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Diffstat (limited to 'doc/examples/resampling_audio.c')
| -rw-r--r-- | doc/examples/resampling_audio.c | 211 |
1 files changed, 211 insertions, 0 deletions
diff --git a/doc/examples/resampling_audio.c b/doc/examples/resampling_audio.c new file mode 100644 index 0000000000..70db9efe05 --- /dev/null +++ b/doc/examples/resampling_audio.c @@ -0,0 +1,211 @@ +/* + * 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 doc/examples/resampling_audio.c + * libswresample API use example. + */ + +#include <libavutil/opt.h> +#include <libavutil/channel_layout.h> +#include <libavutil/samplefmt.h> +#include <libswresample/swresample.h> + +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. + */ +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_free(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); + 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: + if (dst_file) + 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; +} |