Mercurial > mplayer.hg
annotate libmpcodecs/ad_liba52.c @ 15255:e9b4eeb31768
synced with 1.73
author | gabrov |
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date | Sun, 24 Apr 2005 22:16:09 +0000 |
parents | 878745349274 |
children | 3aee7daaa8e5 |
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 |
5341 | 22 static sample_t * a52_samples; |
23 static a52_state_t a52_state; | |
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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-afm/-vfm help implemenetd, some cosmetics of ad/vd codec names/comments
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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 |
5343 | 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){ |
10119 | 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 } |
10119 | 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*/ |
10119 | 75 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8); |
76 memcpy(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; |
10119 | 83 demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8); |
84 if(sh_audio->format!=0x2000) | |
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 sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6; |
5458 | 130 sh->audio_in_minsize=3840; |
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131 a52_level = 1.0; |
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132 return 1; |
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133 } |
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134 |
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135 /** |
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136 * \brief Function to convert the "planar" float format used by liba52 |
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137 * into the interleaved float format used by libaf/libao2. |
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138 * \param in the input buffer containing the planar samples. |
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139 * \param out the output buffer where the interleaved result is stored. |
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140 */ |
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141 static int a52_resample_float(float *in, int16_t *out) |
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142 { |
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143 unsigned long i; |
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144 float *p = (float*) out; |
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145 for (i = 0; i != 256; i++) { |
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146 unsigned long map = channel_map; |
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147 do { |
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148 unsigned long ch = map & 15; |
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149 if (ch == 15) |
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150 *p = 0; |
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151 else |
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152 *p = in[i + ((ch-1)<<8)]; |
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153 p++; |
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154 } while ((map >>= 4)); |
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155 } |
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156 return (int16_t*) p - out; |
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157 } |
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158 |
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159 static int init(sh_audio_t *sh_audio) |
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160 { |
5341 | 161 uint32_t a52_accel=0; |
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162 sample_t level=a52_level, bias=384; |
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163 int flags=0; |
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164 /* Dolby AC3 audio:*/ |
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165 if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE; |
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166 if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX; |
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167 if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT; |
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168 if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW; |
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169 if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT; |
9003 | 170 if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC; |
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171 a52_samples=a52_init (a52_accel); |
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172 if (a52_samples == NULL) { |
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173 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); |
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174 return 0; |
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175 } |
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176 if(a52_fillbuff(sh_audio)<0){ |
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177 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); |
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178 return 0; |
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179 } |
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180 |
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181 |
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182 /* Init a52 dynrng */ |
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183 if (a52_drc_level < 0.001) { |
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184 /* level == 0 --> no compression, init library without callback */ |
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185 a52_drc_action = DRC_NO_COMPRESSION; |
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186 } else if (a52_drc_level > 0.999) { |
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187 /* level == 1 --> full compression, do nothing at all (library default = full compression) */ |
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188 a52_drc_action = DRC_NO_ACTION; |
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189 } else { |
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190 a52_drc_action = DRC_CALLBACK; |
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191 } |
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192 /* Library init for dynrng has to be done for each frame, see decode_audio() */ |
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193 |
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194 |
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195 /* 'a52 cannot upmix' hotfix:*/ |
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196 a52_printinfo(sh_audio); |
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197 sh_audio->channels=audio_output_channels; |
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198 while(sh_audio->channels>0){ |
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199 switch(sh_audio->channels){ |
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200 case 1: a52_flags=A52_MONO; break; |
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201 /* case 2: a52_flags=A52_STEREO; break;*/ |
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202 case 2: a52_flags=A52_DOLBY; break; |
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203 /* case 3: a52_flags=A52_3F; break;*/ |
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204 case 3: a52_flags=A52_2F1R; break; |
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205 case 4: a52_flags=A52_2F2R; break; /* 2+2*/ |
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206 case 5: a52_flags=A52_3F2R; break; |
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207 case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/ |
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208 } |
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209 /* test:*/ |
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210 flags=a52_flags|A52_ADJUST_LEVEL; |
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211 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags); |
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212 if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ |
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213 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n"); |
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214 return 0; |
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215 } |
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216 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags); |
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217 /* frame decoded, let's init resampler:*/ |
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218 channel_map = 0; |
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219 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) { |
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220 if (!(flags & A52_LFE)) { |
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221 switch ((flags<<3) | sh_audio->channels) { |
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222 case (A52_MONO << 3) | 1: channel_map = 0x1; break; |
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223 case (A52_CHANNEL << 3) | 2: |
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224 case (A52_STEREO << 3) | 2: |
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225 case (A52_DOLBY << 3) | 2: channel_map = 0x21; break; |
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226 case (A52_2F1R << 3) | 3: channel_map = 0x321; break; |
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227 case (A52_2F2R << 3) | 4: channel_map = 0x4321; break; |
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228 case (A52_3F << 3) | 5: channel_map = 0x2ff31; break; |
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229 case (A52_3F2R << 3) | 5: channel_map = 0x25431; break; |
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230 } |
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231 } else if (sh_audio->channels == 6) { |
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232 switch (flags & ~A52_LFE) { |
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233 case A52_MONO : channel_map = 0x12ffff; break; |
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234 case A52_CHANNEL: |
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235 case A52_STEREO : |
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236 case A52_DOLBY : channel_map = 0x1fff32; break; |
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237 case A52_3F : channel_map = 0x13ff42; break; |
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238 case A52_2F1R : channel_map = 0x1f4432; break; |
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239 case A52_2F2R : channel_map = 0x1f5432; break; |
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240 case A52_3F2R : channel_map = 0x136542; break; |
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241 } |
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242 } |
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243 if (channel_map) { |
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244 a52_resample = a52_resample_float; |
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245 break; |
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246 } |
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247 } else |
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248 if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break; |
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249 --sh_audio->channels; /* try to decrease no. of channels*/ |
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250 } |
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251 if(sh_audio->channels<=0){ |
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252 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n"); |
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253 return 0; |
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254 } |
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255 return 1; |
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256 } |
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257 |
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258 static void uninit(sh_audio_t *sh) |
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259 { |
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260 } |
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261 |
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262 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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263 { |
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264 switch(cmd) |
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265 { |
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266 case ADCTRL_SKIP_FRAME: |
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267 a52_fillbuff(sh); break; // skip AC3 frame |
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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 } |
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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, a52_samples)){ |
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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 } |
7127 | 313 len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]); |
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314 } |
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315 assert(len <= maxlen); |
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316 return len; |
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317 } |
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318 #endif |