annotate libmpcodecs/ad_ffmpeg.c @ 24402:93b87066f9da

cosmetics: Add comments to some #endif preprocessor directives.
author diego
date Mon, 10 Sep 2007 16:17:13 +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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diego
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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 {
11977
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michael
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142 AVCodecContext *lavc_context = sh->context;
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michael
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143 switch(cmd){
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michael
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144 case ADCTRL_RESYNC_STREAM:
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michael
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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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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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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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160 if (pts != MP_NOPTS_VALUE) {
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161 sh_audio->pts = pts;
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162 sh_audio->pts_bytes = 0;
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163 }
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1582297cc3d2 Use avcodec_decode_audio2 in ad_ffmpeg.c
reimar
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164 y=avcodec_decode_audio2(sh_audio->context,(int16_t*)buf,&len2,start,x);
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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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reimar
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172 maxlen -= len2;
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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 }