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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;}
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