annotate libmpcodecs/ad_liba52.c @ 21055:bc2cf8eb55b3

Move subtitle updating to a separate function, fix inverted condition which caused timing of DVD subtitles to be ignored so that they were shown as soon as they were demuxed rather than in their timed position.
author uau
date Sun, 19 Nov 2006 17:55:38 +0000
parents 44193c0a688f
children 1767c271d710
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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b465ba5897a3 usage of libmpeg2, liba52, mp3lib & svq1 can be disabled
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8
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9 #include "mp_msg.h"
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10 #include "help_mp.h"
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11
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12 #include "ad_internal.h"
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13
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14 #include "cpudetect.h"
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15
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16 #include "libaf/af_format.h"
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17
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18 #include "liba52/a52.h"
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19 #include "liba52/mm_accel.h"
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20
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4bad7f00556e sync with liba52 0.7.4, patch by Emanuele Giaquinta >emanuele.giaquinta ! gmail * com<
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21 static a52_state_t *a52_state;
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22 static uint32_t a52_flags=0;
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23 /** Used by a52_resample_float, it defines the mapping between liba52
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24 * channels and output channels. The ith nibble from the right in the
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25 * hex representation of channel_map is the index of the source
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26 * channel corresponding to the ith output channel. Source channels are
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27 * indexed 1-6. Silent output channels are marked by 0xf. */
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28 static uint32_t channel_map;
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29
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30 #define DRC_NO_ACTION 0
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31 #define DRC_NO_COMPRESSION 1
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32 #define DRC_CALLBACK 2
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33
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34 /** The output is multiplied by this var. Used for volume control */
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35 static sample_t a52_level = 1;
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36 /** The value of the -a52drc switch. */
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37 float a52_drc_level = 1.0;
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38 static int a52_drc_action = DRC_NO_ACTION;
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39
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40 #include "bswap.h"
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41
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42 static ad_info_t info =
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43 {
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44 "AC3 decoding with liba52",
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45 "liba52",
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46 "Nick Kurshev",
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47 "Michel LESPINASSE",
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48 ""
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49 };
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50
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51 LIBAD_EXTERN(liba52)
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52
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53 extern int audio_output_channels;
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54
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55 int a52_fillbuff(sh_audio_t *sh_audio){
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56 int length=0;
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57 int flags=0;
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58 int sample_rate=0;
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59 int bit_rate=0;
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60
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61 sh_audio->a_in_buffer_len=0;
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62 /* sync frame:*/
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63 while(1){
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64 while(sh_audio->a_in_buffer_len<8){
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65 int c=demux_getc(sh_audio->ds);
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66 if(c<0) return -1; /* EOF*/
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67 sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c;
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68 }
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69 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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70 length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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71 if(length>=7 && length<=3840) break; /* we're done.*/
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72 /* bad file => resync*/
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73 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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74 memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
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75 --sh_audio->a_in_buffer_len;
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76 }
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77 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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78 sh_audio->samplerate=sample_rate;
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79 sh_audio->i_bps=bit_rate/8;
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80 sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
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81 demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
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82 if(sh_audio->format!=0x2000)
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83 swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
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84
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85 if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0)
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86 mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n");
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87
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88 return length;
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89 }
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90
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91 /* returns: number of available channels*/
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92 static int a52_printinfo(sh_audio_t *sh_audio){
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93 int flags, sample_rate, bit_rate;
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94 char* mode="unknown";
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95 int channels=0;
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96 a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
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97 switch(flags&A52_CHANNEL_MASK){
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98 case A52_CHANNEL: mode="channel"; channels=2; break;
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99 case A52_MONO: mode="mono"; channels=1; break;
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100 case A52_STEREO: mode="stereo"; channels=2; break;
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101 case A52_3F: mode="3f";channels=3;break;
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102 case A52_2F1R: mode="2f+1r";channels=3;break;
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103 case A52_3F1R: mode="3f+1r";channels=4;break;
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104 case A52_2F2R: mode="2f+2r";channels=4;break;
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105 case A52_3F2R: mode="3f+2r";channels=5;break;
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106 case A52_CHANNEL1: mode="channel1"; channels=2; break;
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107 case A52_CHANNEL2: mode="channel2"; channels=2; break;
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108 case A52_DOLBY: mode="dolby"; channels=2; break;
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109 }
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110 mp_msg(MSGT_DECAUDIO,MSGL_V,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n",
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111 channels, (flags&A52_LFE)?1:0,
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112 mode, (flags&A52_LFE)?"+lfe":"",
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113 sample_rate, bit_rate*0.001f);
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114 return (flags&A52_LFE) ? (channels+1) : channels;
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115 }
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116
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117 sample_t dynrng_call (sample_t c, void *data) {
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118 // fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
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119 return pow((double)c, a52_drc_level);
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120 }
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121
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122
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123 static int preinit(sh_audio_t *sh)
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124 {
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125 /* Dolby AC3 audio: */
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126 /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
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127 if (sh->samplesize < 2) sh->samplesize = 2;
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128 sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
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129 sh->audio_in_minsize=3840;
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130 a52_level = 1.0;
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131 return 1;
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132 }
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133
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134 /**
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135 * \brief Function to convert the "planar" float format used by liba52
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136 * into the interleaved float format used by libaf/libao2.
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137 * \param in the input buffer containing the planar samples.
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138 * \param out the output buffer where the interleaved result is stored.
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139 */
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140 static int a52_resample_float(float *in, int16_t *out)
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141 {
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142 unsigned long i;
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143 float *p = (float*) out;
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144 for (i = 0; i != 256; i++) {
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145 unsigned long map = channel_map;
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146 do {
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147 unsigned long ch = map & 15;
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148 if (ch == 15)
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149 *p = 0;
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150 else
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151 *p = in[i + ((ch-1)<<8)];
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152 p++;
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153 } while ((map >>= 4));
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154 }
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155 return (int16_t*) p - out;
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156 }
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157
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158 static int init(sh_audio_t *sh_audio)
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159 {
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6c1d264cc214 do not export the World
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160 uint32_t a52_accel=0;
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161 sample_t level=a52_level, bias=384;
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162 int flags=0;
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163 /* Dolby AC3 audio:*/
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164 if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE;
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165 if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX;
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166 if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT;
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167 if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW;
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168 if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT;
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169 if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
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170 a52_state=a52_init (a52_accel);
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171 if (a52_state == NULL) {
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172 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n");
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173 return 0;
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174 }
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175 if(a52_fillbuff(sh_audio)<0){
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176 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n");
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177 return 0;
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178 }
12285
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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179
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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180
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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181 /* Init a52 dynrng */
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182 if (a52_drc_level < 0.001) {
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183 /* level == 0 --> no compression, init library without callback */
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184 a52_drc_action = DRC_NO_COMPRESSION;
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185 } else if (a52_drc_level > 0.999) {
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186 /* level == 1 --> full compression, do nothing at all (library default = full compression) */
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187 a52_drc_action = DRC_NO_ACTION;
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188 } else {
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189 a52_drc_action = DRC_CALLBACK;
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190 }
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191 /* Library init for dynrng has to be done for each frame, see decode_audio() */
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192
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193
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194 /* 'a52 cannot upmix' hotfix:*/
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195 a52_printinfo(sh_audio);
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196 sh_audio->channels=audio_output_channels;
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197 while(sh_audio->channels>0){
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198 switch(sh_audio->channels){
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199 case 1: a52_flags=A52_MONO; break;
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200 /* case 2: a52_flags=A52_STEREO; break;*/
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201 case 2: a52_flags=A52_DOLBY; break;
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202 /* case 3: a52_flags=A52_3F; break;*/
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203 case 3: a52_flags=A52_2F1R; break;
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204 case 4: a52_flags=A52_2F2R; break; /* 2+2*/
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205 case 5: a52_flags=A52_3F2R; break;
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206 case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
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207 }
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208 /* test:*/
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209 flags=a52_flags|A52_ADJUST_LEVEL;
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210 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
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211 if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
5340
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212 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n");
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213 return 0;
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214 }
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215 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
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216 /* frame decoded, let's init resampler:*/
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217 channel_map = 0;
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218 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
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219 if (!(flags & A52_LFE)) {
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220 switch ((flags<<3) | sh_audio->channels) {
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221 case (A52_MONO << 3) | 1: channel_map = 0x1; break;
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222 case (A52_CHANNEL << 3) | 2:
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223 case (A52_STEREO << 3) | 2:
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224 case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
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225 case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
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226 case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
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227 case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
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228 case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
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229 }
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230 } else if (sh_audio->channels == 6) {
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diff changeset
231 switch (flags & ~A52_LFE) {
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232 case A52_MONO : channel_map = 0x12ffff; break;
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233 case A52_CHANNEL:
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234 case A52_STEREO :
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235 case A52_DOLBY : channel_map = 0x1fff32; break;
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236 case A52_3F : channel_map = 0x13ff42; break;
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237 case A52_2F1R : channel_map = 0x1f4432; break;
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238 case A52_2F2R : channel_map = 0x1f5432; break;
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239 case A52_3F2R : channel_map = 0x136542; break;
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240 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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241 }
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242 if (channel_map) {
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243 a52_resample = a52_resample_float;
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244 break;
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
245 }
878745349274 Add float output support. Add ADCTRL_QUERY_FORMAT control to report the
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diff changeset
246 } else
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247 if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
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248 --sh_audio->channels; /* try to decrease no. of channels*/
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249 }
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diff changeset
250 if(sh_audio->channels<=0){
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251 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n");
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252 return 0;
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parents:
diff changeset
253 }
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
254 return 1;
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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
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
257 static void uninit(sh_audio_t *sh)
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diff changeset
258 {
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parents:
diff changeset
259 }
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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260
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
261 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
0f12fb7c1c5d imported from MPlayerXP, dlopen() hack removed, some bugs fixed, interface functions changed to static, info->author field added
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262 {
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263 switch(cmd)
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264 {
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265 case ADCTRL_RESYNC_STREAM:
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266 case ADCTRL_SKIP_FRAME:
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267 a52_fillbuff(sh);
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268 return CONTROL_TRUE;
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269 case ADCTRL_SET_VOLUME: {
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270 float vol = *(float*)arg;
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271 if (vol > 60.0) vol = 60.0;
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272 a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
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273 return CONTROL_TRUE;
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274 }
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275 case ADCTRL_QUERY_FORMAT:
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276 if (*(int*)arg == AF_FORMAT_S16_NE ||
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277 *(int*)arg == AF_FORMAT_FLOAT_NE)
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278 return CONTROL_TRUE;
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279 return CONTROL_FALSE;
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280 }
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281 return CONTROL_UNKNOWN;
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282 }
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283
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284 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
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285 {
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286 sample_t level=a52_level, bias=384;
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287 int flags=a52_flags|A52_ADJUST_LEVEL;
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288 int i,len=-1;
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289 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
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290 bias = 0;
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291 if(!sh_audio->a_in_buffer_len)
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292 if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
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293 sh_audio->a_in_buffer_len=0;
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294 if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){
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295 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n");
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296 return len;
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297 }
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298
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299 /* handle dynrng */
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300 if (a52_drc_action != DRC_NO_ACTION) {
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301 if (a52_drc_action == DRC_NO_COMPRESSION)
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302 a52_dynrng(a52_state, NULL, NULL);
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303 else
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304 a52_dynrng(a52_state, dynrng_call, NULL);
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305 }
67b8d64af8c7 a52 dynamic range compression support by Peter Gansterer
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306
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307 len=0;
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308 for (i = 0; i < 6; i++) {
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309 if (a52_block (a52_state)){
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310 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
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311 break;
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312 }
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313 len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]);
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314 }
14817
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315 assert(len <= maxlen);
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316 return len;
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317 }