Mercurial > mplayer.hg
annotate libmpcodecs/ad_faad.c @ 11074:85f0917f80cb
ffmpeg moved to mphq, URLs changed
author | gabucino |
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date | Fri, 10 Oct 2003 21:48:13 +0000 |
parents | 3eda9985bc7d |
children | cef97e8c85c0 |
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 |
10921 | 27 #ifndef USE_INTERNAL_FAAD |
28 #include <faad.h> | |
29 #else | |
10726 | 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 { |
5458 | 45 sh->audio_out_minsize=2048*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 init(sh_audio_t *sh) |
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51 { |
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52 unsigned long faac_samplerate; |
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53 unsigned char faac_channels; |
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54 int faac_init; |
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55 faac_hdec = faacDecOpen(); |
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56 |
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57 // If we don't get the ES descriptor, try manual config |
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58 if(!sh->codecdata_len) { |
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59 #if 1 |
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60 faacDecConfigurationPtr faac_conf; |
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61 /* Set the default object type and samplerate */ |
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62 /* This is useful for RAW AAC files */ |
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63 faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
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64 if(sh->samplerate) |
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65 faac_conf->defSampleRate = sh->samplerate; |
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66 /* XXX: FAAD support FLOAT output, how do we handle |
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67 * that (FAAD_FMT_FLOAT)? ::atmos |
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68 */ |
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69 if(sh->samplesize) |
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70 switch(sh->samplesize){ |
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71 case 1: // 8Bit |
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72 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); |
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73 default: |
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74 case 2: // 16Bit |
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75 faac_conf->outputFormat = FAAD_FMT_16BIT; |
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76 break; |
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77 case 3: // 24Bit |
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78 faac_conf->outputFormat = FAAD_FMT_24BIT; |
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79 break; |
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80 case 4: // 32Bit |
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81 faac_conf->outputFormat = FAAD_FMT_32BIT; |
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82 break; |
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83 } |
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84 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. |
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85 |
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86 faacDecSetConfiguration(faac_hdec, faac_conf); |
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87 #endif |
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88 |
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89 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); |
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90 |
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91 /* init the codec */ |
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92 #if (FAADVERSION <= 11) |
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93 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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94 &faac_samplerate, &faac_channels); |
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95 #else |
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96 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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97 sh->a_in_buffer_len, &faac_samplerate, &faac_channels); |
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98 #endif |
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99 |
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100 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed |
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101 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi |
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102 |
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103 } else { // We have ES DS in codecdata |
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104 /*int i; |
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105 for(i = 0; i < sh_audio->codecdata_len; i++) |
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106 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ |
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107 |
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108 faac_init = faacDecInit2(faac_hdec, sh->codecdata, |
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109 sh->codecdata_len, &faac_samplerate, &faac_channels); |
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110 } |
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111 if(faac_init < 0) { |
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112 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! |
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113 faacDecClose(faac_hdec); |
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114 // XXX: free a_in_buffer here or in uninit? |
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115 return 0; |
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116 } else { |
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117 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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118 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %dHz channels: %d\n", faac_samplerate, faac_channels); |
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119 sh->channels = faac_channels; |
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120 sh->samplerate = faac_samplerate; |
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121 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; |
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122 if(!sh->i_bps) { |
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123 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); |
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124 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos |
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125 } else |
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126 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); |
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127 } |
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128 return 1; |
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129 } |
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130 |
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131 static void uninit(sh_audio_t *sh) |
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132 { |
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133 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); |
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134 faacDecClose(faac_hdec); |
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135 } |
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136 |
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137 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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138 { |
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139 switch(cmd) |
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140 { |
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141 #if 0 |
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142 case ADCTRL_RESYNC_STREAM: |
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143 return CONTROL_TRUE; |
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144 case ADCTRL_SKIP_FRAME: |
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145 return CONTROL_TRUE; |
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146 #endif |
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147 } |
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148 return CONTROL_UNKNOWN; |
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149 } |
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150 |
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151 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) |
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152 { |
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153 int j = 0, len = 0; |
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154 void *faac_sample_buffer; |
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155 |
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156 while(len < minlen) { |
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157 |
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158 /* update buffer for raw aac streams: */ |
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159 if(!sh->codecdata_len) |
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160 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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161 sh->a_in_buffer_len += |
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162 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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163 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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164 } |
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165 |
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166 #ifdef DUMP_AAC_COMPRESSED |
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167 {int i; |
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168 for (i = 0; i < 16; i++) |
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169 printf ("%02X ", sh->a_in_buffer[i]); |
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170 printf ("\n");} |
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171 #endif |
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172 |
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173 if(!sh->codecdata_len){ |
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174 // raw aac stream: |
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175 do { |
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176 #if (FAADVERSION <= 11) |
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177 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j); |
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178 #else |
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179 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer+j, sh->a_in_buffer_len); |
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180 #endif |
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181 |
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182 /* update buffer index after faacDecDecode */ |
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183 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { |
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184 sh->a_in_buffer_len=0; |
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185 } else { |
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186 sh->a_in_buffer_len-=faac_finfo.bytesconsumed; |
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187 memcpy(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); |
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188 } |
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189 |
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190 if(faac_finfo.error > 0) { |
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191 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Trying to resync!\n"); |
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192 j++; |
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193 } else |
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194 break; |
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195 } while(j < FAAD_BUFFLEN); |
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196 } else { |
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197 // packetized (.mp4) aac stream: |
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198 unsigned char* bufptr=NULL; |
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199 int buflen=ds_get_packet(sh->ds, &bufptr); |
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200 if(buflen<=0) break; |
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201 #if (FAADVERSION <= 11) |
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202 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr); |
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203 #else |
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204 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); |
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205 #endif |
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206 // printf("FAAC decoded %d of %d (err: %d) \n",faac_finfo.bytesconsumed,buflen,faac_finfo.error); |
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207 } |
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208 |
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209 if(faac_finfo.error > 0) { |
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210 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |
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211 faacDecGetErrorMessage(faac_finfo.error)); |
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212 } else if (faac_finfo.samples == 0) { |
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213 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); |
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214 } else { |
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215 /* XXX: samples already multiplied by channels! */ |
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216 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%d Bytes)!\n", |
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217 sh->samplesize*faac_finfo.samples); |
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218 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); |
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219 len += sh->samplesize*faac_finfo.samples; |
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220 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); |
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221 } |
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222 } |
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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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223 return len; |
5356
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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224 } |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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225 |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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226 #endif /* !HAVE_FAAD */ |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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227 |