Mercurial > mplayer.hg
annotate libmpcodecs/ad_faad.c @ 29491:99eda963d27a
Fix incorrect channel ordering for lavc audio codecs (specifically ffac3,
ffdca, ffflac, ffaac, fftruehd). In the process, adds support for 32-bit
samples.
author | tack |
---|---|
date | Tue, 18 Aug 2009 22:24:36 +0000 |
parents | 0f1b5b68af32 |
children | 4eae69f3f4f4 |
rev | line source |
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28106 | 1 /* |
2 * MPlayer AAC decoder using libfaad2 | |
3 * | |
28110 | 4 * Copyright (C) 2002 Felix Buenemann <atmosfear at users.sourceforge.net> |
28106 | 5 * |
6 * This file is part of MPlayer. | |
7 * | |
8 * MPlayer is free software; you can redistribute it and/or modify | |
9 * it under the terms of the GNU General Public License as published by | |
10 * the Free Software Foundation; either version 2 of the License, or | |
11 * (at your option) any later version. | |
12 * | |
13 * MPlayer is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
16 * GNU General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU General Public License along | |
19 * with MPlayer; if not, write to the Free Software Foundation, Inc., | |
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. | |
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21 */ |
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22 |
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23 #include <stdio.h> |
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24 #include <stdlib.h> |
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25 #include <unistd.h> |
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26 |
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27 #include "config.h" |
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28 #include "ad_internal.h" |
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Fix all current known multi-channel wrong order problems by adding
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29 #include "libaf/reorder_ch.h" |
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30 |
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31 static ad_info_t info = |
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32 { |
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33 "AAC (MPEG2/4 Advanced Audio Coding)", |
7174 | 34 "faad", |
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35 "Felix Buenemann", |
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36 "faad2", |
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37 "uses libfaad2" |
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38 }; |
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39 |
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40 LIBAD_EXTERN(faad) |
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41 |
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Start unifying names of internal preprocessor directives.
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42 #ifndef CONFIG_FAAD_INTERNAL |
10921 | 43 #include <faad.h> |
44 #else | |
17012 | 45 #include "libfaad2/faad.h" |
10921 | 46 #endif |
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47 |
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48 /* configure maximum supported channels, * |
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49 * this is theoretically max. 64 chans */ |
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50 #define FAAD_MAX_CHANNELS 6 |
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51 #define FAAD_BUFFLEN (FAAD_MIN_STREAMSIZE*FAAD_MAX_CHANNELS) |
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52 |
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53 //#define AAC_DUMP_COMPRESSED |
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54 |
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55 static faacDecHandle faac_hdec; |
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56 static faacDecFrameInfo faac_finfo; |
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57 |
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58 static int preinit(sh_audio_t *sh) |
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59 { |
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Support for HE-AAC. Patch by Loren Merritt <lorenm at u dot washington anotherdot edu>.
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60 sh->audio_out_minsize=8192*FAAD_MAX_CHANNELS; |
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61 sh->audio_in_minsize=FAAD_BUFFLEN; |
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62 return 1; |
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63 } |
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64 |
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65 static int aac_probe(unsigned char *buffer, int len) |
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66 { |
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67 int i = 0, pos = 0; |
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68 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: %d bytes\n", len); |
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69 while(i <= len-4) { |
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70 if( |
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71 ((buffer[i] == 0xff) && ((buffer[i+1] & 0xf6) == 0xf0)) || |
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72 (buffer[i] == 'A' && buffer[i+1] == 'D' && buffer[i+2] == 'I' && buffer[i+3] == 'F') |
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73 ) { |
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74 pos = i; |
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75 break; |
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76 } |
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77 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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78 i++; |
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79 } |
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80 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC_PROBE: ret %d\n", pos); |
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81 return pos; |
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82 } |
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83 |
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84 static int init(sh_audio_t *sh) |
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85 { |
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I think libfaad2 changed recently. The functions faacDecInit() and
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86 unsigned long faac_samplerate; |
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87 unsigned char faac_channels; |
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88 int faac_init, pos = 0; |
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89 faac_hdec = faacDecOpen(); |
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90 |
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91 // If we don't get the ES descriptor, try manual config |
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92 if(!sh->codecdata_len && sh->wf) { |
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93 sh->codecdata_len = sh->wf->cbSize; |
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94 sh->codecdata = malloc(sh->codecdata_len); |
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95 memcpy(sh->codecdata, sh->wf+1, sh->codecdata_len); |
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96 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: codecdata extracted from WAVEFORMATEX\n"); |
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97 } |
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98 if(!sh->codecdata_len) { |
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99 #if 1 |
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100 faacDecConfigurationPtr faac_conf; |
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101 /* Set the default object type and samplerate */ |
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102 /* This is useful for RAW AAC files */ |
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103 faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
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104 if(sh->samplerate) |
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105 faac_conf->defSampleRate = sh->samplerate; |
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106 /* XXX: FAAD support FLOAT output, how do we handle |
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107 * that (FAAD_FMT_FLOAT)? ::atmos |
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108 */ |
15019 | 109 if (audio_output_channels <= 2) faac_conf->downMatrix = 1; |
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110 switch(sh->samplesize){ |
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111 case 1: // 8Bit |
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112 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: 8Bit samplesize not supported by FAAD, assuming 16Bit!\n"); |
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113 default: |
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114 sh->samplesize=2; |
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115 case 2: // 16Bit |
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116 faac_conf->outputFormat = FAAD_FMT_16BIT; |
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117 break; |
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118 case 3: // 24Bit |
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119 faac_conf->outputFormat = FAAD_FMT_24BIT; |
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120 break; |
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121 case 4: // 32Bit |
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122 faac_conf->outputFormat = FAAD_FMT_32BIT; |
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123 break; |
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124 } |
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125 //faac_conf->defObjectType = LTP; // => MAIN, LC, SSR, LTP available. |
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126 |
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127 faacDecSetConfiguration(faac_hdec, faac_conf); |
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128 #endif |
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129 |
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130 sh->a_in_buffer_len = demux_read_data(sh->ds, sh->a_in_buffer, sh->a_in_buffer_size); |
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131 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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132 if(pos) { |
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133 sh->a_in_buffer_len -= pos; |
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134 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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135 sh->a_in_buffer_len += |
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136 demux_read_data(sh->ds,&(sh->a_in_buffer[sh->a_in_buffer_len]), |
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137 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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138 pos = 0; |
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139 } |
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140 |
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141 /* init the codec */ |
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142 faac_init = faacDecInit(faac_hdec, sh->a_in_buffer, |
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143 sh->a_in_buffer_len, &faac_samplerate, &faac_channels); |
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144 |
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145 sh->a_in_buffer_len -= (faac_init > 0)?faac_init:0; // how many bytes init consumed |
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146 // XXX FIXME: shouldn't we memcpy() here in a_in_buffer ?? --A'rpi |
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147 |
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148 } else { // We have ES DS in codecdata |
15019 | 149 faacDecConfigurationPtr faac_conf = faacDecGetCurrentConfiguration(faac_hdec); |
150 if (audio_output_channels <= 2) { | |
151 faac_conf->downMatrix = 1; | |
152 faacDecSetConfiguration(faac_hdec, faac_conf); | |
153 } | |
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154 |
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155 /*int i; |
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156 for(i = 0; i < sh_audio->codecdata_len; i++) |
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157 printf("codecdata_dump %d: 0x%02X\n", i, sh_audio->codecdata[i]);*/ |
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158 |
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159 faac_init = faacDecInit2(faac_hdec, sh->codecdata, |
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160 sh->codecdata_len, &faac_samplerate, &faac_channels); |
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161 } |
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162 if(faac_init < 0) { |
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163 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to initialize the decoder!\n"); // XXX: deal with cleanup! |
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164 faacDecClose(faac_hdec); |
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165 // XXX: free a_in_buffer here or in uninit? |
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166 return 0; |
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167 } else { |
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168 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Decoder init done (%dBytes)!\n", sh->a_in_buffer_len); // XXX: remove or move to debug! |
17366 | 169 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Negotiated samplerate: %ldHz channels: %d\n", faac_samplerate, faac_channels); |
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170 sh->channels = faac_channels; |
15019 | 171 if (audio_output_channels <= 2) sh->channels = faac_channels > 1 ? 2 : 1; |
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172 sh->samplerate = faac_samplerate; |
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173 sh->samplesize=2; |
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174 //sh->o_bps = sh->samplesize*faac_channels*faac_samplerate; |
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175 if(!sh->i_bps) { |
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176 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: compressed input bitrate missing, assuming 128kbit/s!\n"); |
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177 sh->i_bps = 128*1000/8; // XXX: HACK!!! ::atmos |
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178 } else |
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179 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: got %dkbit/s bitrate from MP4 header!\n",sh->i_bps*8/1000); |
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180 } |
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181 return 1; |
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182 } |
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183 |
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184 static void uninit(sh_audio_t *sh) |
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185 { |
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186 mp_msg(MSGT_DECAUDIO,MSGL_V,"FAAD: Closing decoder!\n"); |
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187 faacDecClose(faac_hdec); |
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188 } |
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189 |
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190 static int aac_sync(sh_audio_t *sh) |
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191 { |
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192 int pos = 0; |
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193 if(!sh->codecdata_len) { |
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194 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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195 sh->a_in_buffer_len += |
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196 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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197 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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198 } |
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199 pos = aac_probe(sh->a_in_buffer, sh->a_in_buffer_len); |
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200 if(pos) { |
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201 sh->a_in_buffer_len -= pos; |
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202 memmove(sh->a_in_buffer, &(sh->a_in_buffer[pos]), sh->a_in_buffer_len); |
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203 mp_msg(MSGT_DECAUDIO,MSGL_V, "\nAAC SYNC AFTER %d bytes\n", pos); |
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204 } |
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205 } |
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206 return pos; |
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207 } |
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208 |
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209 static int control(sh_audio_t *sh,int cmd,void* arg, ...) |
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210 { |
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211 switch(cmd) |
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212 { |
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213 case ADCTRL_RESYNC_STREAM: |
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214 aac_sync(sh); |
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215 return CONTROL_TRUE; |
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216 #if 0 |
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217 case ADCTRL_SKIP_FRAME: |
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218 return CONTROL_TRUE; |
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219 #endif |
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220 } |
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221 return CONTROL_UNKNOWN; |
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222 } |
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223 |
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224 #define MAX_FAAD_ERRORS 10 |
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225 static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) |
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226 { |
24120 | 227 int len = 0, last_dec_len = 1, errors = 0; |
228 // int j = 0; | |
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229 void *faac_sample_buffer; |
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230 |
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231 while(len < minlen && last_dec_len > 0 && errors < MAX_FAAD_ERRORS) { |
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232 |
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233 /* update buffer for raw aac streams: */ |
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234 if(!sh->codecdata_len) |
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235 if(sh->a_in_buffer_len < sh->a_in_buffer_size){ |
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236 sh->a_in_buffer_len += |
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237 demux_read_data(sh->ds,&sh->a_in_buffer[sh->a_in_buffer_len], |
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238 sh->a_in_buffer_size - sh->a_in_buffer_len); |
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239 } |
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240 |
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241 #ifdef DUMP_AAC_COMPRESSED |
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242 {int i; |
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243 for (i = 0; i < 16; i++) |
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244 printf ("%02X ", sh->a_in_buffer[i]); |
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245 printf ("\n");} |
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246 #endif |
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247 |
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248 if(!sh->codecdata_len){ |
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249 // raw aac stream: |
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250 do { |
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251 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, sh->a_in_buffer, sh->a_in_buffer_len); |
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252 |
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253 /* update buffer index after faacDecDecode */ |
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254 if(faac_finfo.bytesconsumed >= sh->a_in_buffer_len) { |
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255 sh->a_in_buffer_len=0; |
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256 } else { |
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257 sh->a_in_buffer_len-=faac_finfo.bytesconsumed; |
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258 memmove(sh->a_in_buffer,&sh->a_in_buffer[faac_finfo.bytesconsumed],sh->a_in_buffer_len); |
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259 } |
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260 |
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261 if(faac_finfo.error > 0) { |
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262 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: error: %s, trying to resync!\n", |
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263 faacDecGetErrorMessage(faac_finfo.error)); |
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264 if (sh->a_in_buffer_len <= 0) { |
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265 errors = MAX_FAAD_ERRORS; |
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266 break; |
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267 } |
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268 sh->a_in_buffer_len--; |
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269 memmove(sh->a_in_buffer,&sh->a_in_buffer[1],sh->a_in_buffer_len); |
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270 aac_sync(sh); |
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271 errors++; |
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272 } else |
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273 break; |
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274 } while(errors < MAX_FAAD_ERRORS); |
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275 } else { |
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276 // packetized (.mp4) aac stream: |
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277 unsigned char* bufptr=NULL; |
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278 double pts; |
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279 int buflen=ds_get_packet_pts(sh->ds, &bufptr, &pts); |
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280 if(buflen<=0) break; |
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281 if (pts != MP_NOPTS_VALUE) { |
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282 sh->pts = pts; |
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283 sh->pts_bytes = 0; |
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284 } |
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285 faac_sample_buffer = faacDecDecode(faac_hdec, &faac_finfo, bufptr, buflen); |
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286 } |
15019 | 287 //for (j=0;j<faac_finfo.channels;j++) printf("%d:%d\n", j, faac_finfo.channel_position[j]); |
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288 |
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289 if(faac_finfo.error > 0) { |
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290 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"FAAD: Failed to decode frame: %s \n", |
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291 faacDecGetErrorMessage(faac_finfo.error)); |
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292 } else if (faac_finfo.samples == 0) { |
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293 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Decoded zero samples!\n"); |
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294 } else { |
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295 /* XXX: samples already multiplied by channels! */ |
17366 | 296 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"FAAD: Successfully decoded frame (%ld Bytes)!\n", |
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297 sh->samplesize*faac_finfo.samples); |
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298 |
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299 if (sh->channels >= 5) |
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300 reorder_channel_copy_nch(faac_sample_buffer, |
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301 AF_CHANNEL_LAYOUT_AAC_DEFAULT, |
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302 buf+len, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, |
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303 sh->channels, |
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304 faac_finfo.samples, sh->samplesize); |
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305 else |
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306 memcpy(buf+len,faac_sample_buffer, sh->samplesize*faac_finfo.samples); |
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307 last_dec_len = sh->samplesize*faac_finfo.samples; |
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308 len += last_dec_len; |
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309 sh->pts_bytes += last_dec_len; |
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310 //printf("FAAD: buffer: %d bytes consumed: %d \n", k, faac_finfo.bytesconsumed); |
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311 } |
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312 } |
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313 return len; |
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314 } |