annotate libmpcodecs/ad_liba52.c @ 17050:cd540846d052

Add support to get the list of files from a file containing one filename per line. Patch from devik (devik .at. cdi .dot. cz).
author albeu
date Sat, 26 Nov 2005 17:10:17 +0000
parents 6ff3379a0862
children befc16265bb1
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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1 #include <stdio.h>
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2 #include <stdlib.h>
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3 #include <unistd.h>
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4 #include <math.h>
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5 #include <assert.h>
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6
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7 #include "config.h"
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8 #ifdef USE_LIBA52
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9
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10 #include "mp_msg.h"
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11 #include "help_mp.h"
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12
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13 #include "ad_internal.h"
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14
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15 #include "cpudetect.h"
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16
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17 #include "libaf/af_format.h"
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18
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19 #include "liba52/a52.h"
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20 #include "liba52/mm_accel.h"
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21
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22 static sample_t * a52_samples;
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23 static a52_state_t a52_state;
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24 static uint32_t a52_flags=0;
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25 /** Used by a52_resample_float, it defines the mapping between liba52
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26 * channels and output channels. The ith nibble from the right in the
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27 * hex representation of channel_map is the index of the source
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28 * channel corresponding to the ith output channel. Source channels are
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29 * indexed 1-6. Silent output channels are marked by 0xf. */
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30 static uint32_t channel_map;
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31
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32 #define DRC_NO_ACTION 0
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33 #define DRC_NO_COMPRESSION 1
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34 #define DRC_CALLBACK 2
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35
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36 /** The output is multiplied by this var. Used for volume control */
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37 static sample_t a52_level = 1;
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38 /** The value of the -a52drc switch. */
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39 float a52_drc_level = 1.0;
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40 static int a52_drc_action = DRC_NO_ACTION;
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41
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42 #include "bswap.h"
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43
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44 static ad_info_t info =
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45 {
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46 "AC3 decoding with liba52",
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47 "liba52",
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48 "Nick Kurshev",
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49 "Michel LESPINASSE",
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50 ""
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51 };
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52
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53 LIBAD_EXTERN(liba52)
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54
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55 extern int audio_output_channels;
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56
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57 int a52_fillbuff(sh_audio_t *sh_audio){
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58 int length=0;
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59 int flags=0;
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60 int sample_rate=0;
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61 int bit_rate=0;
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62
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63 sh_audio->a_in_buffer_len=0;
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64 /* sync frame:*/
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65 while(1){
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66 while(sh_audio->a_in_buffer_len<8){
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67 int c=demux_getc(sh_audio->ds);
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68 if(c<0) return -1; /* EOF*/
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69 sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
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70 }
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71 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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72 length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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73 if(length>=7 && length<=3840) break; /* we're done.*/
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74 /* bad file => resync*/
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75 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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76 memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
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77 --sh_audio->a_in_buffer_len;
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78 }
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79 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate);
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80 sh_audio->samplerate=sample_rate;
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81 sh_audio->i_bps=bit_rate/8;
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82 sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
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83 demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
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84 if(sh_audio->format!=0x2000)
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85 swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
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86
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87 if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
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88 mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
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89
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90 return length;
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91 }
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92
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93 /* returns: number of available channels*/
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94 static int a52_printinfo(sh_audio_t *sh_audio){
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95 int flags, sample_rate, bit_rate;
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96 char* mode="unknown";
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97 int channels=0;
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98 a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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99 switch(flags&A52_CHANNEL_MASK){
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100 case A52_CHANNEL: mode="channel"; channels=2; break;
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101 case A52_MONO: mode="mono"; channels=1; break;
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102 case A52_STEREO: mode="stereo"; channels=2; break;
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103 case A52_3F: mode="3f";channels=3;break;
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104 case A52_2F1R: mode="2f+1r";channels=3;break;
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105 case A52_3F1R: mode="3f+1r";channels=4;break;
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106 case A52_2F2R: mode="2f+2r";channels=4;break;
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107 case A52_3F2R: mode="3f+2r";channels=5;break;
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108 case A52_CHANNEL1: mode="channel1"; channels=2; break;
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109 case A52_CHANNEL2: mode="channel2"; channels=2; break;
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110 case A52_DOLBY: mode="dolby"; channels=2; break;
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111 }
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112 mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
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113 channels, (flags&A52_LFE)?1:0,
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114 mode, (flags&A52_LFE)?"+lfe":"",
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115 sample_rate, bit_rate*0.001f);
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116 return (flags&A52_LFE) ? (channels+1) : channels;
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117 }
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118
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119 sample_t dynrng_call (sample_t c, void *data) {
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120 // fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
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121 return pow((double)c, a52_drc_level);
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122 }
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123
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124
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125 static int preinit(sh_audio_t *sh)
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126 {
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127 /* Dolby AC3 audio: */
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128 /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
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129 if (sh->samplesize < 2) sh->samplesize = 2;
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130 sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
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b3d1348b251f audio input buffer allocation/free cleanup
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131 sh->audio_in_minsize=3840;
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132 a52_level = 1.0;
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133 return 1;
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134 }
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135
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136 /**
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137 * \brief Function to convert the "planar" float format used by liba52
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138 * into the interleaved float format used by libaf/libao2.
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139 * \param in the input buffer containing the planar samples.
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140 * \param out the output buffer where the interleaved result is stored.
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141 */
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142 static int a52_resample_float(float *in, int16_t *out)
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143 {
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144 unsigned long i;
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145 float *p = (float*) out;
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146 for (i = 0; i != 256; i++) {
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147 unsigned long map = channel_map;
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148 do {
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149 unsigned long ch = map & 15;
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150 if (ch == 15)
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151 *p = 0;
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152 else
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153 *p = in[i + ((ch-1)<<8)];
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154 p++;
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155 } while ((map >>= 4));
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156 }
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157 return (int16_t*) p - out;
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158 }
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159
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160 static int init(sh_audio_t *sh_audio)
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161 {
5341
6c1d264cc214 do not export the World
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162 uint32_t a52_accel=0;
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163 sample_t level=a52_level, bias=384;
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164 int flags=0;
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165 /* Dolby AC3 audio:*/
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166 if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
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167 if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
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168 if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
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169 if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
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170 if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
9003
c428933c7e54 AltiVec detection code ("borrowed" from FFmpeg and
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171 if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
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172 a52_samples=a52_init (a52_accel);
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173 if (a52_samples == NULL) {
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174 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
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175 return 0;
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176 }
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177 if(a52_fillbuff(sh_audio)<0){
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178 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
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179 return 0;
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180 }
12285
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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181
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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182
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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183 /* Init a52 dynrng */
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184 if (a52_drc_level < 0.001) {
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185 /* level == 0 --> no compression, init library without callback */
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186 a52_drc_action = DRC_NO_COMPRESSION;
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187 } else if (a52_drc_level > 0.999) {
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188 /* level == 1 --> full compression, do nothing at all (library default = full compression) */
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189 a52_drc_action = DRC_NO_ACTION;
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190 } else {
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191 a52_drc_action = DRC_CALLBACK;
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192 }
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193 /* Library init for dynrng has to be done for each frame, see decode_audio() */
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194
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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diff changeset
195
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196 /* 'a52 cannot upmix' hotfix:*/
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197 a52_printinfo(sh_audio);
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198 sh_audio->channels=audio_output_channels;
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diff changeset
199 while(sh_audio->channels>0){
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200 switch(sh_audio->channels){
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diff changeset
201 case 1: a52_flags=A52_MONO; break;
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202 /* case 2: a52_flags=A52_STEREO; break;*/
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203 case 2: a52_flags=A52_DOLBY; break;
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204 /* case 3: a52_flags=A52_3F; break;*/
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205 case 3: a52_flags=A52_2F1R; break;
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206 case 4: a52_flags=A52_2F2R; break; /* 2+2*/
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207 case 5: a52_flags=A52_3F2R; break;
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208 case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
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209 }
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210 /* test:*/
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211 flags=a52_flags|A52_ADJUST_LEVEL;
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212 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
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213 if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
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diff changeset
214 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
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diff changeset
215 return 0;
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diff changeset
216 }
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217 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
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diff changeset
218 /* frame decoded, let's init resampler:*/
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219 channel_map = 0;
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220 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
221 if (!(flags & A52_LFE)) {
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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222 switch ((flags<<3) | sh_audio->channels) {
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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223 case (A52_MONO << 3) | 1: channel_map = 0x1; break;
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224 case (A52_CHANNEL << 3) | 2:
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225 case (A52_STEREO << 3) | 2:
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226 case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
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227 case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
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228 case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
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229 case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
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230 case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
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diff changeset
231 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
232 } else if (sh_audio->channels == 6) {
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233 switch (flags & ~A52_LFE) {
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diff changeset
234 case A52_MONO : channel_map = 0x12ffff; break;
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diff changeset
235 case A52_CHANNEL:
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diff changeset
236 case A52_STEREO :
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237 case A52_DOLBY : channel_map = 0x1fff32; break;
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238 case A52_3F : channel_map = 0x13ff42; break;
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239 case A52_2F1R : channel_map = 0x1f4432; break;
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240 case A52_2F2R : channel_map = 0x1f5432; break;
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241 case A52_3F2R : channel_map = 0x136542; break;
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diff changeset
242 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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243 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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244 if (channel_map) {
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245 a52_resample = a52_resample_float;
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246 break;
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
247 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
248 } else
5340
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diff changeset
249 if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
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diff changeset
250 --sh_audio->channels; /* try to decrease no. of channels*/
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diff changeset
251 }
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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diff changeset
252 if(sh_audio->channels<=0){
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253 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
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254 return 0;
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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parents:
diff changeset
255 }
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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diff changeset
256 return 1;
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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parents:
diff changeset
257 }
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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258
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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diff changeset
259 static void uninit(sh_audio_t *sh)
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diff changeset
260 {
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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parents:
diff changeset
261 }
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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parents:
diff changeset
262
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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263 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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264 {
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265 switch(cmd)
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266 {
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267 case ADCTRL_RESYNC_STREAM:
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268 case ADCTRL_SKIP_FRAME:
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269 a52_fillbuff(sh);
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270 return CONTROL_TRUE;
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271 case ADCTRL_SET_VOLUME: {
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272 float vol = *(float*)arg;
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273 if (vol > 60.0) vol = 60.0;
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274 a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
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275 return CONTROL_TRUE;
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276 }
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277 case ADCTRL_QUERY_FORMAT:
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278 if (*(int*)arg == AF_FORMAT_S16_NE ||
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279 *(int*)arg == AF_FORMAT_FLOAT_NE)
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280 return CONTROL_TRUE;
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281 return CONTROL_FALSE;
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282 }
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283 return CONTROL_UNKNOWN;
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284 }
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285
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286 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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287 {
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288 sample_t level=a52_level, bias=384;
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289 int flags=a52_flags|A52_ADJUST_LEVEL;
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290 int i,len=-1;
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291 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
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292 bias = 0;
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293 if(!sh_audio->a_in_buffer_len)
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294 if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
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295 sh_audio->a_in_buffer_len=0;
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296 if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
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297 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
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298 return len;
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299 }
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300
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301 /* handle dynrng */
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302 if (a52_drc_action != DRC_NO_ACTION) {
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303 if (a52_drc_action == DRC_NO_COMPRESSION)
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304 a52_dynrng(&a52_state, NULL, NULL);
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305 else
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306 a52_dynrng(&a52_state, dynrng_call, NULL);
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307 }
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308
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309 len=0;
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310 for (i = 0; i < 6; i++) {
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311 if (a52_block (&a52_state, a52_samples)){
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312 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
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313 break;
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314 }
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315 len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]);
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316 }
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317 assert(len <= maxlen);
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318 return len;
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319 }
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320 #endif