Mercurial > mplayer.hg
annotate libmpcodecs/ad_faad.c @ 10065:422876da049e
Error in the aspect ratio calculation, plus typos and rewordings.
Noticed by Jason Tackaberry <tack@auc.ca>.
author | diego |
---|---|
date | Sun, 04 May 2003 23:27:47 +0000 |
parents | 6fa743f3094b |
children | bae3e8c81ada |
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" |
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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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-afm/-vfm help implemenetd, some cosmetics of ad/vd codec names/comments
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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 |
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27 #include <faad.h> |
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28 |
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29 /* configure maximum supported channels, * |
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30 * this is theoretically max. 64 chans */ |
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31 #define FAAD_MAX_CHANNELS 6 |
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32 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) |
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33 |
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34 //#define AAC_DUMP_COMPRESSED |
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35 |
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36 static faacDecHandle faac_hdec; |
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37 static faacDecFrameInfo faac_finfo; |
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38 |
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39 static int preinit(sh_audio_t *sh) |
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40 { |
5458 | 41 sh->audio_out_minsize=2048*FAAD_MAX_CHANNELS; |
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42 sh->audio_in_minsize=FAAD_BUFFLEN; |
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43 return 1; |
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44 } |
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45 |
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46 static int init(sh_audio_t *sh) |
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47 { |
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48 unsigned long faac_samplerate; |
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49 unsigned char faac_channels; |
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50 int faac_init; |
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51 faac_hdec = faacDecOpen(); |
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52 |
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53 // If we don't get the ES descriptor, try manual config |
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54 if(!sh->codecdata_len) { |
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55 #if 1 |
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56 faacDecConfigurationPtr faac_conf; |
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57 /* Set the default object type and samplerate */ |
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58 /* This is useful for RAW AAC files */ |
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59 faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
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60 if(sh->samplerate) |
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61 faac_conf->defSampleRate = sh->samplerate; |
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62 /* XXX: FAAD support FLOAT output, how do we handle |
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63 * that (FAAD_FMT_FLOAT)? ::atmos |
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64 */ |
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65 if(sh->samplesize) |
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66 switch(sh->samplesize){ |
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67 case 1: // 8Bit |
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68 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); |
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69 default: |
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70 case 2: // 16Bit |
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71 faac_conf->outputFormat = FAAD_FMT_16BIT; |
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72 break; |
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73 case 3: // 24Bit |
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74 faac_conf->outputFormat = FAAD_FMT_24BIT; |
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75 break; |
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76 case 4: // 32Bit |
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77 faac_conf->outputFormat = FAAD_FMT_32BIT; |
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78 break; |
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79 } |
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80 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. |
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81 |
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82 faacDecSetConfiguration(faac_hdec, faac_conf); |
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83 #endif |
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84 |
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85 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); |
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86 |
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87 /* init the codec */ |
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libfaad2 v1.0, v1.1, v1.2 detection, and API change workaround in ad_faad.c
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88 #if (FAADVERSION <= 11) |
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89 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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90 &faac_samplerate, &faac_channels); |
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91 #else |
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92 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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93 sh->a_in_buffer_len, &faac_samplerate, &faac_channels); |
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94 #endif |
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95 |
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96 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed |
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97 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi |
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98 |
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99 } else { // We have ES DS in codecdata |
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100 /*int i; |
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101 for(i = 0; i < sh_audio->codecdata_len; i++) |
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102 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ |
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103 |
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104 faac_init = faacDecInit2(faac_hdec, sh->codecdata, |
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105 sh->codecdata_len, &faac_samplerate, &faac_channels); |
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106 } |
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107 if(faac_init < 0) { |
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108 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! |
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109 faacDecClose(faac_hdec); |
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110 // XXX: free a_in_buffer here or in uninit? |
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111 return 0; |
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112 } else { |
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113 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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114 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); |
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115 sh->channels = faac_channels; |
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116 sh->samplerate = faac_samplerate; |
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117 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; |
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118 if(!sh->i_bps) { |
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119 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); |
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120 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos |
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121 } else |
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122 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); |
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123 } |
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124 return 1; |
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125 } |
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126 |
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127 static void uninit(sh_audio_t *sh) |
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128 { |
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129 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); |
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130 faacDecClose(faac_hdec); |
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131 } |
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132 |
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133 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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134 { |
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135 switch(cmd) |
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136 { |
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137 #if 0 |
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138 case ADCTRL_RESYNC_STREAM: |
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139 return CONTROL_TRUE; |
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140 case ADCTRL_SKIP_FRAME: |
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141 return CONTROL_TRUE; |
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142 #endif |
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143 } |
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144 return CONTROL_UNKNOWN; |
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145 } |
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146 |
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147 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) |
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148 { |
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149 int j = 0, len = 0; |
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150 void *faac_sample_buffer; |
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151 |
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152 while(len < minlen) { |
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153 |
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154 /* update buffer for raw aac streams: */ |
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155 if(!sh->codecdata_len) |
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156 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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157 sh->a_in_buffer_len += |
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158 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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159 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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160 } |
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161 |
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162 #ifdef DUMP_AAC_COMPRESSED |
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163 {int i; |
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164 for (i = 0; i < 16; i++) |
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165 printf ("%02X ", sh->a_in_buffer[i]); |
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166 printf ("\n");} |
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167 #endif |
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168 |
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169 if(!sh->codecdata_len){ |
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170 // raw aac stream: |
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171 do { |
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172 #if (FAADVERSION <= 11) |
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173 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j); |
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174 #else |
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175 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j, sh->a_in_buffer_len); |
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176 #endif |
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177 |
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178 /* update buffer index after faacDecDecode */ |
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179 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { |
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180 sh->a_in_buffer_len=0; |
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181 } else { |
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182 sh->a_in_buffer_len-=faac_finfo.bytesconsumed; |
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183 memcpy(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); |
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184 } |
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185 |
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186 if(faac_finfo.error > 0) { |
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187 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n"); |
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188 j++; |
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189 } else |
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190 break; |
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191 } while(j < FAAD_BUFFLEN); |
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192 } else { |
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193 // packetized (.mp4) aac stream: |
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194 unsigned char* bufptr=NULL; |
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195 int buflen=ds_get_packet(sh->ds, &bufptr); |
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196 if(buflen<=0) break; |
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197 #if (FAADVERSION <= 11) |
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198 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr); |
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199 #else |
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200 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); |
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201 #endif |
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202 // printf("FAAC decoded %d of %d (err: %d) \n",faac_finfo.bytesconsumed,buflen,faac_finfo.error); |
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203 } |
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204 |
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205 if(faac_finfo.error > 0) { |
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206 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |
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207 faacDecGetErrorMessage(faac_finfo.error)); |
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208 } else if (faac_finfo.samples == 0) { |
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209 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); |
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210 } else { |
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211 /* XXX: samples already multiplied by channels! */ |
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212 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n", |
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213 sh->samplesize*faac_finfo.samples); |
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214 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); |
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215 len += sh->samplesize*faac_finfo.samples; |
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216 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); |
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217 } |
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218 } |
3c5d5626d3a1
Don't kill me, pure indent change as I forgot to reindent when copy and pasting from dec_audio.c, as this is the first revision it doesn't break changelog!
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219 return len; |
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739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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220 } |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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221 |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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222 #endif /* !HAVE_FAAD */ |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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223 |