forked from KolibriOS/kolibrios
342 lines
12 KiB
C
342 lines
12 KiB
C
/*
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* Copyright (c) 2002 Fabrice Bellard
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <stdint.h>
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#include <stdio.h>
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#include "libavutil/avstring.h"
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#include "libavutil/common.h"
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#include "libavutil/lfg.h"
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#include "libavutil/libm.h"
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#include "libavutil/log.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avresample.h"
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static double dbl_rand(AVLFG *lfg)
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{
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return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
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}
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#define PUT_FUNC(name, fmt, type, expr) \
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static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
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int channels, int sample, int ch, \
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double v_dbl) \
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{ \
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type v = expr; \
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type **out = (type **)data; \
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if (av_sample_fmt_is_planar(sample_fmt)) \
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out[ch][sample] = v; \
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else \
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out[0][sample * channels + ch] = v; \
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}
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PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
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PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
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PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
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PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
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PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
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static void put_sample(void **data, enum AVSampleFormat sample_fmt,
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int channels, int sample, int ch, double v_dbl)
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{
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switch (av_get_packed_sample_fmt(sample_fmt)) {
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case AV_SAMPLE_FMT_U8:
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put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
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break;
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case AV_SAMPLE_FMT_S16:
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put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
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break;
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case AV_SAMPLE_FMT_S32:
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put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
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break;
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case AV_SAMPLE_FMT_FLT:
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put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
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break;
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case AV_SAMPLE_FMT_DBL:
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put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
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break;
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}
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}
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static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
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int channels, int sample_rate, int nb_samples)
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{
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int i, ch, k;
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double v, f, a, ampa;
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double tabf1[AVRESAMPLE_MAX_CHANNELS];
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double tabf2[AVRESAMPLE_MAX_CHANNELS];
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double taba[AVRESAMPLE_MAX_CHANNELS];
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#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
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k = 0;
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/* 1 second of single freq sine at 1000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
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v = sin(a) * 0.30;
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for (ch = 0; ch < channels; ch++)
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PUT_SAMPLE
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a += M_PI * 1000.0 * 2.0 / sample_rate;
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}
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/* 1 second of varying frequency between 100 and 10000 Hz */
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a = 0;
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for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
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v = sin(a) * 0.30;
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for (ch = 0; ch < channels; ch++)
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PUT_SAMPLE
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f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
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a += M_PI * f * 2.0 / sample_rate;
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}
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/* 0.5 second of low amplitude white noise */
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for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
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v = dbl_rand(rnd) * 0.30;
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for (ch = 0; ch < channels; ch++)
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PUT_SAMPLE
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}
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/* 0.5 second of high amplitude white noise */
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for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
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v = dbl_rand(rnd);
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for (ch = 0; ch < channels; ch++)
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PUT_SAMPLE
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}
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/* 1 second of unrelated ramps for each channel */
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for (ch = 0; ch < channels; ch++) {
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taba[ch] = 0;
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tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
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tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
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}
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for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
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for (ch = 0; ch < channels; ch++) {
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v = sin(taba[ch]) * 0.30;
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PUT_SAMPLE
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f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
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taba[ch] += M_PI * f * 2.0 / sample_rate;
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}
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}
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/* 2 seconds of 500 Hz with varying volume */
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a = 0;
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ampa = 0;
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for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
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for (ch = 0; ch < channels; ch++) {
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double amp = (1.0 + sin(ampa)) * 0.15;
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if (ch & 1)
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amp = 0.30 - amp;
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v = sin(a) * amp;
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PUT_SAMPLE
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a += M_PI * 500.0 * 2.0 / sample_rate;
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ampa += M_PI * 2.0 / sample_rate;
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}
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}
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}
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/* formats, rates, and layouts are ordered for priority in testing.
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e.g. 'avresample-test 4 2 2' will test all input/output combinations of
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S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
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static const enum AVSampleFormat formats[] = {
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_S32,
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AV_SAMPLE_FMT_U8P,
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AV_SAMPLE_FMT_U8,
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_DBL,
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};
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static const int rates[] = {
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48000,
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44100,
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16000
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};
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static const uint64_t layouts[] = {
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AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_5POINT1,
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AV_CH_LAYOUT_7POINT1,
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};
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int main(int argc, char **argv)
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{
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AVAudioResampleContext *s;
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AVLFG rnd;
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int ret = 0;
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uint8_t *in_buf = NULL;
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uint8_t *out_buf = NULL;
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unsigned int in_buf_size;
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unsigned int out_buf_size;
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uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
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uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
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int in_linesize;
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int out_linesize;
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uint64_t in_ch_layout;
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int in_channels;
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enum AVSampleFormat in_fmt;
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int in_rate;
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uint64_t out_ch_layout;
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int out_channels;
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enum AVSampleFormat out_fmt;
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int out_rate;
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int num_formats, num_rates, num_layouts;
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int i, j, k, l, m, n;
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num_formats = 2;
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num_rates = 2;
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num_layouts = 2;
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if (argc > 1) {
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if (!av_strncasecmp(argv[1], "-h", 3)) {
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av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
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"[<num sample rates> [<num channel layouts>]]]\n"
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"Default is 2 2 2\n");
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return 0;
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}
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num_formats = strtol(argv[1], NULL, 0);
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num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
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}
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if (argc > 2) {
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num_rates = strtol(argv[2], NULL, 0);
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num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
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}
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if (argc > 3) {
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num_layouts = strtol(argv[3], NULL, 0);
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num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
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}
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av_log_set_level(AV_LOG_DEBUG);
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av_lfg_init(&rnd, 0xC0FFEE);
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in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
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AV_SAMPLE_FMT_DBLP, 0);
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out_buf_size = in_buf_size;
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in_buf = av_malloc(in_buf_size);
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if (!in_buf)
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goto end;
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out_buf = av_malloc(out_buf_size);
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if (!out_buf)
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goto end;
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s = avresample_alloc_context();
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if (!s) {
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av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
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ret = 1;
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goto end;
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}
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for (i = 0; i < num_formats; i++) {
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in_fmt = formats[i];
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for (k = 0; k < num_layouts; k++) {
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in_ch_layout = layouts[k];
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in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
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for (m = 0; m < num_rates; m++) {
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in_rate = rates[m];
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ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
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in_channels, in_rate * 6,
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in_fmt, 0);
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if (ret < 0) {
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av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
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goto end;
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}
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audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
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for (j = 0; j < num_formats; j++) {
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out_fmt = formats[j];
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for (l = 0; l < num_layouts; l++) {
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out_ch_layout = layouts[l];
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out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
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for (n = 0; n < num_rates; n++) {
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out_rate = rates[n];
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av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
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av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
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in_channels, out_channels, in_rate, out_rate);
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ret = av_samples_fill_arrays(out_data, &out_linesize,
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out_buf, out_channels,
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out_rate * 6, out_fmt, 0);
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if (ret < 0) {
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av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
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goto end;
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}
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av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
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av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
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av_opt_set_int(s, "in_sample_rate", in_rate, 0);
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av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
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av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
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av_opt_set_int(s, "out_sample_rate", out_rate, 0);
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av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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ret = avresample_open(s);
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if (ret < 0) {
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av_log(s, AV_LOG_ERROR, "Error opening context\n");
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goto end;
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}
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ret = avresample_convert(s, out_data, out_linesize, out_rate * 6,
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in_data, in_linesize, in_rate * 6);
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if (ret < 0) {
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char errbuf[256];
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av_strerror(ret, errbuf, sizeof(errbuf));
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av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
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goto end;
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}
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av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
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in_rate * 6, ret);
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if (avresample_get_delay(s) > 0)
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av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
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avresample_get_delay(s));
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if (avresample_available(s) > 0)
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av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
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avresample_available(s));
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av_log(NULL, AV_LOG_INFO, "\n");
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avresample_close(s);
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}
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}
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}
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}
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}
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}
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ret = 0;
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end:
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av_freep(&in_buf);
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av_freep(&out_buf);
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avresample_free(&s);
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return ret;
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}
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