annotate resample.c @ 11486:1bf7af4db35a libavcodec

Add ff_ prefix to dwt functions
author mru
date Sun, 14 Mar 2010 17:50:16 +0000
parents ff9ae6249d18
children 7dd2a45249a9
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1 /*
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2 * samplerate conversion for both audio and video
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3 * Copyright (c) 2000 Fabrice Bellard
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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
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22 /**
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23 * @file libavcodec/resample.c
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24 * samplerate conversion for both audio and video
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25 */
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26
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27 #include "avcodec.h"
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28 #include "audioconvert.h"
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29 #include "opt.h"
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30
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31 struct AVResampleContext;
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32
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33 static const char *context_to_name(void *ptr)
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34 {
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35 return "audioresample";
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36 }
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37
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38 static const AVOption options[] = {{NULL}};
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39 static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options };
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40
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41 struct ReSampleContext {
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42 struct AVResampleContext *resample_context;
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43 short *temp[2];
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44 int temp_len;
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45 float ratio;
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46 /* channel convert */
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47 int input_channels, output_channels, filter_channels;
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48 AVAudioConvert *convert_ctx[2];
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49 enum SampleFormat sample_fmt[2]; ///< input and output sample format
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50 unsigned sample_size[2]; ///< size of one sample in sample_fmt
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51 short *buffer[2]; ///< buffers used for conversion to S16
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52 unsigned buffer_size[2]; ///< sizes of allocated buffers
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53 };
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54
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55 /* n1: number of samples */
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56 static void stereo_to_mono(short *output, short *input, int n1)
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57 {
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58 short *p, *q;
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59 int n = n1;
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60
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61 p = input;
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62 q = output;
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63 while (n >= 4) {
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64 q[0] = (p[0] + p[1]) >> 1;
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65 q[1] = (p[2] + p[3]) >> 1;
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66 q[2] = (p[4] + p[5]) >> 1;
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67 q[3] = (p[6] + p[7]) >> 1;
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68 q += 4;
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69 p += 8;
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70 n -= 4;
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71 }
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72 while (n > 0) {
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73 q[0] = (p[0] + p[1]) >> 1;
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74 q++;
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75 p += 2;
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76 n--;
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77 }
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78 }
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79
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80 /* n1: number of samples */
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81 static void mono_to_stereo(short *output, short *input, int n1)
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82 {
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83 short *p, *q;
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84 int n = n1;
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85 int v;
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86
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87 p = input;
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88 q = output;
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89 while (n >= 4) {
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90 v = p[0]; q[0] = v; q[1] = v;
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91 v = p[1]; q[2] = v; q[3] = v;
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92 v = p[2]; q[4] = v; q[5] = v;
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93 v = p[3]; q[6] = v; q[7] = v;
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94 q += 8;
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95 p += 4;
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96 n -= 4;
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97 }
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98 while (n > 0) {
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99 v = p[0]; q[0] = v; q[1] = v;
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100 q += 2;
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101 p += 1;
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102 n--;
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103 }
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104 }
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105
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106 /* XXX: should use more abstract 'N' channels system */
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107 static void stereo_split(short *output1, short *output2, short *input, int n)
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108 {
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109 int i;
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110
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111 for(i=0;i<n;i++) {
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112 *output1++ = *input++;
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113 *output2++ = *input++;
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114 }
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115 }
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116
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117 static void stereo_mux(short *output, short *input1, short *input2, int n)
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118 {
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119 int i;
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120
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121 for(i=0;i<n;i++) {
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122 *output++ = *input1++;
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123 *output++ = *input2++;
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124 }
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125 }
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126
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127 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
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128 {
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129 int i;
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130 short l,r;
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131
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132 for(i=0;i<n;i++) {
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133 l=*input1++;
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134 r=*input2++;
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135 *output++ = l; /* left */
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136 *output++ = (l/2)+(r/2); /* center */
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137 *output++ = r; /* right */
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138 *output++ = 0; /* left surround */
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139 *output++ = 0; /* right surroud */
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140 *output++ = 0; /* low freq */
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141 }
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142 }
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143
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144 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
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145 int output_rate, int input_rate,
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146 enum SampleFormat sample_fmt_out,
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147 enum SampleFormat sample_fmt_in,
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148 int filter_length, int log2_phase_count,
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149 int linear, double cutoff)
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150 {
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151 ReSampleContext *s;
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152
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153 if ( input_channels > 2)
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154 {
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155 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
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156 return NULL;
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157 }
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158
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159 s = av_mallocz(sizeof(ReSampleContext));
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160 if (!s)
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161 {
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162 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
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163 return NULL;
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164 }
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165
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166 s->ratio = (float)output_rate / (float)input_rate;
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167
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168 s->input_channels = input_channels;
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169 s->output_channels = output_channels;
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170
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171 s->filter_channels = s->input_channels;
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172 if (s->output_channels < s->filter_channels)
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173 s->filter_channels = s->output_channels;
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174
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175 s->sample_fmt [0] = sample_fmt_in;
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176 s->sample_fmt [1] = sample_fmt_out;
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177 s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
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178 s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
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179
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180 if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
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181 if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
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182 s->sample_fmt[0], 1, NULL, 0))) {
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183 av_log(s, AV_LOG_ERROR,
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184 "Cannot convert %s sample format to s16 sample format\n",
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185 avcodec_get_sample_fmt_name(s->sample_fmt[0]));
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186 av_free(s);
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187 return NULL;
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188 }
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189 }
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190
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191 if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
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192 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
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193 SAMPLE_FMT_S16, 1, NULL, 0))) {
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194 av_log(s, AV_LOG_ERROR,
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195 "Cannot convert s16 sample format to %s sample format\n",
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196 avcodec_get_sample_fmt_name(s->sample_fmt[1]));
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197 av_audio_convert_free(s->convert_ctx[0]);
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198 av_free(s);
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199 return NULL;
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200 }
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201 }
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202
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203 /*
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204 * AC-3 output is the only case where filter_channels could be greater than 2.
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205 * input channels can't be greater than 2, so resample the 2 channels and then
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206 * expand to 6 channels after the resampling.
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207 */
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208 if(s->filter_channels>2)
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209 s->filter_channels = 2;
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210
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211 #define TAPS 16
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212 s->resample_context= av_resample_init(output_rate, input_rate,
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213 filter_length, log2_phase_count, linear, cutoff);
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214
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215 *(const AVClass**)s->resample_context = &audioresample_context_class;
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216
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217 return s;
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218 }
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219
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220 #if LIBAVCODEC_VERSION_MAJOR < 53
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221 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
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222 int output_rate, int input_rate)
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223 {
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224 return av_audio_resample_init(output_channels, input_channels,
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225 output_rate, input_rate,
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226 SAMPLE_FMT_S16, SAMPLE_FMT_S16,
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227 TAPS, 10, 0, 0.8);
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228 }
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229 #endif
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230
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231 /* resample audio. 'nb_samples' is the number of input samples */
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232 /* XXX: optimize it ! */
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233 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
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234 {
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235 int i, nb_samples1;
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236 short *bufin[2];
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237 short *bufout[2];
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238 short *buftmp2[2], *buftmp3[2];
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239 short *output_bak = NULL;
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240 int lenout;
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241
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242 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
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243 /* nothing to do */
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244 memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
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245 return nb_samples;
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246 }
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247
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248 if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
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249 int istride[1] = { s->sample_size[0] };
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250 int ostride[1] = { 2 };
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251 const void *ibuf[1] = { input };
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252 void *obuf[1];
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253 unsigned input_size = nb_samples*s->input_channels*2;
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254
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255 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
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256 av_free(s->buffer[0]);
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257 s->buffer_size[0] = input_size;
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258 s->buffer[0] = av_malloc(s->buffer_size[0]);
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259 if (!s->buffer[0]) {
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260 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
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261 return 0;
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262 }
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263 }
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264
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265 obuf[0] = s->buffer[0];
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266
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267 if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
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268 ibuf, istride, nb_samples*s->input_channels) < 0) {
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269 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
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270 return 0;
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271 }
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272
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273 input = s->buffer[0];
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274 }
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275
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276 lenout= 4*nb_samples * s->ratio + 16;
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277
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278 if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
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279 output_bak = output;
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280
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281 if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
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282 av_free(s->buffer[1]);
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283 s->buffer_size[1] = lenout;
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284 s->buffer[1] = av_malloc(s->buffer_size[1]);
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285 if (!s->buffer[1]) {
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286 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
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287 return 0;
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288 }
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289 }
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290
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291 output = s->buffer[1];
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292 }
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293
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294 /* XXX: move those malloc to resample init code */
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295 for(i=0; i<s->filter_channels; i++){
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296 bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
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297 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
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298 buftmp2[i] = bufin[i] + s->temp_len;
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299 }
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300
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301 /* make some zoom to avoid round pb */
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302 bufout[0]= av_malloc( lenout * sizeof(short) );
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303 bufout[1]= av_malloc( lenout * sizeof(short) );
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304
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305 if (s->input_channels == 2 &&
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306 s->output_channels == 1) {
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307 buftmp3[0] = output;
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308 stereo_to_mono(buftmp2[0], input, nb_samples);
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309 } else if (s->output_channels >= 2 && s->input_channels == 1) {
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310 buftmp3[0] = bufout[0];
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311 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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312 } else if (s->output_channels >= 2) {
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313 buftmp3[0] = bufout[0];
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314 buftmp3[1] = bufout[1];
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315 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
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316 } else {
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317 buftmp3[0] = output;
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318 memcpy(buftmp2[0], input, nb_samples*sizeof(short));
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319 }
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320
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321 nb_samples += s->temp_len;
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322
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323 /* resample each channel */
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324 nb_samples1 = 0; /* avoid warning */
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325 for(i=0;i<s->filter_channels;i++) {
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326 int consumed;
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327 int is_last= i+1 == s->filter_channels;
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328
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329 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
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330 s->temp_len= nb_samples - consumed;
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331 s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
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332 memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
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333 }
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334
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335 if (s->output_channels == 2 && s->input_channels == 1) {
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336 mono_to_stereo(output, buftmp3[0], nb_samples1);
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337 } else if (s->output_channels == 2) {
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338 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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339 } else if (s->output_channels == 6) {
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340 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
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341 }
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342
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343 if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
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344 int istride[1] = { 2 };
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345 int ostride[1] = { s->sample_size[1] };
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346 const void *ibuf[1] = { output };
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347 void *obuf[1] = { output_bak };
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348
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349 if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
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350 ibuf, istride, nb_samples1*s->output_channels) < 0) {
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351 av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
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352 return 0;
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353 }
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354 }
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355
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356 for(i=0; i<s->filter_channels; i++)
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357 av_free(bufin[i]);
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358
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359 av_free(bufout[0]);
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360 av_free(bufout[1]);
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361 return nb_samples1;
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362 }
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363
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364 void audio_resample_close(ReSampleContext *s)
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365 {
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366 av_resample_close(s->resample_context);
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367 av_freep(&s->temp[0]);
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368 av_freep(&s->temp[1]);
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369 av_freep(&s->buffer[0]);
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370 av_freep(&s->buffer[1]);
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371 av_audio_convert_free(s->convert_ctx[0]);
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372 av_audio_convert_free(s->convert_ctx[1]);
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373 av_free(s);
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374 }