/* * Copyright (c) 2002 Fabrice Bellard * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> * * 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 */ #include <stdint.h> #include <stdio.h> #include "libavutil/avstring.h" #include "libavutil/common.h" #include "libavutil/lfg.h" #include "libavutil/libm.h" #include "libavutil/log.h" #include "libavutil/mem.h" #include "libavutil/opt.h" #include "libavutil/samplefmt.h" #include "avresample.h" static double dbl_rand(AVLFG *lfg) { return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0; } #define PUT_FUNC(name, fmt, type, expr) \ static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\ int channels, int sample, int ch, \ double v_dbl) \ { \ type v = expr; \ type **out = (type **)data; \ if (av_sample_fmt_is_planar(sample_fmt)) \ out[ch][sample] = v; \ else \ out[0][sample * channels + ch] = v; \ } PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128)) PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15)))) PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31)))) PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl) PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl) static void put_sample(void **data, enum AVSampleFormat sample_fmt, int channels, int sample, int ch, double v_dbl) { switch (av_get_packed_sample_fmt(sample_fmt)) { case AV_SAMPLE_FMT_U8: put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl); break; case AV_SAMPLE_FMT_S16: put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl); break; case AV_SAMPLE_FMT_S32: put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl); break; case AV_SAMPLE_FMT_FLT: put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl); break; case AV_SAMPLE_FMT_DBL: put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl); break; } } static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt, int channels, int sample_rate, int nb_samples) { int i, ch, k; double v, f, a, ampa; double tabf1[AVRESAMPLE_MAX_CHANNELS]; double tabf2[AVRESAMPLE_MAX_CHANNELS]; double taba[AVRESAMPLE_MAX_CHANNELS]; #define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v); k = 0; /* 1 second of single freq sine at 1000 Hz */ a = 0; for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { v = sin(a) * 0.30; for (ch = 0; ch < channels; ch++) PUT_SAMPLE a += M_PI * 1000.0 * 2.0 / sample_rate; } /* 1 second of varying frequency between 100 and 10000 Hz */ a = 0; for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { v = sin(a) * 0.30; for (ch = 0; ch < channels; ch++) PUT_SAMPLE f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate); a += M_PI * f * 2.0 / sample_rate; } /* 0.5 second of low amplitude white noise */ for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { v = dbl_rand(rnd) * 0.30; for (ch = 0; ch < channels; ch++) PUT_SAMPLE } /* 0.5 second of high amplitude white noise */ for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { v = dbl_rand(rnd); for (ch = 0; ch < channels; ch++) PUT_SAMPLE } /* 1 second of unrelated ramps for each channel */ for (ch = 0; ch < channels; ch++) { taba[ch] = 0; tabf1[ch] = 100 + av_lfg_get(rnd) % 5000; tabf2[ch] = 100 + av_lfg_get(rnd) % 5000; } for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { for (ch = 0; ch < channels; ch++) { v = sin(taba[ch]) * 0.30; PUT_SAMPLE f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate); taba[ch] += M_PI * f * 2.0 / sample_rate; } } /* 2 seconds of 500 Hz with varying volume */ a = 0; ampa = 0; for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) { for (ch = 0; ch < channels; ch++) { double amp = (1.0 + sin(ampa)) * 0.15; if (ch & 1) amp = 0.30 - amp; v = sin(a) * amp; PUT_SAMPLE a += M_PI * 500.0 * 2.0 / sample_rate; ampa += M_PI * 2.0 / sample_rate; } } } /* formats, rates, and layouts are ordered for priority in testing. e.g. 'avresample-test 4 2 2' will test all input/output combinations of S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */ static const enum AVSampleFormat formats[] = { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_DBL, }; static const int rates[] = { 48000, 44100, 16000 }; static const uint64_t layouts[] = { AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_7POINT1, }; int main(int argc, char **argv) { AVAudioResampleContext *s; AVLFG rnd; int ret = 0; uint8_t *in_buf = NULL; uint8_t *out_buf = NULL; unsigned int in_buf_size; unsigned int out_buf_size; uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; int in_linesize; int out_linesize; uint64_t in_ch_layout; int in_channels; enum AVSampleFormat in_fmt; int in_rate; uint64_t out_ch_layout; int out_channels; enum AVSampleFormat out_fmt; int out_rate; int num_formats, num_rates, num_layouts; int i, j, k, l, m, n; num_formats = 2; num_rates = 2; num_layouts = 2; if (argc > 1) { if (!av_strncasecmp(argv[1], "-h", 3)) { av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> " "[<num sample rates> [<num channel layouts>]]]\n" "Default is 2 2 2\n"); return 0; } num_formats = strtol(argv[1], NULL, 0); num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats)); } if (argc > 2) { num_rates = strtol(argv[2], NULL, 0); num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates)); } if (argc > 3) { num_layouts = strtol(argv[3], NULL, 0); num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts)); } av_log_set_level(AV_LOG_DEBUG); av_lfg_init(&rnd, 0xC0FFEE); in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6, AV_SAMPLE_FMT_DBLP, 0); out_buf_size = in_buf_size; in_buf = av_malloc(in_buf_size); if (!in_buf) goto end; out_buf = av_malloc(out_buf_size); if (!out_buf) goto end; s = avresample_alloc_context(); if (!s) { av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n"); ret = 1; goto end; } for (i = 0; i < num_formats; i++) { in_fmt = formats[i]; for (k = 0; k < num_layouts; k++) { in_ch_layout = layouts[k]; in_channels = av_get_channel_layout_nb_channels(in_ch_layout); for (m = 0; m < num_rates; m++) { in_rate = rates[m]; ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf, in_channels, in_rate * 6, in_fmt, 0); if (ret < 0) { av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n"); goto end; } audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6); for (j = 0; j < num_formats; j++) { out_fmt = formats[j]; for (l = 0; l < num_layouts; l++) { out_ch_layout = layouts[l]; out_channels = av_get_channel_layout_nb_channels(out_ch_layout); for (n = 0; n < num_rates; n++) { out_rate = rates[n]; av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n", av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt), in_channels, out_channels, in_rate, out_rate); ret = av_samples_fill_arrays(out_data, &out_linesize, out_buf, out_channels, out_rate * 6, out_fmt, 0); if (ret < 0) { av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n"); goto end; } av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0); av_opt_set_int(s, "in_sample_fmt", in_fmt, 0); av_opt_set_int(s, "in_sample_rate", in_rate, 0); av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0); av_opt_set_int(s, "out_sample_fmt", out_fmt, 0); av_opt_set_int(s, "out_sample_rate", out_rate, 0); av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); ret = avresample_open(s); if (ret < 0) { av_log(s, AV_LOG_ERROR, "Error opening context\n"); goto end; } ret = avresample_convert(s, out_data, out_linesize, out_rate * 6, in_data, in_linesize, in_rate * 6); if (ret < 0) { char errbuf[256]; av_strerror(ret, errbuf, sizeof(errbuf)); av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf); goto end; } av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n", in_rate * 6, ret); if (avresample_get_delay(s) > 0) av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n", avresample_get_delay(s)); if (avresample_available(s) > 0) av_log(NULL, AV_LOG_INFO, "%d samples available for output\n", avresample_available(s)); av_log(NULL, AV_LOG_INFO, "\n"); avresample_close(s); } } } } } } ret = 0; end: av_freep(&in_buf); av_freep(&out_buf); avresample_free(&s); return ret; }