Mercurial > mplayer.hg
annotate libmpcodecs/ad_liba52.c @ 29269:4d9de809b174
Add a hack to detect when we are writing into a Windows pipe since the fseek
incorrectly does not fail like it should.
This ensures we will not incorrectly append the file header at the end.
Based on patch by Zhou Zongyi [zhouzongyi at pset.suntec.net]
author | reimar |
---|---|
date | Sat, 16 May 2009 13:59:53 +0000 |
parents | 0f1b5b68af32 |
children | bbb6ebec87a0 |
rev | line source |
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Give _XOPEN_SOURCE #define an explicit 600 value. Fixes build on Open Solaris.
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1 #define _XOPEN_SOURCE 600 |
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2 #include <stdio.h> |
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3 #include <stdlib.h> |
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4 #include <unistd.h> |
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5 #include <math.h> |
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6 #include <assert.h> |
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7 |
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8 #include "config.h" |
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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 #include "mpbswap.h" |
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13 |
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14 #include "ad_internal.h" |
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15 |
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16 #include "cpudetect.h" |
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17 |
17012 | 18 #include "libaf/af_format.h" |
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19 |
27588 | 20 #ifdef CONFIG_LIBA52_INTERNAL |
17012 | 21 #include "liba52/a52.h" |
22 #include "liba52/mm_accel.h" | |
27588 | 23 #else |
24 #include <a52dec/a52.h> | |
25 #include <a52dec/mm_accel.h> | |
26 int (* a52_resample) (float * _f, int16_t * s16); | |
27 #endif | |
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28 |
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sync with liba52 0.7.4, patch by Emanuele Giaquinta >emanuele.giaquinta ! gmail * com<
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29 static a52_state_t *a52_state; |
5341 | 30 static uint32_t a52_flags=0; |
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31 /** Used by a52_resample_float, it defines the mapping between liba52 |
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32 * channels and output channels. The ith nibble from the right in the |
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33 * hex representation of channel_map is the index of the source |
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34 * channel corresponding to the ith output channel. Source channels are |
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35 * indexed 1-6. Silent output channels are marked by 0xf. */ |
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36 static uint32_t channel_map; |
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37 |
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38 #define DRC_NO_ACTION 0 |
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39 #define DRC_NO_COMPRESSION 1 |
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40 #define DRC_CALLBACK 2 |
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41 |
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42 /** The output is multiplied by this var. Used for volume control */ |
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43 static sample_t a52_level = 1; |
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44 /** The value of the -a52drc switch. */ |
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45 float a52_drc_level = 1.0; |
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46 static int a52_drc_action = DRC_NO_ACTION; |
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47 |
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48 static ad_info_t info = |
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49 { |
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-afm/-vfm help implemenetd, some cosmetics of ad/vd codec names/comments
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50 "AC3 decoding with liba52", |
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51 "liba52", |
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52 "Nick Kurshev", |
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53 "Michel LESPINASSE", |
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54 "" |
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55 }; |
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56 |
5343 | 57 LIBAD_EXTERN(liba52) |
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58 |
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59 int a52_fillbuff(sh_audio_t *sh_audio){ |
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60 int length=0; |
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61 int flags=0; |
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62 int sample_rate=0; |
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63 int bit_rate=0; |
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64 |
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65 sh_audio->a_in_buffer_len=0; |
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66 /* sync frame:*/ |
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67 while(1){ |
10119 | 68 while(sh_audio->a_in_buffer_len<8){ |
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69 int c=demux_getc(sh_audio->ds); |
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70 if(c<0) return -1; /* EOF*/ |
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71 sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c; |
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72 } |
10119 | 73 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8); |
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74 length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); |
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75 if(length>=7 && length<=3840) break; /* we're done.*/ |
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76 /* bad file => resync*/ |
10119 | 77 if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8); |
15560 | 78 memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7); |
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79 --sh_audio->a_in_buffer_len; |
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80 } |
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81 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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82 sh_audio->samplerate=sample_rate; |
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83 sh_audio->i_bps=bit_rate/8; |
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84 sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2; |
10119 | 85 demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8); |
86 if(sh_audio->format!=0x2000) | |
87 swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8); | |
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88 |
27588 | 89 #ifdef CONFIG_LIBA52_INTERNAL |
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90 if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0) |
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91 mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n"); |
27588 | 92 #endif |
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93 |
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94 return length; |
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95 } |
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96 |
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97 /* returns: number of available channels*/ |
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98 static int a52_printinfo(sh_audio_t *sh_audio){ |
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99 int flags, sample_rate, bit_rate; |
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100 char* mode="unknown"; |
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101 int channels=0; |
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102 a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); |
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103 switch(flags&A52_CHANNEL_MASK){ |
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104 case A52_CHANNEL: mode="channel"; channels=2; break; |
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105 case A52_MONO: mode="mono"; channels=1; break; |
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106 case A52_STEREO: mode="stereo"; channels=2; break; |
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107 case A52_3F: mode="3f";channels=3;break; |
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108 case A52_2F1R: mode="2f+1r";channels=3;break; |
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109 case A52_3F1R: mode="3f+1r";channels=4;break; |
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110 case A52_2F2R: mode="2f+2r";channels=4;break; |
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111 case A52_3F2R: mode="3f+2r";channels=5;break; |
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112 case A52_CHANNEL1: mode="channel1"; channels=2; break; |
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113 case A52_CHANNEL2: mode="channel2"; channels=2; break; |
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114 case A52_DOLBY: mode="dolby"; channels=2; break; |
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115 } |
19944 | 116 mp_msg(MSGT_DECAUDIO,MSGL_V,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n", |
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117 channels, (flags&A52_LFE)?1:0, |
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118 mode, (flags&A52_LFE)?"+lfe":"", |
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119 sample_rate, bit_rate*0.001f); |
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120 return (flags&A52_LFE) ? (channels+1) : channels; |
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121 } |
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122 |
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123 sample_t dynrng_call (sample_t c, void *data) { |
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124 // fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level)); |
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125 return pow((double)c, a52_drc_level); |
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126 } |
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127 |
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128 |
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129 static int preinit(sh_audio_t *sh) |
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130 { |
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131 /* Dolby AC3 audio: */ |
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132 /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */ |
27588 | 133 #ifdef CONFIG_LIBA52_INTERNAL |
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134 if (sh->samplesize < 2) sh->samplesize = 2; |
27588 | 135 #else |
136 if (sh->samplesize < 4) sh->samplesize = 4; | |
137 #endif | |
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138 sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6; |
5458 | 139 sh->audio_in_minsize=3840; |
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140 a52_level = 1.0; |
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141 return 1; |
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142 } |
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143 |
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144 /** |
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145 * \brief Function to convert the "planar" float format used by liba52 |
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146 * into the interleaved float format used by libaf/libao2. |
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147 * \param in the input buffer containing the planar samples. |
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148 * \param out the output buffer where the interleaved result is stored. |
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149 */ |
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150 static int a52_resample_float(float *in, int16_t *out) |
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151 { |
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152 unsigned long i; |
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153 float *p = (float*) out; |
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154 for (i = 0; i != 256; i++) { |
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155 unsigned long map = channel_map; |
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156 do { |
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157 unsigned long ch = map & 15; |
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158 if (ch == 15) |
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159 *p = 0; |
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160 else |
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161 *p = in[i + ((ch-1)<<8)]; |
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162 p++; |
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163 } while ((map >>= 4)); |
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164 } |
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165 return (int16_t*) p - out; |
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166 } |
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167 |
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168 static int init(sh_audio_t *sh_audio) |
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169 { |
5341 | 170 uint32_t a52_accel=0; |
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171 sample_t level=a52_level, bias=384; |
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172 int flags=0; |
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173 /* Dolby AC3 audio:*/ |
27587 | 174 #ifdef MM_ACCEL_X86_SSE |
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175 if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE; |
27587 | 176 #endif |
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177 if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX; |
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178 if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT; |
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179 if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW; |
27587 | 180 #ifdef MM_ACCEL_X86_3DNOWEXT |
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181 if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT; |
27587 | 182 #endif |
183 #ifdef MM_ACCEL_PPC_ALTIVEC | |
9003 | 184 if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC; |
27587 | 185 #endif |
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186 a52_state=a52_init (a52_accel); |
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187 if (a52_state == NULL) { |
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188 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); |
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189 return 0; |
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190 } |
27588 | 191 #ifndef CONFIG_LIBA52_INTERNAL |
192 sh_audio->sample_format = AF_FORMAT_FLOAT_NE; | |
193 #endif | |
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194 if(a52_fillbuff(sh_audio)<0){ |
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195 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); |
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196 return 0; |
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197 } |
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198 |
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199 |
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200 /* Init a52 dynrng */ |
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201 if (a52_drc_level < 0.001) { |
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202 /* level == 0 --> no compression, init library without callback */ |
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203 a52_drc_action = DRC_NO_COMPRESSION; |
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204 } else if (a52_drc_level > 0.999) { |
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205 /* level == 1 --> full compression, do nothing at all (library default = full compression) */ |
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206 a52_drc_action = DRC_NO_ACTION; |
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207 } else { |
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208 a52_drc_action = DRC_CALLBACK; |
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209 } |
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210 /* Library init for dynrng has to be done for each frame, see decode_audio() */ |
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211 |
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212 |
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213 /* 'a52 cannot upmix' hotfix:*/ |
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214 a52_printinfo(sh_audio); |
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215 sh_audio->channels=audio_output_channels; |
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216 while(sh_audio->channels>0){ |
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217 switch(sh_audio->channels){ |
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218 case 1: a52_flags=A52_MONO; break; |
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219 /* case 2: a52_flags=A52_STEREO; break;*/ |
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220 case 2: a52_flags=A52_DOLBY; break; |
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221 /* case 3: a52_flags=A52_3F; break;*/ |
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222 case 3: a52_flags=A52_2F1R; break; |
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223 case 4: a52_flags=A52_2F2R; break; /* 2+2*/ |
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224 case 5: a52_flags=A52_3F2R; break; |
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225 case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/ |
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226 } |
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227 /* test:*/ |
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228 flags=a52_flags|A52_ADJUST_LEVEL; |
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229 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags); |
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230 if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ |
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231 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n"); |
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232 return 0; |
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233 } |
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234 mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags); |
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235 /* frame decoded, let's init resampler:*/ |
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236 channel_map = 0; |
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237 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) { |
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238 if (!(flags & A52_LFE)) { |
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239 switch ((flags<<3) | sh_audio->channels) { |
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240 case (A52_MONO << 3) | 1: channel_map = 0x1; break; |
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241 case (A52_CHANNEL << 3) | 2: |
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242 case (A52_STEREO << 3) | 2: |
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243 case (A52_DOLBY << 3) | 2: channel_map = 0x21; break; |
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244 case (A52_2F1R << 3) | 3: channel_map = 0x321; break; |
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245 case (A52_2F2R << 3) | 4: channel_map = 0x4321; break; |
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246 case (A52_3F << 3) | 5: channel_map = 0x2ff31; break; |
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247 case (A52_3F2R << 3) | 5: channel_map = 0x25431; break; |
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248 } |
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249 } else if (sh_audio->channels == 6) { |
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250 switch (flags & ~A52_LFE) { |
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251 case A52_MONO : channel_map = 0x12ffff; break; |
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252 case A52_CHANNEL: |
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253 case A52_STEREO : |
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254 case A52_DOLBY : channel_map = 0x1fff32; break; |
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255 case A52_3F : channel_map = 0x13ff42; break; |
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256 case A52_2F1R : channel_map = 0x1f4432; break; |
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257 case A52_2F2R : channel_map = 0x1f5432; break; |
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258 case A52_3F2R : channel_map = 0x136542; break; |
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259 } |
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260 } |
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261 if (channel_map) { |
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262 a52_resample = a52_resample_float; |
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263 break; |
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264 } |
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265 } else |
27588 | 266 #ifdef CONFIG_LIBA52_INTERNAL |
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267 if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break; |
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268 --sh_audio->channels; /* try to decrease no. of channels*/ |
27588 | 269 #else |
270 break; | |
271 #endif | |
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272 } |
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273 if(sh_audio->channels<=0){ |
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274 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n"); |
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275 return 0; |
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276 } |
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277 return 1; |
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278 } |
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279 |
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280 static void uninit(sh_audio_t *sh) |
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281 { |
26923 | 282 a52_free(a52_state); |
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283 } |
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284 |
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285 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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286 { |
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287 switch(cmd) |
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288 { |
15558 | 289 case ADCTRL_RESYNC_STREAM: |
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290 case ADCTRL_SKIP_FRAME: |
15558 | 291 a52_fillbuff(sh); |
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292 return CONTROL_TRUE; |
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293 case ADCTRL_SET_VOLUME: { |
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294 float vol = *(float*)arg; |
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295 if (vol > 60.0) vol = 60.0; |
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296 a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0); |
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297 return CONTROL_TRUE; |
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298 } |
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299 case ADCTRL_QUERY_FORMAT: |
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300 if (*(int*)arg == AF_FORMAT_S16_NE || |
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301 *(int*)arg == AF_FORMAT_FLOAT_NE) |
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302 return CONTROL_TRUE; |
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303 return CONTROL_FALSE; |
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304 } |
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305 return CONTROL_UNKNOWN; |
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306 } |
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307 |
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308 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) |
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309 { |
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310 sample_t level=a52_level, bias=384; |
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311 int flags=a52_flags|A52_ADJUST_LEVEL; |
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312 int i,len=-1; |
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313 if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) |
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314 bias = 0; |
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315 if(!sh_audio->a_in_buffer_len) |
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316 if(a52_fillbuff(sh_audio)<0) return len; /* EOF */ |
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317 sh_audio->a_in_buffer_len=0; |
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318 if (a52_frame (a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ |
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319 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n"); |
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320 return len; |
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321 } |
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322 |
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323 /* handle dynrng */ |
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324 if (a52_drc_action != DRC_NO_ACTION) { |
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325 if (a52_drc_action == DRC_NO_COMPRESSION) |
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326 a52_dynrng(a52_state, NULL, NULL); |
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327 else |
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328 a52_dynrng(a52_state, dynrng_call, NULL); |
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329 } |
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330 |
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331 len=0; |
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332 for (i = 0; i < 6; i++) { |
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333 if (a52_block (a52_state)){ |
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334 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n"); |
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335 break; |
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336 } |
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337 len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]); |
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338 } |
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339 assert(len <= maxlen); |
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340 return len; |
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341 } |