buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
192	outlink->time_base) -
193	av_rescale(delay, outlink->sample_rate,
194	inlink->sample_rate);
195	} else
196	buf_out->pts = s->next_pts;
197
198	s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
199
200	ff_filter_samples(outlink, buf_out);
201	}
202	avfilter_unref_buffer(buf);
203	} else
204	ff_filter_samples(outlink, buf);
205	}
206
207	AVFilter avfilter_af_resample = {
208	.name = "resample",
209	.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
210	.priv_size = sizeof(ResampleContext),
211
212	.uninit = uninit,
213	.query_formats = query_formats,
214
215	.inputs = (const AVFilterPad[]) {{ .name = "default",
216	.type = AVMEDIA_TYPE_AUDIO,
217	.filter_samples = filter_samples,
218	.min_perms = AV_PERM_READ },
219	{ .name = NULL}},
220	.outputs = (const AVFilterPad[]) {{ .name = "default",
221	.type = AVMEDIA_TYPE_AUDIO,
222	.config_props = config_output,
223	.request_frame = request_frame },
224	{ .name = NULL}},
225	};