diff options
author | Clément Bœsch <clement@stupeflix.com> | 2016-06-22 14:44:29 +0300 |
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committer | Clément Bœsch <clement@stupeflix.com> | 2016-06-22 14:44:34 +0300 |
commit | 5d48e4eafa6c4559683892b8638d10508125f3cf (patch) | |
tree | 0d5becbd08d6ae319497dd32cbe9b67f4670cc5e /libavresample/tests | |
parent | 989a614b707dcff8abdffe28dc24ec64a83b2837 (diff) | |
parent | a6a750c7ef240b72ce01e9653343a0ddf247d196 (diff) |
Merge commit 'a6a750c7ef240b72ce01e9653343a0ddf247d196'
* commit 'a6a750c7ef240b72ce01e9653343a0ddf247d196':
tests: Move all test programs to a subdirectory
Merged-by: Clément Bœsch <clement@stupeflix.com>
Diffstat (limited to 'libavresample/tests')
-rw-r--r-- | libavresample/tests/.gitignore | 1 | ||||
-rw-r--r-- | libavresample/tests/avresample.c | 342 |
2 files changed, 343 insertions, 0 deletions
diff --git a/libavresample/tests/.gitignore b/libavresample/tests/.gitignore new file mode 100644 index 0000000000..1e15871d54 --- /dev/null +++ b/libavresample/tests/.gitignore @@ -0,0 +1 @@ +/avresample diff --git a/libavresample/tests/avresample.c b/libavresample/tests/avresample.c new file mode 100644 index 0000000000..8c377bae84 --- /dev/null +++ b/libavresample/tests/avresample.c @@ -0,0 +1,342 @@ +/* + * 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 "libavresample/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; +} |