annotate libmpcodecs/ad_ffmpeg.c @ 24576:6704a924d4aa

According to MSDN a thread must call CoUninitialize once for each successful call it has made to CoInitialize or CoInitializeEx, including any call that returns S_FALSE. Only the CoUninitialize call corresponding to the CoInitialize or CoInitializeEx call that initialized the library can close it. patch by Gianluigi Tiesi, mplayer netfarm it
author diego
date Sun, 23 Sep 2007 20:37:33 +0000
parents 352d7d9422b5
children dfa8a510c81c
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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1 #include <stdio.h>
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2 #include <stdlib.h>
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3 #include <unistd.h>
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4
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5 #include "config.h"
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6 #include "mp_msg.h"
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7 #include "help_mp.h"
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8
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9 #include "ad_internal.h"
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10
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11 #include "mpbswap.h"
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12
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13 static ad_info_t info =
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14 {
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15 "FFmpeg/libavcodec audio decoders",
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16 "ffmpeg",
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17 "Nick Kurshev",
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18 "ffmpeg.sf.net",
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19 ""
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20 };
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21
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22 LIBAD_EXTERN(ffmpeg)
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23
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24 #define assert(x)
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25
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26 #ifdef USE_LIBAVCODEC_SO
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27 #include <ffmpeg/avcodec.h>
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28 #else
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29 #include "avcodec.h"
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30 #endif
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31
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32 extern int avcodec_inited;
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33
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34 static int preinit(sh_audio_t *sh)
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35 {
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36 sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
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37 return 1;
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38 }
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39
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40 static int init(sh_audio_t *sh_audio)
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41 {
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42 int x;
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43 AVCodecContext *lavc_context;
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44 AVCodec *lavc_codec;
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45
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46 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
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47 if(!avcodec_inited){
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48 avcodec_init();
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49 avcodec_register_all();
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50 avcodec_inited=1;
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51 }
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52
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53 lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
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54 if(!lavc_codec){
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55 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
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56 return 0;
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57 }
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58
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59 lavc_context = avcodec_alloc_context();
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60 sh_audio->context=lavc_context;
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61
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62 lavc_context->sample_rate = sh_audio->samplerate;
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63 lavc_context->bit_rate = sh_audio->i_bps * 8;
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64 if(sh_audio->wf){
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65 lavc_context->channels = sh_audio->wf->nChannels;
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66 lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
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67 lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
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68 lavc_context->block_align = sh_audio->wf->nBlockAlign;
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69 lavc_context->bits_per_sample = sh_audio->wf->wBitsPerSample;
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70 }
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71 lavc_context->request_channels = audio_output_channels;
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72 lavc_context->codec_tag = sh_audio->format; //FOURCC
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73 lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
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74
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75 /* alloc extra data */
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76 if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
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77 lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
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78 lavc_context->extradata_size = sh_audio->wf->cbSize;
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79 memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
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80 lavc_context->extradata_size);
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81 }
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82
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83 // for QDM2
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84 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
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85 {
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86 lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
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87 lavc_context->extradata_size = sh_audio->codecdata_len;
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88 memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
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89 lavc_context->extradata_size);
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90 }
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91
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92 /* open it */
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93 if (avcodec_open(lavc_context, lavc_codec) < 0) {
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94 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
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95 return 0;
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96 }
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97 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
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98
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99 // printf("\nFOURCC: 0x%X\n",sh_audio->format);
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100 if(sh_audio->format==0x3343414D){
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101 // MACE 3:1
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102 sh_audio->ds->ss_div = 2*3; // 1 samples/packet
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103 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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104 } else
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105 if(sh_audio->format==0x3643414D){
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106 // MACE 6:1
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107 sh_audio->ds->ss_div = 2*6; // 1 samples/packet
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108 sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
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109 }
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110
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111 // Decode at least 1 byte: (to get header filled)
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112 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
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113 if(x>0) sh_audio->a_buffer_len=x;
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114
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115 sh_audio->channels=lavc_context->channels;
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116 sh_audio->samplerate=lavc_context->sample_rate;
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117 sh_audio->i_bps=lavc_context->bit_rate/8;
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118 if(sh_audio->wf){
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119 // If the decoder uses the wrong number of channels all is lost anyway.
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120 // sh_audio->channels=sh_audio->wf->nChannels;
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121 if (sh_audio->wf->nSamplesPerSec)
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122 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
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123 if (sh_audio->wf->nAvgBytesPerSec)
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124 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
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125 }
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126 sh_audio->samplesize=2;
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127 return 1;
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128 }
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129
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130 static void uninit(sh_audio_t *sh)
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131 {
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132 AVCodecContext *lavc_context = sh->context;
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133
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134 if (avcodec_close(lavc_context) < 0)
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135 mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec);
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136 av_freep(&lavc_context->extradata);
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137 av_freep(&lavc_context);
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138 }
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139
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140 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
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141 {
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142 AVCodecContext *lavc_context = sh->context;
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143 switch(cmd){
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144 case ADCTRL_RESYNC_STREAM:
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145 avcodec_flush_buffers(lavc_context);
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146 return CONTROL_TRUE;
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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_audio,unsigned char *buf,int minlen,int maxlen)
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152 {
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153 unsigned char *start=NULL;
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154 int y,len=-1;
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155 while(len<minlen){
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reimar
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156 int len2=maxlen;
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caac2ca98168 4 - Implement a better way to calculate current audio pts and use it for
rtognimp
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157 double pts;
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rtognimp
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158 int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
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159 if(x<=0) break; // error
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rtognimp
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160 if (pts != MP_NOPTS_VALUE) {
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161 sh_audio->pts = pts;
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rtognimp
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162 sh_audio->pts_bytes = 0;
caac2ca98168 4 - Implement a better way to calculate current audio pts and use it for
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163 }
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1582297cc3d2 Use avcodec_decode_audio2 in ad_ffmpeg.c
reimar
parents: 21507
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164 y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x);
11977
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165 //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
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166 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
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167 if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
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168 if(len2>0){
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169 //len=len2;break;
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170 if(len<0) len=len2; else len+=len2;
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171 buf+=len2;
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1582297cc3d2 Use avcodec_decode_audio2 in ad_ffmpeg.c
reimar
parents: 21507
diff changeset
172 maxlen -= len2;
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caac2ca98168 4 - Implement a better way to calculate current audio pts and use it for
rtognimp
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173 sh_audio->pts_bytes += len2;
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174 }
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175 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
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176 }
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177 return len;
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178 }