Mercurial > mplayer.hg
annotate libmpcodecs/ad_faad.c @ 17208:ea284c209b6c
Removing obsolete, and until recently, misdocumented option -verbose .
author | ods15 |
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date | Sat, 17 Dec 2005 18:10:51 +0000 |
parents | 6ff3379a0862 |
children | 934380353fd6 |
rev | line source |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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1 /* ad_faad.c - MPlayer AAC decoder using libfaad2 |
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2 * This file is part of MPlayer, see http://mplayerhq.hu/ for info. |
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3 * (c)2002 by Felix Buenemann <atmosfear at users.sourceforge.net> |
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4 * File licensed under the GPL, see http://www.fsf.org/ for more info. |
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5 */ |
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6 |
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7 #include <stdio.h> |
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8 #include <stdlib.h> |
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9 #include <unistd.h> |
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10 |
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11 #include "config.h" |
739cad21c32c
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12 #include "ad_internal.h" |
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13 |
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14 #ifdef HAVE_FAAD |
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15 |
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16 static ad_info_t info = |
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17 { |
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18 "AAC (MPEG2/4 Advanced Audio Coding)", |
7174 | 19 "faad", |
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20 "Felix Buenemann", |
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21 "faad2", |
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22 "uses libfaad2" |
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23 }; |
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24 |
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25 LIBAD_EXTERN(faad) |
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26 |
10921 | 27 #ifndef USE_INTERNAL_FAAD |
28 #include <faad.h> | |
29 #else | |
17012 | 30 #include "libfaad2/faad.h" |
10921 | 31 #endif |
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32 |
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33 /* configure maximum supported channels, * |
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34 * this is theoretically max. 64 chans */ |
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35 #define FAAD_MAX_CHANNELS 6 |
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36 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) |
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37 |
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38 //#define AAC_DUMP_COMPRESSED |
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39 |
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40 static faacDecHandle faac_hdec; |
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41 static faacDecFrameInfo faac_finfo; |
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42 |
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43 static int preinit(sh_audio_t *sh) |
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44 { |
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Support for HE-AAC. Patch by Loren Merritt <lorenm at u dot washington anotherdot edu>.
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45 sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS; |
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46 sh->audio_in_minsize=FAAD_BUFFLEN; |
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47 return 1; |
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48 } |
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49 |
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50 static int aac_probe(unsigned char *buffer, int len) |
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51 { |
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52 int i = 0, pos = 0; |
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53 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len); |
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54 while(i <= len-4) { |
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55 if( |
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56 ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) || |
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57 (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F') |
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58 ) { |
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59 pos = i; |
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60 break; |
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61 } |
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62 mp_msg(MSGT_DECAUDIO,MSGL_V, "AUDIO PAYLOAD: %x %x %x %x\n", buffer[i], buffer[i+1], buffer[i+2], buffer[i+3]); |
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63 i++; |
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64 } |
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65 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos); |
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66 return pos; |
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67 } |
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68 |
15019 | 69 extern int audio_output_channels; |
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70 static int init(sh_audio_t *sh) |
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71 { |
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72 unsigned long faac_samplerate; |
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73 unsigned char faac_channels; |
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74 int faac_init, pos = 0; |
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75 faac_hdec = faacDecOpen(); |
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76 |
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77 // If we don't get the ES descriptor, try manual config |
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78 if(!sh->codecdata_len && sh->wf) { |
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79 sh->codecdata_len = sh->wf->cbSize; |
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80 sh->codecdata = (char*)(sh->wf+1); |
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81 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n"); |
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82 } |
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83 if(!sh->codecdata_len) { |
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84 #if 1 |
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85 faacDecConfigurationPtr faac_conf; |
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86 /* Set the default object type and samplerate */ |
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87 /* This is useful for RAW AAC files */ |
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88 faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
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89 if(sh->samplerate) |
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90 faac_conf->defSampleRate = sh->samplerate; |
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91 /* XXX: FAAD support FLOAT output, how do we handle |
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92 * that (FAAD_FMT_FLOAT)? ::atmos |
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93 */ |
15019 | 94 if (audio_output_channels <= 2) faac_conf->downMatrix = 1; |
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95 switch(sh->samplesize){ |
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96 case 1: // 8Bit |
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97 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); |
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98 default: |
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99 sh->samplesize=2; |
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100 case 2: // 16Bit |
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101 faac_conf->outputFormat = FAAD_FMT_16BIT; |
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102 break; |
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103 case 3: // 24Bit |
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104 faac_conf->outputFormat = FAAD_FMT_24BIT; |
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105 break; |
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106 case 4: // 32Bit |
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107 faac_conf->outputFormat = FAAD_FMT_32BIT; |
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108 break; |
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109 } |
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110 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. |
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111 |
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112 faacDecSetConfiguration(faac_hdec, faac_conf); |
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113 #endif |
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114 |
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115 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); |
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116 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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117 if(pos) { |
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118 sh->a_in_buffer_len -= pos; |
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119 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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120 sh->a_in_buffer_len += |
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121 demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]), |
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122 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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123 pos = 0; |
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124 } |
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125 |
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126 /* init the codec */ |
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127 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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128 sh->a_in_buffer_len, &faac_samplerate, &faac_channels); |
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129 |
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130 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed |
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131 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi |
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132 |
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133 } else { // We have ES DS in codecdata |
15019 | 134 faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
135 if (audio_output_channels <= 2) { | |
136 faac_conf->downMatrix = 1; | |
137 faacDecSetConfiguration(faac_hdec, faac_conf); | |
138 } | |
139 | |
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140 /*int i; |
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141 for(i = 0; i < sh_audio->codecdata_len; i++) |
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142 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ |
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143 |
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144 faac_init = faacDecInit2(faac_hdec, sh->codecdata, |
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145 sh->codecdata_len, &faac_samplerate, &faac_channels); |
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146 } |
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147 if(faac_init < 0) { |
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148 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! |
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149 faacDecClose(faac_hdec); |
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150 // XXX: free a_in_buffer here or in uninit? |
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151 return 0; |
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152 } else { |
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153 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug! |
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154 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); |
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155 sh->channels = faac_channels; |
15019 | 156 if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1; |
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157 sh->samplerate = faac_samplerate; |
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158 sh->samplesize=2; |
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159 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; |
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160 if(!sh->i_bps) { |
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161 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); |
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162 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos |
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163 } else |
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164 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); |
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165 } |
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166 return 1; |
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167 } |
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168 |
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169 static void uninit(sh_audio_t *sh) |
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170 { |
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171 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); |
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172 faacDecClose(faac_hdec); |
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173 } |
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174 |
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175 static int aac_sync(sh_audio_t *sh) |
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176 { |
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177 int pos = 0; |
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178 if(!sh->codecdata_len) { |
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179 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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180 sh->a_in_buffer_len += |
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181 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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182 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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183 } |
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184 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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185 if(pos) { |
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186 sh->a_in_buffer_len -= pos; |
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187 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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188 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos); |
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189 } |
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190 } |
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191 return pos; |
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192 } |
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193 |
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194 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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195 { |
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196 switch(cmd) |
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197 { |
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198 case ADCTRL_RESYNC_STREAM: |
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199 aac_sync(sh); |
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200 return CONTROL_TRUE; |
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201 #if 0 |
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202 case ADCTRL_SKIP_FRAME: |
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203 return CONTROL_TRUE; |
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204 #endif |
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205 } |
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206 return CONTROL_UNKNOWN; |
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207 } |
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208 |
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209 #define MAX_FAAD_ERRORS 10 |
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210 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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211 { |
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212 int j = 0, len = 0, last_dec_len = 1, errors = 0; |
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213 void *faac_sample_buffer; |
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214 |
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215 while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) { |
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216 |
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217 /* update buffer for raw aac streams: */ |
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218 if(!sh->codecdata_len) |
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219 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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220 sh->a_in_buffer_len += |
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221 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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222 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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223 } |
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224 |
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225 #ifdef DUMP_AAC_COMPRESSED |
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226 {int i; |
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227 for (i = 0; i < 16; i++) |
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228 printf ("%02X ", sh->a_in_buffer[i]); |
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229 printf ("\n");} |
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230 #endif |
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231 |
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232 if(!sh->codecdata_len){ |
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233 // raw aac stream: |
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234 do { |
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235 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j, sh->a_in_buffer_len); |
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236 |
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237 /* update buffer index after faacDecDecode */ |
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238 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { |
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239 sh->a_in_buffer_len=0; |
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240 } else { |
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241 sh->a_in_buffer_len-=faac_finfo.bytesconsumed; |
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memcpy and memmove both copy memory, but when using memcpy the source and destination must not overlap, but here, they did overlap.
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242 memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); |
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243 } |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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244 |
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245 if(faac_finfo.error > 0) { |
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246 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n", |
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247 faacDecGetErrorMessage(faac_finfo.error)); |
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248 j++; |
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249 errors++; |
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250 } else |
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251 break; |
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252 } while(j < FAAD_BUFFLEN && errors < MAX_FAAD_ERRORS); |
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253 } else { |
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254 // packetized (.mp4) aac stream: |
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255 unsigned char* bufptr=NULL; |
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256 int buflen=ds_get_packet(sh->ds, &bufptr); |
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257 if(buflen<=0) break; |
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258 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); |
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259 } |
15019 | 260 //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]); |
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261 |
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262 if(faac_finfo.error > 0) { |
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263 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |
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264 faacDecGetErrorMessage(faac_finfo.error)); |
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265 } else if (faac_finfo.samples == 0) { |
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266 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); |
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267 } else { |
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268 /* XXX: samples already multiplied by channels! */ |
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269 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n", |
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270 sh->samplesize*faac_finfo.samples); |
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271 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); |
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272 last_dec_len = sh->samplesize*faac_finfo.samples; |
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273 len += last_dec_len; |
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274 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); |
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275 } |
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276 } |
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277 return len; |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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278 } |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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279 |
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280 #endif /* !HAVE_FAAD */ |
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281 |