annotate lib/libswresample/swresample.h @ 2575:a76d408c5132 tip

DrawTranslucent -> DrawTextureGrayShade Removed old texture drawing stuff
author a.parshin
date Wed, 09 Mar 2016 01:39:52 +0200
parents 992d2e6f907d
children
rev   line source
2134
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1 /*
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2 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
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3 *
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4 * This file is part of libswresample
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5 *
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6 * libswresample is free software; you can redistribute it and/or
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7 * modify it under the terms of the GNU Lesser General Public
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8 * License as published by the Free Software Foundation; either
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9 * version 2.1 of the License, or (at your option) any later version.
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10 *
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11 * libswresample is distributed in the hope that it will be useful,
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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14 * Lesser General Public License for more details.
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15 *
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16 * You should have received a copy of the GNU Lesser General Public
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17 * License along with libswresample; if not, write to the Free Software
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18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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19 */
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20
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21 #ifndef SWRESAMPLE_SWRESAMPLE_H
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22 #define SWRESAMPLE_SWRESAMPLE_H
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23
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24 /**
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25 * @file
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26 * @ingroup lswr
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27 * libswresample public header
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28 */
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29
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30 /**
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31 * @defgroup lswr Libswresample
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32 * @{
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33 *
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34 * Libswresample (lswr) is a library that handles audio resampling, sample
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35 * format conversion and mixing.
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36 *
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37 * Interaction with lswr is done through SwrContext, which is
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38 * allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
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39 * must be set with the @ref avoptions API.
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40 *
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41 * For example the following code will setup conversion from planar float sample
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42 * format to interleaved signed 16-bit integer, downsampling from 48kHz to
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43 * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
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44 * matrix):
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45 * @code
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46 * SwrContext *swr = swr_alloc();
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47 * av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
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48 * av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
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49 * av_opt_set_int(swr, "in_sample_rate", 48000, 0);
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50 * av_opt_set_int(swr, "out_sample_rate", 44100, 0);
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51 * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
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52 * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
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53 * @endcode
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54 *
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55 * Once all values have been set, it must be initialized with swr_init(). If
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56 * you need to change the conversion parameters, you can change the parameters
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57 * as described above, or by using swr_alloc_set_opts(), then call swr_init()
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58 * again.
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59 *
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60 * The conversion itself is done by repeatedly calling swr_convert().
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61 * Note that the samples may get buffered in swr if you provide insufficient
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62 * output space or if sample rate conversion is done, which requires "future"
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63 * samples. Samples that do not require future input can be retrieved at any
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64 * time by using swr_convert() (in_count can be set to 0).
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65 * At the end of conversion the resampling buffer can be flushed by calling
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66 * swr_convert() with NULL in and 0 in_count.
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67 *
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68 * The delay between input and output, can at any time be found by using
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69 * swr_get_delay().
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70 *
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71 * The following code demonstrates the conversion loop assuming the parameters
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72 * from above and caller-defined functions get_input() and handle_output():
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73 * @code
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74 * uint8_t **input;
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75 * int in_samples;
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76 *
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77 * while (get_input(&input, &in_samples)) {
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78 * uint8_t *output;
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79 * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
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80 * in_samples, 44100, 48000, AV_ROUND_UP);
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81 * av_samples_alloc(&output, NULL, 2, out_samples,
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82 * AV_SAMPLE_FMT_S16, 0);
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83 * out_samples = swr_convert(swr, &output, out_samples,
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84 * input, in_samples);
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85 * handle_output(output, out_samples);
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86 * av_freep(&output);
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87 * }
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88 * @endcode
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89 *
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90 * When the conversion is finished, the conversion
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91 * context and everything associated with it must be freed with swr_free().
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92 * There will be no memory leak if the data is not completely flushed before
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93 * swr_free().
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94 */
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95
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96 #include <stdint.h>
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97 #include "lib/libavutil/samplefmt.h"
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98
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99 #include "lib/libswresample/version.h"
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100
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101 #if LIBSWRESAMPLE_VERSION_MAJOR < 1
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102 #define SWR_CH_MAX 32 ///< Maximum number of channels
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103 #endif
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104
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105 #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
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106 //TODO use int resample ?
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107 //long term TODO can we enable this dynamically?
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108
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109 enum SwrDitherType {
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110 SWR_DITHER_NONE = 0,
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111 SWR_DITHER_RECTANGULAR,
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112 SWR_DITHER_TRIANGULAR,
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113 SWR_DITHER_TRIANGULAR_HIGHPASS,
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114
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115 SWR_DITHER_NS = 64, ///< not part of API/ABI
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116 SWR_DITHER_NS_LIPSHITZ,
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117 SWR_DITHER_NS_F_WEIGHTED,
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118 SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
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119 SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
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120 SWR_DITHER_NS_SHIBATA,
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121 SWR_DITHER_NS_LOW_SHIBATA,
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122 SWR_DITHER_NS_HIGH_SHIBATA,
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123 SWR_DITHER_NB, ///< not part of API/ABI
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124 };
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125
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126 /** Resampling Engines */
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127 enum SwrEngine {
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128 SWR_ENGINE_SWR, /**< SW Resampler */
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129 SWR_ENGINE_SOXR, /**< SoX Resampler */
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130 SWR_ENGINE_NB, ///< not part of API/ABI
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131 };
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132
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133 /** Resampling Filter Types */
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134 enum SwrFilterType {
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135 SWR_FILTER_TYPE_CUBIC, /**< Cubic */
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136 SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
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137 SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
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138 };
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139
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140 typedef struct SwrContext SwrContext;
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141
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142 /**
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143 * Get the AVClass for swrContext. It can be used in combination with
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144 * AV_OPT_SEARCH_FAKE_OBJ for examining options.
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145 *
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146 * @see av_opt_find().
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147 */
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148 const AVClass *swr_get_class(void);
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149
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150 /**
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151 * Allocate SwrContext.
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152 *
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153 * If you use this function you will need to set the parameters (manually or
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154 * with swr_alloc_set_opts()) before calling swr_init().
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155 *
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156 * @see swr_alloc_set_opts(), swr_init(), swr_free()
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157 * @return NULL on error, allocated context otherwise
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158 */
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159 struct SwrContext *swr_alloc(void);
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160
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161 /**
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162 * Initialize context after user parameters have been set.
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163 *
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164 * @return AVERROR error code in case of failure.
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165 */
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166 int swr_init(struct SwrContext *s);
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167
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168 /**
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169 * Allocate SwrContext if needed and set/reset common parameters.
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170 *
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171 * This function does not require s to be allocated with swr_alloc(). On the
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172 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
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173 * on the allocated context.
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174 *
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175 * @param s Swr context, can be NULL
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176 * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
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177 * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
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178 * @param out_sample_rate output sample rate (frequency in Hz)
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179 * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
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180 * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
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181 * @param in_sample_rate input sample rate (frequency in Hz)
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182 * @param log_offset logging level offset
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183 * @param log_ctx parent logging context, can be NULL
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184 *
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185 * @see swr_init(), swr_free()
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186 * @return NULL on error, allocated context otherwise
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187 */
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188 struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
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189 int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
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190 int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
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191 int log_offset, void *log_ctx);
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192
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193 /**
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194 * Free the given SwrContext and set the pointer to NULL.
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195 */
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196 void swr_free(struct SwrContext **s);
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197
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198 /**
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199 * Convert audio.
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200 *
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201 * in and in_count can be set to 0 to flush the last few samples out at the
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202 * end.
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203 *
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204 * If more input is provided than output space then the input will be buffered.
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205 * You can avoid this buffering by providing more output space than input.
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206 * Convertion will run directly without copying whenever possible.
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207 *
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208 * @param s allocated Swr context, with parameters set
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209 * @param out output buffers, only the first one need be set in case of packed audio
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210 * @param out_count amount of space available for output in samples per channel
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211 * @param in input buffers, only the first one need to be set in case of packed audio
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212 * @param in_count number of input samples available in one channel
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213 *
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214 * @return number of samples output per channel, negative value on error
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215 */
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216 int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
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217 const uint8_t **in , int in_count);
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218
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219 /**
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220 * Convert the next timestamp from input to output
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221 * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
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222 *
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223 * @note There are 2 slightly differently behaving modes.
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224 * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
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225 * in this case timestamps will be passed through with delays compensated
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226 * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
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227 * in this case the output timestamps will match output sample numbers
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228 *
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229 * @param pts timestamp for the next input sample, INT64_MIN if unknown
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230 * @return the output timestamp for the next output sample
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231 */
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232 int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
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233
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234 /**
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235 * Activate resampling compensation.
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236 */
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237 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
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238
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239 /**
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240 * Set a customized input channel mapping.
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241 *
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242 * @param s allocated Swr context, not yet initialized
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243 * @param channel_map customized input channel mapping (array of channel
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244 * indexes, -1 for a muted channel)
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245 * @return AVERROR error code in case of failure.
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246 */
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247 int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
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248
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249 /**
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250 * Set a customized remix matrix.
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251 *
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252 * @param s allocated Swr context, not yet initialized
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253 * @param matrix remix coefficients; matrix[i + stride * o] is
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254 * the weight of input channel i in output channel o
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255 * @param stride offset between lines of the matrix
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256 * @return AVERROR error code in case of failure.
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257 */
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258 int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
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259
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260 /**
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261 * Drops the specified number of output samples.
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262 */
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263 int swr_drop_output(struct SwrContext *s, int count);
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264
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265 /**
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266 * Injects the specified number of silence samples.
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267 */
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268 int swr_inject_silence(struct SwrContext *s, int count);
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269
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270 /**
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271 * Gets the delay the next input sample will experience relative to the next output sample.
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272 *
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273 * Swresample can buffer data if more input has been provided than available
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274 * output space, also converting between sample rates needs a delay.
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275 * This function returns the sum of all such delays.
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276 * The exact delay is not necessarily an integer value in either input or
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277 * output sample rate. Especially when downsampling by a large value, the
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278 * output sample rate may be a poor choice to represent the delay, similarly
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279 * for upsampling and the input sample rate.
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280 *
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281 * @param s swr context
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282 * @param base timebase in which the returned delay will be
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283 * if its set to 1 the returned delay is in seconds
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284 * if its set to 1000 the returned delay is in milli seconds
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285 * if its set to the input sample rate then the returned delay is in input samples
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286 * if its set to the output sample rate then the returned delay is in output samples
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287 * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
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288 * @returns the delay in 1/base units.
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289 */
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290 int64_t swr_get_delay(struct SwrContext *s, int64_t base);
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291
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292 /**
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293 * Return the LIBSWRESAMPLE_VERSION_INT constant.
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294 */
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295 unsigned swresample_version(void);
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296
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297 /**
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298 * Return the swr build-time configuration.
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299 */
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300 const char *swresample_configuration(void);
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301
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302 /**
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303 * Return the swr license.
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304 */
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305 const char *swresample_license(void);
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306
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307 /**
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308 * @}
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309 */
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310
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311 #endif /* SWRESAMPLE_SWRESAMPLE_H */