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/*

 * 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;

    /* 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;

    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 = av_rescale_rnd(avresample_get_delay(s->avr),

                                        outlink->sample_rate,

                                        ctx->inputs[0]->sample_rate,

                                        AV_ROUND_UP);

        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 = av_rescale_rnd(in->nb_samples + delay,

                                    outlink->sample_rate, inlink->sample_rate,

                                    AV_ROUND_UP);

        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;

            if (in->pts != AV_NOPTS_VALUE) {

                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;

            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 avfilter_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,

};