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/*

 * Copyright (c) 2012 Stefano Sabatini

 *

 * Permission is hereby granted, free of charge, to any person obtaining a copy

 * of this software and associated documentation files (the "Software"), to deal

 * in the Software without restriction, including without limitation the rights

 * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell

 * copies of the Software, and to permit persons to whom the Software is

 * furnished to do so, subject to the following conditions:

 *

 * The above copyright notice and this permission notice shall be included in

 * all copies or substantial portions of the Software.

 *

 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR

 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,

 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL

 * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER

 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,

 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN

 * THE SOFTWARE.

 */

/**

 * @example resampling_audio.c

 * libswresample API use example.

 */

#include 

#include 

#include 

#include 

static int get_format_from_sample_fmt(const char **fmt,

                                      enum AVSampleFormat sample_fmt)

{

    int i;

    struct sample_fmt_entry {

        enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;

    } sample_fmt_entries[] = {

        { AV_SAMPLE_FMT_U8,  "u8",    "u8"    },

        { AV_SAMPLE_FMT_S16, "s16be", "s16le" },

        { AV_SAMPLE_FMT_S32, "s32be", "s32le" },

        { AV_SAMPLE_FMT_FLT, "f32be", "f32le" },

        { AV_SAMPLE_FMT_DBL, "f64be", "f64le" },

    };

    *fmt = NULL;

    for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {

        struct sample_fmt_entry *entry = &sample_fmt_entries[i];

        if (sample_fmt == entry->sample_fmt) {

            *fmt = AV_NE(entry->fmt_be, entry->fmt_le);

            return 0;

        }

    }

    fprintf(stderr,

            "Sample format %s not supported as output format\n",

            av_get_sample_fmt_name(sample_fmt));

    return AVERROR(EINVAL);

}

/**

 * Fill dst buffer with nb_samples, generated starting from t.

 */

static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)

{

    int i, j;

    double tincr = 1.0 / sample_rate, *dstp = dst;

    const double c = 2 * M_PI * 440.0;

    /* generate sin tone with 440Hz frequency and duplicated channels */

    for (i = 0; i < nb_samples; i++) {

        *dstp = sin(c * *t);

        for (j = 1; j < nb_channels; j++)

            dstp[j] = dstp[0];

        dstp += nb_channels;

        *t += tincr;

    }

}

int main(int argc, char **argv)

{

    int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;

    int src_rate = 48000, dst_rate = 44100;

    uint8_t **src_data = NULL, **dst_data = NULL;

    int src_nb_channels = 0, dst_nb_channels = 0;

    int src_linesize, dst_linesize;

    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;

    enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;

    const char *dst_filename = NULL;

    FILE *dst_file;

    int dst_bufsize;

    const char *fmt;

    struct SwrContext *swr_ctx;

    double t;

    int ret;

    if (argc != 2) {

        fprintf(stderr, "Usage: %s output_file\n"

                "API example program to show how to resample an audio stream with libswresample.\n"

                "This program generates a series of audio frames, resamples them to a specified "

                "output format and rate and saves them to an output file named output_file.\n",

            argv[0]);

        exit(1);

    }

    dst_filename = argv[1];

    dst_file = fopen(dst_filename, "wb");

    if (!dst_file) {

        fprintf(stderr, "Could not open destination file %s\n", dst_filename);

        exit(1);

    }

    /* create resampler context */

    swr_ctx = swr_alloc();

    if (!swr_ctx) {

        fprintf(stderr, "Could not allocate resampler context\n");

        ret = AVERROR(ENOMEM);

        goto end;

    }

    /* set options */

    av_opt_set_int(swr_ctx, "in_channel_layout",    src_ch_layout, 0);

    av_opt_set_int(swr_ctx, "in_sample_rate",       src_rate, 0);

    av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);

    av_opt_set_int(swr_ctx, "out_channel_layout",    dst_ch_layout, 0);

    av_opt_set_int(swr_ctx, "out_sample_rate",       dst_rate, 0);

    av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);

    /* initialize the resampling context */

    if ((ret = swr_init(swr_ctx)) < 0) {

        fprintf(stderr, "Failed to initialize the resampling context\n");

        goto end;

    }

    /* allocate source and destination samples buffers */

    src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);

    ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,

                                             src_nb_samples, src_sample_fmt, 0);

    if (ret < 0) {

        fprintf(stderr, "Could not allocate source samples\n");

        goto end;

    }

    /* compute the number of converted samples: buffering is avoided

     * ensuring that the output buffer will contain at least all the

     * converted input samples */

    max_dst_nb_samples = dst_nb_samples =

        av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

    /* buffer is going to be directly written to a rawaudio file, no alignment */

    dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);

    ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,

                                             dst_nb_samples, dst_sample_fmt, 0);

    if (ret < 0) {

        fprintf(stderr, "Could not allocate destination samples\n");

        goto end;

    }

    t = 0;

    do {

        /* generate synthetic audio */

        fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);

        /* compute destination number of samples */

        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +

                                        src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);

        if (dst_nb_samples > max_dst_nb_samples) {

            av_freep(&dst_data[0]);

            ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,

                                   dst_nb_samples, dst_sample_fmt, 1);

            if (ret < 0)

                break;

            max_dst_nb_samples = dst_nb_samples;

        }

        /* convert to destination format */

        ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);

        if (ret < 0) {

            fprintf(stderr, "Error while converting\n");

            goto end;

        }

        dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,

                                                 ret, dst_sample_fmt, 1);

        if (dst_bufsize < 0) {

            fprintf(stderr, "Could not get sample buffer size\n");

            goto end;

        }

        printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);

        fwrite(dst_data[0], 1, dst_bufsize, dst_file);

    } while (t < 10);

    if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)

        goto end;

    fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"

            "ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",

            fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);

end:

    fclose(dst_file);

    if (src_data)

        av_freep(&src_data[0]);

    av_freep(&src_data);

    if (dst_data)

        av_freep(&dst_data[0]);

    av_freep(&dst_data);

    swr_free(&swr_ctx);

    return ret < 0;

}