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342 lines
12 KiB
342 lines
12 KiB
/* |
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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 "libavresample/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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