mirror of https://git.ffmpeg.org/ffmpeg.git
224 lines
8.2 KiB
C
224 lines
8.2 KiB
C
/*
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* Copyright (c) 2012 Stefano Sabatini
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
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* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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/**
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* @example doc/examples/resampling_audio.c
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* libswresample API use example.
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*/
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#include <libavutil/opt.h>
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#include <libavutil/channel_layout.h>
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#include <libavutil/samplefmt.h>
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#include <libswresample/swresample.h>
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static int get_format_from_sample_fmt(const char **fmt,
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enum AVSampleFormat sample_fmt)
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{
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int i;
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struct sample_fmt_entry {
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enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
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} sample_fmt_entries[] = {
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{ AV_SAMPLE_FMT_U8, "u8", "u8" },
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{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
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{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
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{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
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{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
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};
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*fmt = NULL;
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for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
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struct sample_fmt_entry *entry = &sample_fmt_entries[i];
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if (sample_fmt == entry->sample_fmt) {
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*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
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return 0;
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}
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}
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fprintf(stderr,
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"Sample format %s not supported as output format\n",
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av_get_sample_fmt_name(sample_fmt));
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return AVERROR(EINVAL);
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}
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/**
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* Fill dst buffer with nb_samples, generated starting from t.
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*/
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void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
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{
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int i, j;
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double tincr = 1.0 / sample_rate, *dstp = dst;
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const double c = 2 * M_PI * 440.0;
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/* generate sin tone with 440Hz frequency and duplicated channels */
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for (i = 0; i < nb_samples; i++) {
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*dstp = sin(c * *t);
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for (j = 1; j < nb_channels; j++)
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dstp[j] = dstp[0];
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dstp += nb_channels;
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*t += tincr;
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}
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}
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int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
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int nb_samples, enum AVSampleFormat sample_fmt, int align)
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{
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int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
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*data = av_malloc(sizeof(*data) * nb_planes);
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if (!*data)
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return AVERROR(ENOMEM);
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return av_samples_alloc(*data, linesize, nb_channels,
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nb_samples, sample_fmt, align);
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}
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int main(int argc, char **argv)
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{
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int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
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int src_rate = 48000, dst_rate = 44100;
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uint8_t **src_data = NULL, **dst_data = NULL;
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int src_nb_channels = 0, dst_nb_channels = 0;
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int src_linesize, dst_linesize;
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int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
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enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
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const char *dst_filename = NULL;
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FILE *dst_file;
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int dst_bufsize;
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const char *fmt;
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struct SwrContext *swr_ctx;
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double t;
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int ret;
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if (argc != 2) {
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fprintf(stderr, "Usage: %s output_file\n"
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"API example program to show how to resample an audio stream with libswresample.\n"
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"This program generates a series of audio frames, resamples them to a specified "
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"output format and rate and saves them to an output file named output_file.\n",
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argv[0]);
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exit(1);
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}
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dst_filename = argv[1];
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dst_file = fopen(dst_filename, "wb");
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if (!dst_file) {
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fprintf(stderr, "Could not open destination file %s\n", dst_filename);
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exit(1);
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}
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/* create resampler context */
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swr_ctx = swr_alloc();
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if (!swr_ctx) {
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fprintf(stderr, "Could not allocate resampler context\n");
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ret = AVERROR(ENOMEM);
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goto end;
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}
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/* set options */
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av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
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av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
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av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
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av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
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av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
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/* initialize the resampling context */
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if ((ret = swr_init(swr_ctx)) < 0) {
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fprintf(stderr, "Failed to initialize the resampling context\n");
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goto end;
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}
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/* allocate source and destination samples buffers */
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src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
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ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
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src_nb_samples, src_sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate source samples\n");
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goto end;
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}
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/* compute the number of converted samples: buffering is avoided
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* ensuring that the output buffer will contain at least all the
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* converted input samples */
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max_dst_nb_samples = dst_nb_samples =
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av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
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/* buffer is going to be directly written to a rawaudio file, no alignment */
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dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
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ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
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dst_nb_samples, dst_sample_fmt, 0);
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if (ret < 0) {
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fprintf(stderr, "Could not allocate destination samples\n");
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goto end;
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}
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t = 0;
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do {
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/* generate synthetic audio */
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fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
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/* compute destination number of samples */
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dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
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src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
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if (dst_nb_samples > max_dst_nb_samples) {
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av_free(dst_data[0]);
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ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
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dst_nb_samples, dst_sample_fmt, 1);
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if (ret < 0)
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break;
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max_dst_nb_samples = dst_nb_samples;
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}
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/* convert to destination format */
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ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
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if (ret < 0) {
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fprintf(stderr, "Error while converting\n");
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goto end;
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}
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dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
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ret, dst_sample_fmt, 1);
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printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
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fwrite(dst_data[0], 1, dst_bufsize, dst_file);
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} while (t < 10);
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if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
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goto end;
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fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
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"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
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fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
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end:
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if (dst_file)
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fclose(dst_file);
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if (src_data)
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av_freep(&src_data[0]);
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av_freep(&src_data);
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if (dst_data)
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av_freep(&dst_data[0]);
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av_freep(&dst_data);
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swr_free(&swr_ctx);
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return ret < 0;
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}
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