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4349 | Serge | 1 | /* |
2 | * Copyright (c) 2002 Fabrice Bellard |
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3 | * Copyright (c) 2012 Justin Ruggles |
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4 | * |
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5 | * This file is part of FFmpeg. |
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6 | * |
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7 | * FFmpeg is free software; you can redistribute it and/or |
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8 | * modify it under the terms of the GNU Lesser General Public |
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9 | * License as published by the Free Software Foundation; either |
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10 | * version 2.1 of the License, or (at your option) any later version. |
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11 | * |
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12 | * FFmpeg is distributed in the hope that it will be useful, |
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13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 | * Lesser General Public License for more details. |
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16 | * |
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17 | * You should have received a copy of the GNU Lesser General Public |
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18 | * License along with FFmpeg; if not, write to the Free Software |
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19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 | */ |
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21 | |||
22 | #include |
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23 | #include |
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24 | |||
25 | #include "libavutil/avstring.h" |
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26 | #include "libavutil/common.h" |
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27 | #include "libavutil/lfg.h" |
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28 | #include "libavutil/libm.h" |
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29 | #include "libavutil/log.h" |
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30 | #include "libavutil/mem.h" |
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31 | #include "libavutil/opt.h" |
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32 | #include "libavutil/samplefmt.h" |
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33 | #include "avresample.h" |
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34 | |||
35 | static double dbl_rand(AVLFG *lfg) |
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36 | { |
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37 | return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0; |
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38 | } |
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39 | |||
40 | #define PUT_FUNC(name, fmt, type, expr) \ |
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41 | static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\ |
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42 | int channels, int sample, int ch, \ |
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43 | double v_dbl) \ |
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44 | { \ |
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45 | type v = expr; \ |
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46 | type **out = (type **)data; \ |
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47 | if (av_sample_fmt_is_planar(sample_fmt)) \ |
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48 | out[ch][sample] = v; \ |
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49 | else \ |
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50 | out[0][sample * channels + ch] = v; \ |
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51 | } |
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52 | |||
53 | PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128)) |
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54 | PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15)))) |
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55 | PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31)))) |
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56 | PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl) |
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57 | PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl) |
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58 | |||
59 | static void put_sample(void **data, enum AVSampleFormat sample_fmt, |
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60 | int channels, int sample, int ch, double v_dbl) |
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61 | { |
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62 | switch (av_get_packed_sample_fmt(sample_fmt)) { |
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63 | case AV_SAMPLE_FMT_U8: |
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64 | put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl); |
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65 | break; |
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66 | case AV_SAMPLE_FMT_S16: |
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67 | put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl); |
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68 | break; |
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69 | case AV_SAMPLE_FMT_S32: |
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70 | put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl); |
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71 | break; |
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72 | case AV_SAMPLE_FMT_FLT: |
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73 | put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl); |
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74 | break; |
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75 | case AV_SAMPLE_FMT_DBL: |
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76 | put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl); |
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77 | break; |
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78 | } |
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79 | } |
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80 | |||
81 | static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt, |
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82 | int channels, int sample_rate, int nb_samples) |
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83 | { |
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84 | int i, ch, k; |
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85 | double v, f, a, ampa; |
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86 | double tabf1[AVRESAMPLE_MAX_CHANNELS]; |
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87 | double tabf2[AVRESAMPLE_MAX_CHANNELS]; |
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88 | double taba[AVRESAMPLE_MAX_CHANNELS]; |
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89 | |||
90 | #define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v); |
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91 | |||
92 | k = 0; |
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93 | |||
94 | /* 1 second of single freq sine at 1000 Hz */ |
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95 | a = 0; |
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96 | for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { |
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97 | v = sin(a) * 0.30; |
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98 | for (ch = 0; ch < channels; ch++) |
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99 | PUT_SAMPLE |
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100 | a += M_PI * 1000.0 * 2.0 / sample_rate; |
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101 | } |
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102 | |||
103 | /* 1 second of varying frequency between 100 and 10000 Hz */ |
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104 | a = 0; |
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105 | for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { |
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106 | v = sin(a) * 0.30; |
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107 | for (ch = 0; ch < channels; ch++) |
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108 | PUT_SAMPLE |
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109 | f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate); |
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110 | a += M_PI * f * 2.0 / sample_rate; |
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111 | } |
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112 | |||
113 | /* 0.5 second of low amplitude white noise */ |
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114 | for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { |
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115 | v = dbl_rand(rnd) * 0.30; |
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116 | for (ch = 0; ch < channels; ch++) |
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117 | PUT_SAMPLE |
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118 | } |
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119 | |||
120 | /* 0.5 second of high amplitude white noise */ |
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121 | for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { |
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122 | v = dbl_rand(rnd); |
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123 | for (ch = 0; ch < channels; ch++) |
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124 | PUT_SAMPLE |
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125 | } |
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126 | |||
127 | /* 1 second of unrelated ramps for each channel */ |
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128 | for (ch = 0; ch < channels; ch++) { |
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129 | taba[ch] = 0; |
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130 | tabf1[ch] = 100 + av_lfg_get(rnd) % 5000; |
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131 | tabf2[ch] = 100 + av_lfg_get(rnd) % 5000; |
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132 | } |
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133 | for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { |
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134 | for (ch = 0; ch < channels; ch++) { |
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135 | v = sin(taba[ch]) * 0.30; |
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136 | PUT_SAMPLE |
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137 | f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate); |
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138 | taba[ch] += M_PI * f * 2.0 / sample_rate; |
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139 | } |
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140 | } |
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141 | |||
142 | /* 2 seconds of 500 Hz with varying volume */ |
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143 | a = 0; |
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144 | ampa = 0; |
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145 | for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) { |
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146 | for (ch = 0; ch < channels; ch++) { |
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147 | double amp = (1.0 + sin(ampa)) * 0.15; |
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148 | if (ch & 1) |
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149 | amp = 0.30 - amp; |
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150 | v = sin(a) * amp; |
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151 | PUT_SAMPLE |
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152 | a += M_PI * 500.0 * 2.0 / sample_rate; |
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153 | ampa += M_PI * 2.0 / sample_rate; |
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154 | } |
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155 | } |
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156 | } |
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157 | |||
158 | /* formats, rates, and layouts are ordered for priority in testing. |
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159 | e.g. 'avresample-test 4 2 2' will test all input/output combinations of |
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160 | S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */ |
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161 | |||
162 | static const enum AVSampleFormat formats[] = { |
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163 | AV_SAMPLE_FMT_S16, |
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164 | AV_SAMPLE_FMT_FLTP, |
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165 | AV_SAMPLE_FMT_S16P, |
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166 | AV_SAMPLE_FMT_FLT, |
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167 | AV_SAMPLE_FMT_S32P, |
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168 | AV_SAMPLE_FMT_S32, |
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169 | AV_SAMPLE_FMT_U8P, |
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170 | AV_SAMPLE_FMT_U8, |
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171 | AV_SAMPLE_FMT_DBLP, |
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172 | AV_SAMPLE_FMT_DBL, |
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173 | }; |
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174 | |||
175 | static const int rates[] = { |
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176 | 48000, |
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177 | 44100, |
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178 | 16000 |
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179 | }; |
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180 | |||
181 | static const uint64_t layouts[] = { |
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182 | AV_CH_LAYOUT_STEREO, |
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183 | AV_CH_LAYOUT_MONO, |
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184 | AV_CH_LAYOUT_5POINT1, |
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185 | AV_CH_LAYOUT_7POINT1, |
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186 | }; |
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187 | |||
188 | int main(int argc, char **argv) |
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189 | { |
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190 | AVAudioResampleContext *s; |
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191 | AVLFG rnd; |
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192 | int ret = 0; |
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193 | uint8_t *in_buf = NULL; |
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194 | uint8_t *out_buf = NULL; |
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195 | unsigned int in_buf_size; |
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196 | unsigned int out_buf_size; |
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197 | uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; |
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198 | uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; |
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199 | int in_linesize; |
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200 | int out_linesize; |
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201 | uint64_t in_ch_layout; |
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202 | int in_channels; |
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203 | enum AVSampleFormat in_fmt; |
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204 | int in_rate; |
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205 | uint64_t out_ch_layout; |
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206 | int out_channels; |
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207 | enum AVSampleFormat out_fmt; |
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208 | int out_rate; |
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209 | int num_formats, num_rates, num_layouts; |
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210 | int i, j, k, l, m, n; |
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211 | |||
212 | num_formats = 2; |
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213 | num_rates = 2; |
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214 | num_layouts = 2; |
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215 | if (argc > 1) { |
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216 | if (!av_strncasecmp(argv[1], "-h", 3)) { |
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217 | av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [ |
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218 | "[ |
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219 | "Default is 2 2 2\n"); |
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220 | return 0; |
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221 | } |
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222 | num_formats = strtol(argv[1], NULL, 0); |
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223 | num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats)); |
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224 | } |
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225 | if (argc > 2) { |
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226 | num_rates = strtol(argv[2], NULL, 0); |
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227 | num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates)); |
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228 | } |
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229 | if (argc > 3) { |
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230 | num_layouts = strtol(argv[3], NULL, 0); |
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231 | num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts)); |
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232 | } |
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233 | |||
234 | av_log_set_level(AV_LOG_DEBUG); |
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235 | |||
236 | av_lfg_init(&rnd, 0xC0FFEE); |
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237 | |||
238 | in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6, |
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239 | AV_SAMPLE_FMT_DBLP, 0); |
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240 | out_buf_size = in_buf_size; |
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241 | |||
242 | in_buf = av_malloc(in_buf_size); |
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243 | if (!in_buf) |
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244 | goto end; |
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245 | out_buf = av_malloc(out_buf_size); |
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246 | if (!out_buf) |
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247 | goto end; |
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248 | |||
249 | s = avresample_alloc_context(); |
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250 | if (!s) { |
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251 | av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n"); |
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252 | ret = 1; |
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253 | goto end; |
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254 | } |
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255 | |||
256 | for (i = 0; i < num_formats; i++) { |
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257 | in_fmt = formats[i]; |
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258 | for (k = 0; k < num_layouts; k++) { |
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259 | in_ch_layout = layouts[k]; |
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260 | in_channels = av_get_channel_layout_nb_channels(in_ch_layout); |
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261 | for (m = 0; m < num_rates; m++) { |
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262 | in_rate = rates[m]; |
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263 | |||
264 | ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf, |
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265 | in_channels, in_rate * 6, |
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266 | in_fmt, 0); |
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267 | if (ret < 0) { |
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268 | av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n"); |
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269 | goto end; |
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270 | } |
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271 | audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6); |
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272 | |||
273 | for (j = 0; j < num_formats; j++) { |
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274 | out_fmt = formats[j]; |
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275 | for (l = 0; l < num_layouts; l++) { |
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276 | out_ch_layout = layouts[l]; |
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277 | out_channels = av_get_channel_layout_nb_channels(out_ch_layout); |
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278 | for (n = 0; n < num_rates; n++) { |
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279 | out_rate = rates[n]; |
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280 | |||
281 | av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n", |
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282 | av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt), |
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283 | in_channels, out_channels, in_rate, out_rate); |
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284 | |||
285 | ret = av_samples_fill_arrays(out_data, &out_linesize, |
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286 | out_buf, out_channels, |
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287 | out_rate * 6, out_fmt, 0); |
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288 | if (ret < 0) { |
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289 | av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n"); |
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290 | goto end; |
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291 | } |
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292 | |||
293 | av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0); |
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294 | av_opt_set_int(s, "in_sample_fmt", in_fmt, 0); |
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295 | av_opt_set_int(s, "in_sample_rate", in_rate, 0); |
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296 | av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0); |
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297 | av_opt_set_int(s, "out_sample_fmt", out_fmt, 0); |
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298 | av_opt_set_int(s, "out_sample_rate", out_rate, 0); |
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299 | |||
300 | av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); |
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301 | |||
302 | ret = avresample_open(s); |
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303 | if (ret < 0) { |
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304 | av_log(s, AV_LOG_ERROR, "Error opening context\n"); |
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305 | goto end; |
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306 | } |
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307 | |||
308 | ret = avresample_convert(s, out_data, out_linesize, out_rate * 6, |
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309 | in_data, in_linesize, in_rate * 6); |
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310 | if (ret < 0) { |
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311 | char errbuf[256]; |
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312 | av_strerror(ret, errbuf, sizeof(errbuf)); |
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313 | av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf); |
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314 | goto end; |
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315 | } |
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316 | av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n", |
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317 | in_rate * 6, ret); |
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318 | if (avresample_get_delay(s) > 0) |
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319 | av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n", |
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320 | avresample_get_delay(s)); |
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321 | if (avresample_available(s) > 0) |
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322 | av_log(NULL, AV_LOG_INFO, "%d samples available for output\n", |
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323 | avresample_available(s)); |
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324 | av_log(NULL, AV_LOG_INFO, "\n"); |
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325 | |||
326 | avresample_close(s); |
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327 | } |
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328 | } |
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329 | } |
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330 | } |
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331 | } |
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332 | } |
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333 | |||
334 | ret = 0; |
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335 | |||
336 | end: |
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337 | av_freep(&in_buf); |
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338 | av_freep(&out_buf); |
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339 | avresample_free(&s); |
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340 | return ret; |
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341 | }>>>>>>>>>>>>>>>>>>>>>>>>>>>>>><>><>><> |