Mercurial > mplayer.hg
annotate libmpcodecs/ad_faad.c @ 20747:d639a26d9141
added definition of DEMUXER_CTRL_RESYNC
author | nicodvb |
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date | Tue, 07 Nov 2006 22:51:18 +0000 |
parents | 9cd5e242121e |
children | f29d31547c31 |
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. |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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5 */ |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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6 |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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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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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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11 #include "config.h" |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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12 #include "ad_internal.h" |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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13 |
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14 static ad_info_t info = |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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15 { |
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16 "AAC (MPEG2/4 Advanced Audio Coding)", |
7174 | 17 "faad", |
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18 "Felix Buenemann", |
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19 "faad2", |
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20 "uses libfaad2" |
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21 }; |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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22 |
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23 LIBAD_EXTERN(faad) |
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24 |
19348 | 25 #ifndef USE_FAAD_INTERNAL |
10921 | 26 #include <faad.h> |
27 #else | |
17012 | 28 #include "libfaad2/faad.h" |
10921 | 29 #endif |
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30 |
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31 /* configure maximum supported channels, * |
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32 * this is theoretically max. 64 chans */ |
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33 #define FAAD_MAX_CHANNELS 6 |
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34 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) |
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35 |
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36 //#define AAC_DUMP_COMPRESSED |
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37 |
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38 static faacDecHandle faac_hdec; |
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39 static faacDecFrameInfo faac_finfo; |
739cad21c32c
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40 |
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41 static int preinit(sh_audio_t *sh) |
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42 { |
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Support for HE-AAC. Patch by Loren Merritt <lorenm at u dot washington anotherdot edu>.
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43 sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS; |
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44 sh->audio_in_minsize=FAAD_BUFFLEN; |
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45 return 1; |
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46 } |
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47 |
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48 static int aac_probe(unsigned char *buffer, int len) |
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49 { |
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50 int i = 0, pos = 0; |
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51 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len); |
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52 while(i <= len-4) { |
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53 if( |
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54 ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) || |
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55 (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F') |
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56 ) { |
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57 pos = i; |
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58 break; |
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59 } |
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60 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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61 i++; |
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62 } |
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63 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos); |
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64 return pos; |
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65 } |
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66 |
15019 | 67 extern int audio_output_channels; |
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68 static int init(sh_audio_t *sh) |
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69 { |
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70 unsigned long faac_samplerate; |
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71 unsigned char faac_channels; |
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72 int faac_init, pos = 0; |
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73 faac_hdec = faacDecOpen(); |
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74 |
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75 // If we don't get the ES descriptor, try manual config |
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76 if(!sh->codecdata_len && sh->wf) { |
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77 sh->codecdata_len = sh->wf->cbSize; |
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78 sh->codecdata = (char*)(sh->wf+1); |
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79 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n"); |
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80 } |
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81 if(!sh->codecdata_len) { |
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82 #if 1 |
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83 faacDecConfigurationPtr faac_conf; |
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84 /* Set the default object type and samplerate */ |
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85 /* This is useful for RAW AAC files */ |
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86 faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
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87 if(sh->samplerate) |
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88 faac_conf->defSampleRate = sh->samplerate; |
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89 /* XXX: FAAD support FLOAT output, how do we handle |
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90 * that (FAAD_FMT_FLOAT)? ::atmos |
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91 */ |
15019 | 92 if (audio_output_channels <= 2) faac_conf->downMatrix = 1; |
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93 switch(sh->samplesize){ |
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94 case 1: // 8Bit |
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95 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); |
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96 default: |
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97 sh->samplesize=2; |
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98 case 2: // 16Bit |
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99 faac_conf->outputFormat = FAAD_FMT_16BIT; |
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100 break; |
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101 case 3: // 24Bit |
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102 faac_conf->outputFormat = FAAD_FMT_24BIT; |
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103 break; |
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104 case 4: // 32Bit |
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105 faac_conf->outputFormat = FAAD_FMT_32BIT; |
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106 break; |
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107 } |
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108 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. |
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109 |
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110 faacDecSetConfiguration(faac_hdec, faac_conf); |
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111 #endif |
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112 |
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113 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); |
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114 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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115 if(pos) { |
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116 sh->a_in_buffer_len -= pos; |
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117 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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118 sh->a_in_buffer_len += |
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119 demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]), |
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120 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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121 pos = 0; |
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122 } |
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123 |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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124 /* init the codec */ |
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125 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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126 sh->a_in_buffer_len, &faac_samplerate, &faac_channels); |
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127 |
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128 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed |
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129 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi |
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130 |
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131 } else { // We have ES DS in codecdata |
15019 | 132 faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
133 if (audio_output_channels <= 2) { | |
134 faac_conf->downMatrix = 1; | |
135 faacDecSetConfiguration(faac_hdec, faac_conf); | |
136 } | |
137 | |
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138 /*int i; |
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139 for(i = 0; i < sh_audio->codecdata_len; i++) |
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140 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ |
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141 |
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142 faac_init = faacDecInit2(faac_hdec, sh->codecdata, |
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143 sh->codecdata_len, &faac_samplerate, &faac_channels); |
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144 } |
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145 if(faac_init < 0) { |
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146 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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147 faacDecClose(faac_hdec); |
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148 // XXX: free a_in_buffer here or in uninit? |
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149 return 0; |
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150 } else { |
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151 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug! |
17366 | 152 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels); |
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153 sh->channels = faac_channels; |
15019 | 154 if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1; |
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155 sh->samplerate = faac_samplerate; |
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156 sh->samplesize=2; |
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157 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; |
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158 if(!sh->i_bps) { |
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159 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); |
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160 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos |
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161 } else |
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162 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); |
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163 } |
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164 return 1; |
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165 } |
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166 |
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167 static void uninit(sh_audio_t *sh) |
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168 { |
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169 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); |
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170 faacDecClose(faac_hdec); |
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171 } |
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172 |
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173 static int aac_sync(sh_audio_t *sh) |
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174 { |
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175 int pos = 0; |
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176 if(!sh->codecdata_len) { |
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177 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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178 sh->a_in_buffer_len += |
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179 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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180 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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181 } |
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182 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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183 if(pos) { |
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184 sh->a_in_buffer_len -= pos; |
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185 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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186 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos); |
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187 } |
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188 } |
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189 return pos; |
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190 } |
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191 |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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192 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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193 { |
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194 switch(cmd) |
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195 { |
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196 case ADCTRL_RESYNC_STREAM: |
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197 aac_sync(sh); |
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198 return CONTROL_TRUE; |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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199 #if 0 |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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200 case ADCTRL_SKIP_FRAME: |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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201 return CONTROL_TRUE; |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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202 #endif |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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203 } |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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204 return CONTROL_UNKNOWN; |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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205 } |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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206 |
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attila
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207 #define MAX_FAAD_ERRORS 10 |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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208 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) |
739cad21c32c
Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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209 { |
16337
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210 int j = 0, len = 0, last_dec_len = 1, errors = 0; |
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211 void *faac_sample_buffer; |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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212 |
16337
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attila
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213 while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) { |
6894
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214 |
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215 /* update buffer for raw aac streams: */ |
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216 if(!sh->codecdata_len) |
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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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217 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
3c5d5626d3a1
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218 sh->a_in_buffer_len += |
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atmos4
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219 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
3c5d5626d3a1
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220 sh->a_in_buffer_size - sh->a_in_buffer_len); |
3c5d5626d3a1
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221 } |
5356
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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222 |
739cad21c32c
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223 #ifdef DUMP_AAC_COMPRESSED |
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224 {int i; |
3c5d5626d3a1
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225 for (i = 0; i < 16; i++) |
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226 printf ("%02X ", sh->a_in_buffer[i]); |
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227 printf ("\n");} |
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Port dec_audio.c faad aac decoder to libmpcodecs, cleaned up code, improved buffering scheme.
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228 #endif |
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229 |
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230 if(!sh->codecdata_len){ |
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231 // raw aac stream: |
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232 do { |
18644
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gpoirier
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233 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len); |
9321
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234 |
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235 /* update buffer index after faacDecDecode */ |
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236 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { |
3c5d5626d3a1
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237 sh->a_in_buffer_len=0; |
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!
atmos4
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238 } else { |
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239 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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240 memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); |
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241 } |
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242 |
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243 if(faac_finfo.error > 0) { |
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244 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n", |
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245 faacDecGetErrorMessage(faac_finfo.error)); |
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in case of errors after decoding quit the main decoding
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246 if (sh->a_in_buffer_len <= 0) { |
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247 errors = MAX_FAAD_ERRORS; |
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248 break; |
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249 } |
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250 sh->a_in_buffer_len--; |
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251 memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len); |
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252 aac_sync(sh); |
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253 errors++; |
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254 } else |
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255 break; |
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256 } while(errors < MAX_FAAD_ERRORS); |
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257 } else { |
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258 // packetized (.mp4) aac stream: |
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259 unsigned char* bufptr=NULL; |
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260 double pts; |
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261 int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts); |
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262 if(buflen<=0) break; |
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263 if (pts != MP_NOPTS_VALUE) { |
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264 sh->pts = pts; |
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265 sh->pts_bytes = 0; |
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266 } |
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libfaad2 v1.0, v1.1, v1.2 detection, and API change workaround in ad_faad.c
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267 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); |
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268 } |
15019 | 269 //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]); |
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270 |
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271 if(faac_finfo.error > 0) { |
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272 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |
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273 faacDecGetErrorMessage(faac_finfo.error)); |
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274 } else if (faac_finfo.samples == 0) { |
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275 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); |
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276 } else { |
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277 /* XXX: samples already multiplied by channels! */ |
17366 | 278 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n", |
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279 sh->samplesize*faac_finfo.samples); |
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280 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); |
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281 last_dec_len = sh->samplesize*faac_finfo.samples; |
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282 len += last_dec_len; |
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283 sh->pts_bytes += last_dec_len; |
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284 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); |
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285 } |
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286 } |
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287 return len; |
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288 } |