Mercurial > mplayer.hg
annotate dec_audio.c @ 3473:8a46f6a9efd0
Preparing to next acceleration level
author | nick |
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date | Wed, 12 Dec 2001 18:48:07 +0000 |
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1 |
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2 #include <stdio.h> |
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3 #include <stdlib.h> |
1430 | 4 #include <unistd.h> |
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5 |
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6 #include "config.h" |
1567 | 7 #include "mp_msg.h" |
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8 #include "help_mp.h" |
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10 extern int verbose; // defined in mplayer.c |
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11 |
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12 #include "stream.h" |
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13 #include "demuxer.h" |
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14 |
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15 #include "codec-cfg.h" |
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16 #include "stheader.h" |
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17 |
2563 | 18 #include "dec_audio.h" |
19 | |
20 //========================================================================== | |
21 | |
22 #include "libao2/afmt.h" | |
23 | |
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24 #include "dll_init.h" |
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25 |
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26 #include "mp3lib/mp3.h" |
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27 #include "libac3/ac3.h" |
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28 |
3413 | 29 #include "liba52/a52.h" |
30 static sample_t * a52_samples; | |
31 static a52_state_t a52_state; | |
32 | |
3426 | 33 #include "g72x/g72x.h" |
34 static G72x_DATA g72x_data; | |
3413 | 35 |
883 | 36 #include "alaw.h" |
37 | |
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38 #include "xa/xa_gsm.h" |
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39 |
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40 #include "ac3-iec958.h" |
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41 |
2420 | 42 #include "ima4.h" |
43 | |
3145 | 44 #include "cpudetect.h" |
45 | |
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46 /* used for ac3surround decoder - set using -channels option */ |
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47 int audio_output_channels = 2; |
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48 |
2563 | 49 #ifdef USE_FAKE_MONO |
50 int fakemono=0; | |
51 #endif | |
52 | |
1291 | 53 #ifdef USE_DIRECTSHOW |
604 | 54 #include "loader/DirectShow/DS_AudioDec.h" |
3442 | 55 static void* ds_adec=NULL; |
1291 | 56 #endif |
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57 |
1828 | 58 #ifdef HAVE_OGGVORBIS |
59 /* XXX is math.h really needed? - atmos */ | |
60 #include <math.h> | |
61 #include <vorbis/codec.h> | |
2341 | 62 |
63 typedef struct ov_struct_st { | |
64 ogg_sync_state oy; /* sync and verify incoming physical bitstream */ | |
65 ogg_stream_state os; /* take physical pages, weld into a logical | |
66 stream of packets */ | |
67 ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */ | |
68 ogg_packet op; /* one raw packet of data for decode */ | |
3145 | 69 |
2341 | 70 vorbis_info vi; /* struct that stores all the static vorbis bitstream |
71 settings */ | |
72 vorbis_comment vc; /* struct that stores all the bitstream user comments */ | |
73 vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ | |
74 vorbis_block vb; /* local working space for packet->PCM decode */ | |
75 } ov_struct_t; | |
1828 | 76 #endif |
77 | |
1927 | 78 #ifdef USE_LIBAVCODEC |
2496 | 79 #ifdef USE_LIBAVCODEC_SO |
80 #include <libffmpeg/avcodec.h> | |
81 #else | |
1927 | 82 #include "libavcodec/avcodec.h" |
2496 | 83 #endif |
1927 | 84 static AVCodec *lavc_codec=NULL; |
85 static AVCodecContext lavc_context; | |
86 extern int avcodec_inited; | |
87 #endif | |
88 | |
2415 | 89 |
90 | |
91 #ifdef USE_LIBMAD | |
92 #include <mad.h> | |
93 static struct mad_stream mad_stream; | |
94 static struct mad_frame mad_frame; | |
95 static struct mad_synth mad_synth; | |
96 | |
97 // ensure buffer is filled with some data | |
98 static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length) | |
99 { | |
100 if(sh_audio->a_in_buffer_len < length) { | |
101 int len = demux_read_data(sh_audio->ds, sh_audio->a_in_buffer+sh_audio->a_in_buffer_len, length-sh_audio->a_in_buffer_len); | |
102 sh_audio->a_in_buffer_len += len; | |
103 } | |
104 } | |
105 | |
106 static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms) | |
107 { | |
108 int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer; | |
109 if(delta != 0) { | |
110 sh_audio->a_in_buffer_len -= delta; | |
111 memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len); | |
112 } | |
113 } | |
114 | |
115 | |
116 static inline | |
117 signed short mad_scale(mad_fixed_t sample) | |
118 { | |
119 /* round */ | |
120 sample += (1L << (MAD_F_FRACBITS - 16)); | |
121 | |
122 /* clip */ | |
123 if (sample >= MAD_F_ONE) | |
124 sample = MAD_F_ONE - 1; | |
125 else if (sample < -MAD_F_ONE) | |
126 sample = -MAD_F_ONE; | |
127 | |
128 /* quantize */ | |
129 return sample >> (MAD_F_FRACBITS + 1 - 16); | |
130 } | |
131 #endif | |
132 | |
133 | |
3413 | 134 static int a52_fillbuff(sh_audio_t *sh_audio){ |
135 int length=0; | |
136 int flags=0; | |
137 int sample_rate=0; | |
138 int bit_rate=0; | |
139 while(1){ | |
140 while(sh_audio->a_in_buffer_len<7){ | |
141 int c=demux_getc(sh_audio->ds); | |
142 if(c<0) return -1; // EOF | |
143 sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c; | |
144 } | |
145 length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); | |
146 if(!length){ | |
147 // bad file => resync | |
148 memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6); | |
149 --sh_audio->a_in_buffer_len; | |
150 continue; | |
151 } | |
152 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate); | |
153 if(length<7 || length>3840){ | |
154 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: invalid frame length: %d\n",length); | |
155 continue; | |
156 } | |
157 sh_audio->samplerate=sample_rate; | |
158 sh_audio->i_bps=bit_rate/8; | |
159 demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+7,length-7); | |
160 return length; | |
161 } | |
162 } | |
2415 | 163 |
1927 | 164 int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen); |
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165 |
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166 |
1289 | 167 static sh_audio_t* dec_audio_sh=NULL; |
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168 |
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169 // AC3 decoder buffer callback: |
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170 static void ac3_fill_buffer(uint8_t **start,uint8_t **end){ |
1289 | 171 int len=ds_get_packet(dec_audio_sh->ds,start); |
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172 //printf("<ac3:%d>\n",len); |
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173 if(len<0) |
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174 *start = *end = NULL; |
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175 else |
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176 *end = *start + len; |
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177 } |
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178 |
1289 | 179 // MP3 decoder buffer callback: |
180 int mplayer_audio_read(char *buf,int size){ | |
181 int len; | |
182 len=demux_read_data(dec_audio_sh->ds,buf,size); | |
183 return len; | |
184 } | |
185 | |
296 | 186 int init_audio(sh_audio_t *sh_audio){ |
303 | 187 int driver=sh_audio->codec->driver; |
296 | 188 |
189 sh_audio->samplesize=2; | |
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190 #if WORDS_BIGENDIAN |
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191 sh_audio->sample_format=AFMT_S16_BE; |
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192 #else |
758 | 193 sh_audio->sample_format=AFMT_S16_LE; |
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194 #endif |
303 | 195 sh_audio->samplerate=0; |
401 | 196 //sh_audio->pcm_bswap=0; |
1529 | 197 sh_audio->o_bps=0; |
303 | 198 |
758 | 199 sh_audio->a_buffer_size=0; |
303 | 200 sh_audio->a_buffer=NULL; |
296 | 201 |
746 | 202 sh_audio->a_in_buffer_len=0; |
203 | |
758 | 204 // setup required min. in/out buffer size: |
205 sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM | |
206 | |
207 switch(driver){ | |
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208 case AFM_ACM: |
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209 #ifndef USE_WIN32DLL |
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210 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport); |
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211 driver=0; |
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212 #else |
296 | 213 // Win32 ACM audio codec: |
214 if(init_acm_audio_codec(sh_audio)){ | |
746 | 215 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
296 | 216 sh_audio->channels=sh_audio->o_wf.nChannels; |
217 sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec; | |
758 | 218 // if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384; |
219 // sh_audio->a_buffer_size=sh_audio->audio_out_minsize; | |
220 // if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST) | |
221 // sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; | |
296 | 222 } else { |
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223 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror); |
303 | 224 driver=0; |
296 | 225 } |
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226 #endif |
758 | 227 break; |
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228 case AFM_DSHOW: |
296 | 229 #ifndef USE_DIRECTSHOW |
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230 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio); |
303 | 231 driver=0; |
296 | 232 #else |
233 // Win32 DShow audio codec: | |
340 | 234 // printf("DShow_audio: channs=%d rate=%d\n",sh_audio->channels,sh_audio->samplerate); |
3442 | 235 if(!(ds_adec=DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf))){ |
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236 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll); |
303 | 237 driver=0; |
238 } else { | |
746 | 239 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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240 sh_audio->channels=sh_audio->wf->nChannels; |
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241 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
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242 sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign; |
296 | 243 if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; |
244 sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; | |
245 sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); | |
246 sh_audio->a_in_buffer_len=0; | |
758 | 247 sh_audio->audio_out_minsize=16384; |
296 | 248 } |
249 #endif | |
758 | 250 break; |
1828 | 251 case AFM_VORBIS: |
252 #ifndef HAVE_OGGVORBIS | |
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253 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis); |
1828 | 254 driver=0; |
255 #else | |
256 /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */ | |
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257 sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame |
1828 | 258 #endif |
259 break; | |
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260 case AFM_PCM: |
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261 case AFM_DVDPCM: |
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262 case AFM_ALAW: |
758 | 263 // PCM, aLaw |
264 sh_audio->audio_out_minsize=2048; | |
265 break; | |
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266 case AFM_AC3: |
3413 | 267 case AFM_A52: |
758 | 268 // Dolby AC3 audio: |
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269 // however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame |
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270 sh_audio->audio_out_minsize=audio_output_channels*2*256*6; |
758 | 271 break; |
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272 case AFM_HWAC3: |
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273 // Dolby AC3 audio: |
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274 sh_audio->audio_out_minsize=4*256*6; |
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275 sh_audio->sample_format = AFMT_AC3; |
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276 break; |
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277 case AFM_GSM: |
758 | 278 // MS-GSM audio codec: |
279 sh_audio->audio_out_minsize=4*320; | |
280 break; | |
2420 | 281 case AFM_IMA4: |
282 // IMA-ADPCM 4:1 audio codec: | |
283 sh_audio->audio_out_minsize=4096; //4*IMA4_SAMPLES_PER_BLOCK; | |
284 sh_audio->ds->ss_div=IMA4_SAMPLES_PER_BLOCK; | |
285 sh_audio->ds->ss_mul=IMA4_BLOCK_SIZE; | |
286 break; | |
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287 case AFM_MPEG: |
758 | 288 // MPEG Audio: |
289 sh_audio->audio_out_minsize=4608; | |
290 break; | |
3426 | 291 case AFM_G72X: |
292 // g72x_reader_init(&g72x_data,G723_16_BITS_PER_SAMPLE); | |
293 g72x_reader_init(&g72x_data,G723_24_BITS_PER_SAMPLE); | |
294 // g72x_reader_init(&g72x_data,G721_32_BITS_PER_SAMPLE); | |
295 // g72x_reader_init(&g72x_data,G721_40_BITS_PER_SAMPLE); | |
296 sh_audio->audio_out_minsize=g72x_data.samplesperblock*4; | |
297 break; | |
1927 | 298 case AFM_FFMPEG: |
299 #ifndef USE_LIBAVCODEC | |
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300 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport); |
1927 | 301 return 0; |
302 #else | |
303 // FFmpeg Audio: | |
304 sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; | |
305 break; | |
306 #endif | |
2415 | 307 |
308 #ifdef USE_LIBMAD | |
309 case AFM_MAD: | |
310 printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__); | |
311 sh_audio->audio_out_minsize=4608; | |
312 if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; | |
313 sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; | |
314 sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); | |
315 sh_audio->a_in_buffer_len=0; | |
316 break; | |
317 #endif | |
296 | 318 } |
319 | |
303 | 320 if(!driver) return 0; |
296 | 321 |
322 // allocate audio out buffer: | |
758 | 323 sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc. |
324 | |
1567 | 325 mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n", |
758 | 326 sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size); |
327 | |
296 | 328 sh_audio->a_buffer=malloc(sh_audio->a_buffer_size); |
758 | 329 if(!sh_audio->a_buffer){ |
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330 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf); |
758 | 331 return 0; |
332 } | |
296 | 333 memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size); |
334 sh_audio->a_buffer_len=0; | |
335 | |
303 | 336 switch(driver){ |
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337 #ifdef USE_WIN32DLL |
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338 case AFM_ACM: { |
758 | 339 int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size); |
296 | 340 if(ret<0){ |
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341 mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret); |
303 | 342 driver=0; |
296 | 343 } |
640 | 344 sh_audio->a_buffer_len=ret; |
303 | 345 break; |
296 | 346 } |
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347 #endif |
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348 case AFM_PCM: { |
296 | 349 // AVI PCM Audio: |
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350 WAVEFORMATEX *h=sh_audio->wf; |
746 | 351 sh_audio->i_bps=h->nAvgBytesPerSec; |
296 | 352 sh_audio->channels=h->nChannels; |
353 sh_audio->samplerate=h->nSamplesPerSec; | |
354 sh_audio->samplesize=(h->wBitsPerSample+7)/8; | |
758 | 355 switch(sh_audio->format){ // hardware formats: |
356 case 0x6: sh_audio->sample_format=AFMT_A_LAW;break; | |
357 case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break; | |
358 case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break; | |
359 case 0x50: sh_audio->sample_format=AFMT_MPEG;break; | |
2547 | 360 case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break; |
758 | 361 // case 0x2000: sh_audio->sample_format=AFMT_AC3; |
362 default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8; | |
363 } | |
401 | 364 break; |
365 } | |
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366 case AFM_DVDPCM: { |
401 | 367 // DVD PCM Audio: |
368 sh_audio->channels=2; | |
369 sh_audio->samplerate=48000; | |
746 | 370 sh_audio->i_bps=2*2*48000; |
296 | 371 // sh_audio->pcm_bswap=1; |
303 | 372 break; |
373 } | |
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374 case AFM_AC3: { |
296 | 375 // Dolby AC3 audio: |
1289 | 376 dec_audio_sh=sh_audio; // save sh_audio for the callback: |
296 | 377 ac3_config.fill_buffer_callback = ac3_fill_buffer; |
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378 ac3_config.num_output_ch = audio_output_channels; |
296 | 379 ac3_config.flags = 0; |
3145 | 380 if(gCpuCaps.hasMMX){ |
296 | 381 ac3_config.flags |= AC3_MMX_ENABLE; |
3145 | 382 } |
383 if(gCpuCaps.has3DNow){ | |
296 | 384 ac3_config.flags |= AC3_3DNOW_ENABLE; |
3145 | 385 } |
296 | 386 ac3_init(); |
387 sh_audio->ac3_frame = ac3_decode_frame(); | |
388 if(sh_audio->ac3_frame){ | |
746 | 389 ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame; |
390 sh_audio->samplerate=fr->sampling_rate; | |
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391 sh_audio->channels=ac3_config.num_output_ch; |
746 | 392 // 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples |
393 //sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256); | |
394 sh_audio->i_bps=fr->bit_rate*(1000/8); | |
303 | 395 } else { |
396 driver=0; // bad frame -> disable audio | |
397 } | |
398 break; | |
399 } | |
3413 | 400 case AFM_A52: { |
401 // Dolby AC3 audio: | |
402 int accel=0; // should contain mmx/sse/etc flags | |
403 a52_samples=a52_init (accel); | |
404 if (a52_samples == NULL) { | |
405 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); | |
406 driver=0;break; | |
407 } | |
408 sh_audio->a_in_buffer_size=3840; | |
409 sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); | |
410 sh_audio->a_in_buffer_len=0; | |
411 if(a52_fillbuff(sh_audio)<0){ | |
412 mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); | |
413 driver=0;break; | |
414 } | |
415 sh_audio->channels=audio_output_channels; | |
416 break; | |
417 } | |
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418 case AFM_HWAC3: { |
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419 unsigned char *buffer; |
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420 struct hwac3info ai; |
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421 int len, skipped; |
1529 | 422 len = ds_get_packet(sh_audio->ds, &buffer); // maybe 1 packet is not enough, |
423 // at least for mpeg, PS packets contain about max. 2000 bytes of data. | |
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424 if(ac3_iec958_parse_syncinfo(buffer, len, &ai, &skipped) < 0) { |
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425 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_AC3notvalid); |
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426 driver = 0; |
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427 break; |
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428 } |
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429 if(ai.samplerate != 48000) { |
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430 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_AC3only48k); |
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431 driver = 0; |
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432 break; |
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433 } |
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434 sh_audio->samplerate=ai.samplerate; |
1529 | 435 sh_audio->samplesize=ai.framesize; |
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436 sh_audio->channels=1; |
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437 sh_audio->i_bps=ai.bitrate*(1000/8); |
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438 sh_audio->ac3_frame=malloc(6144); |
1529 | 439 sh_audio->o_bps=sh_audio->i_bps; // XXX FIXME!!! XXX |
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440 break; |
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441 } |
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442 case AFM_ALAW: { |
296 | 443 // aLaw audio codec: |
432
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444 sh_audio->channels=sh_audio->wf->nChannels; |
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445 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
746 | 446 sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate; |
303 | 447 break; |
448 } | |
3426 | 449 case AFM_G72X: { |
450 // GSM 723 audio codec: | |
451 sh_audio->channels=sh_audio->wf->nChannels; | |
452 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
453 sh_audio->i_bps=(sh_audio->samplerate/g72x_data.samplesperblock)*g72x_data.blocksize; | |
454 break; | |
455 } | |
1927 | 456 #ifdef USE_LIBAVCODEC |
457 case AFM_FFMPEG: { | |
458 int x; | |
459 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
460 if(!avcodec_inited){ | |
461 avcodec_init(); | |
462 avcodec_register_all(); | |
463 avcodec_inited=1; | |
464 } | |
465 lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll); | |
466 if(!lavc_codec){ | |
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467 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
1927 | 468 return 0; |
469 } | |
470 memset(&lavc_context, 0, sizeof(lavc_context)); | |
471 /* open it */ | |
472 if (avcodec_open(&lavc_context, lavc_codec) < 0) { | |
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473 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
1927 | 474 return 0; |
475 } | |
476 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n"); | |
477 | |
478 // Decode at least 1 byte: (to get header filled) | |
479 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); | |
480 if(x>0) sh_audio->a_buffer_len=x; | |
481 | |
482 #if 1 | |
483 sh_audio->channels=lavc_context.channels; | |
484 sh_audio->samplerate=lavc_context.sample_rate; | |
485 sh_audio->i_bps=lavc_context.bit_rate/8; | |
486 #else | |
487 sh_audio->channels=sh_audio->wf->nChannels; | |
488 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
489 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; | |
490 #endif | |
491 break; | |
492 } | |
493 #endif | |
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494 case AFM_GSM: { |
296 | 495 // MS-GSM audio codec: |
496 GSM_Init(); | |
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497 sh_audio->channels=sh_audio->wf->nChannels; |
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498 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
746 | 499 // decodes 65 byte -> 320 short |
500 // 1 sec: sh_audio->channels*sh_audio->samplerate samples | |
501 // 1 frame: 320 samples | |
502 sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320; // 1:10 | |
303 | 503 break; |
296 | 504 } |
2420 | 505 case AFM_IMA4: { |
506 // IMA-ADPCM 4:1 audio codec: | |
507 sh_audio->channels=sh_audio->wf->nChannels; | |
508 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
509 // decodes 34 byte -> 64 short | |
510 sh_audio->i_bps=IMA4_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/IMA4_SAMPLES_PER_BLOCK; // 1:4 | |
511 break; | |
512 } | |
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513 case AFM_MPEG: { |
296 | 514 // MPEG Audio: |
1289 | 515 dec_audio_sh=sh_audio; // save sh_audio for the callback: |
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|
516 #ifdef USE_FAKE_MONO |
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517 MP3_Init(fakemono); |
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518 #else |
296 | 519 MP3_Init(); |
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520 #endif |
296 | 521 MP3_samplerate=MP3_channels=0; |
522 // printf("[\n"); | |
523 sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1); | |
524 // printf("]\n"); | |
525 sh_audio->channels=2; // hack | |
526 sh_audio->samplerate=MP3_samplerate; | |
746 | 527 sh_audio->i_bps=MP3_bitrate*(1000/8); |
303 | 528 break; |
529 } | |
1828 | 530 #ifdef HAVE_OGGVORBIS |
531 case AFM_VORBIS: { | |
532 // OggVorbis Audio: | |
533 #if 0 /* just here for reference - atmos */ | |
534 ogg_sync_state oy; /* sync and verify incoming physical bitstream */ | |
535 ogg_stream_state os; /* take physical pages, weld into a logical | |
536 stream of packets */ | |
537 ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */ | |
538 ogg_packet op; /* one raw packet of data for decode */ | |
539 | |
540 vorbis_info vi; /* struct that stores all the static vorbis bitstream | |
541 settings */ | |
542 vorbis_comment vc; /* struct that stores all the bitstream user comments */ | |
543 vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ | |
544 vorbis_block vb; /* local working space for packet->PCM decode */ | |
545 #else | |
546 /* nix, nada, rien, nothing, nem, nüx */ | |
547 #endif | |
548 | |
549 uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header, | |
550 maybe use ogg_uint32_t */ | |
551 //int i; | |
552 int ret; | |
553 char *buffer; | |
554 ogg_packet hdr; | |
555 //ov_struct_t *s=&sh_audio->ov; | |
556 sh_audio->ov=malloc(sizeof(ov_struct_t)); | |
557 //s=&sh_audio->ov; | |
558 | |
559 vorbis_info_init(&sh_audio->ov->vi); | |
560 vorbis_comment_init(&sh_audio->ov->vc); | |
561 | |
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562 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize); |
1828 | 563 memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t)); |
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564 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]); |
1828 | 565 /*for(i=12; i <= 40; i+=2) { // header bruteforce :) |
566 memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t)); | |
567 printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]); | |
568 }*/ | |
569 | |
570 /* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos | |
571 hdr.packet=NULL; | |
572 hdr.b_o_s = 1; /* beginning of stream for first packet */ | |
573 hdr.bytes = hdrsizes[0]; | |
574 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
575 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes); | |
576 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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577 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n"); |
1828 | 578 hdr.b_o_s = 0; |
579 hdr.bytes = hdrsizes[1]; | |
580 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
581 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes); | |
582 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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diff
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|
583 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n"); |
1828 | 584 hdr.bytes = hdrsizes[2]; |
585 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
586 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes); | |
587 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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diff
changeset
|
588 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n"); |
1828 | 589 hdr.bytes=0; |
590 hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */ | |
591 /* done with the headers */ | |
592 | |
593 | |
594 /* Throw the comments plus a few lines about the bitstream we're | |
595 decoding */ | |
596 { | |
597 char **ptr=sh_audio->ov->vc.user_comments; | |
598 while(*ptr){ | |
1837
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parents:
1834
diff
changeset
|
599 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr); |
1828 | 600 ++ptr; |
601 } | |
1837
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arpi
parents:
1834
diff
changeset
|
602 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",sh_audio->ov->vi.channels,sh_audio->ov->vi.rate,sh_audio->ov->vi.bitrate_nominal/1000, (sh_audio->ov->vi.bitrate_lower!=sh_audio->ov->vi.bitrate_nominal)||(sh_audio->ov->vi.bitrate_upper!=sh_audio->ov->vi.bitrate_nominal)?'V':'C'); |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
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|
603 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor); |
1828 | 604 } |
605 sh_audio->channels=sh_audio->ov->vi.channels; | |
606 sh_audio->samplerate=sh_audio->ov->vi.rate; | |
607 sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8; | |
608 | |
609 // printf("[\n"); | |
1837
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diff
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|
610 // sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels; |
1828 | 611 // printf("]\n"); |
612 | |
613 /* OK, got and parsed all three headers. Initialize the Vorbis | |
614 packet->PCM decoder. */ | |
615 vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */ | |
616 vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode | |
617 so multiple block decodes can | |
618 proceed in parallel. We could init | |
619 multiple vorbis_block structures | |
620 for vd here */ | |
621 //printf("OggVorbis: synthesis and block init done.\n"); | |
622 ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */ | |
623 | |
624 while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) { | |
625 if(ret == -1) | |
1837
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diff
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|
626 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n"); |
1828 | 627 else |
628 if(ret == 0) { | |
1837
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diff
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|
629 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n"); |
1828 | 630 /* submit a a_buffer_len block to libvorbis' Ogg layer */ |
1837
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diff
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|
631 buffer=ogg_sync_buffer(&sh_audio->ov->oy,256); |
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ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
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diff
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|
632 ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256)); |
1828 | 633 } |
634 } | |
1837
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parents:
1834
diff
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|
635 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n"); |
1834
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Now oggvorbis decoding properly works, but a/v sync is bad.
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636 ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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637 as they'll also become apparent |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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|
638 at packetout */ |
1828 | 639 |
640 /* Get the serial number and set up the rest of decode. */ | |
641 /* serialno first; use it to set up a logical stream */ | |
642 ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og)); | |
643 | |
1837
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parents:
1834
diff
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|
644 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n"); |
1828 | 645 |
646 break; | |
647 } | |
648 #endif | |
2415 | 649 |
650 #ifdef USE_LIBMAD | |
651 case AFM_MAD: | |
652 { | |
653 printf(__FILE__ ":%d:mad: initialising\n", __LINE__); | |
654 mad_frame_init(&mad_frame); | |
655 mad_stream_init(&mad_stream); | |
656 | |
657 printf(__FILE__ ":%d:mad: preparing buffer\n", __LINE__); | |
658 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
659 mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len); | |
660 mad_stream_sync(&mad_stream); | |
661 mad_synth_init(&mad_synth); | |
662 | |
663 if(mad_frame_decode(&mad_frame, &mad_stream) == 0) | |
664 { | |
665 printf(__FILE__ ":%d:mad: post processing buffer\n", __LINE__); | |
666 mad_postprocess_buffer(sh_audio, &mad_stream); | |
667 } | |
668 else | |
669 { | |
670 printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__); | |
671 } | |
672 | |
2434
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
673 switch (mad_frame.header.mode) |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
674 { |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
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diff
changeset
|
675 case MAD_MODE_SINGLE_CHANNEL: |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
676 sh_audio->channels=1; |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
677 break; |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
678 case MAD_MODE_DUAL_CHANNEL: |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
679 case MAD_MODE_JOINT_STEREO: |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
680 case MAD_MODE_STEREO: |
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audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
681 sh_audio->channels=2; |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
682 break; |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
683 default: |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
684 mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n"); |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
685 } |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
686 mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n", |
571538d73e08
audio channels nr is identified from libmad's channel mode
alex
parents:
2432
diff
changeset
|
687 sh_audio->channels, mad_frame.header.mode); |
2432 | 688 /* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */ |
689 #if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13) | |
690 sh_audio->samplerate=mad_frame.header.samplerate; | |
691 #else | |
2415 | 692 sh_audio->samplerate=mad_frame.header.sfreq; |
2432 | 693 #endif |
2415 | 694 sh_audio->i_bps=mad_frame.header.bitrate; |
695 printf(__FILE__ ":%d:mad: continuing\n", __LINE__); | |
696 break; | |
697 } | |
698 #endif | |
296 | 699 } |
700 | |
701 if(!sh_audio->channels || !sh_audio->samplerate){ | |
1973
5216f108cb4f
all error/warn/info messages moved to help_mp-en.h for translation
arpi
parents:
1927
diff
changeset
|
702 mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio); |
303 | 703 driver=0; |
296 | 704 } |
705 | |
303 | 706 if(!driver){ |
707 if(sh_audio->a_buffer) free(sh_audio->a_buffer); | |
1313
d797ecb9778f
sh_audio->a_buffer is freed twice (here and in mplayer.c), can result to
jkeil
parents:
1309
diff
changeset
|
708 sh_audio->a_buffer=NULL; |
303 | 709 return 0; |
710 } | |
296 | 711 |
1529 | 712 if(!sh_audio->o_bps) |
303 | 713 sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize; |
714 return driver; | |
296 | 715 } |
291 | 716 |
746 | 717 // Audio decoding: |
291 | 718 |
746 | 719 // Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc) |
720 // buffer length is 'maxlen' bytes, it shouldn't be exceeded... | |
721 | |
722 int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){ | |
291 | 723 int len=-1; |
303 | 724 switch(sh_audio->codec->driver){ |
1927 | 725 #ifdef USE_LIBAVCODEC |
726 case AFM_FFMPEG: { | |
727 unsigned char *start=NULL; | |
728 int y; | |
729 while(len<minlen){ | |
730 int len2=0; | |
731 int x=ds_get_packet(sh_audio->ds,&start); | |
732 if(x<=0) break; // error | |
2049
df41903fd7d7
VfW stuff moved to dll_init, warnings fixed, using dll_init.h
arpi
parents:
1973
diff
changeset
|
733 y=avcodec_decode_audio(&lavc_context,(INT16*)buf,&len2,start,x); |
1973
5216f108cb4f
all error/warn/info messages moved to help_mp-en.h for translation
arpi
parents:
1927
diff
changeset
|
734 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
1927 | 735 if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
736 if(len2>0){ | |
737 //len=len2;break; | |
738 if(len<0) len=len2; else len+=len2; | |
739 buf+=len2; | |
740 } | |
741 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); | |
742 } | |
743 } | |
744 break; | |
745 #endif | |
1410
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
746 case AFM_MPEG: // MPEG layer 2 or 3 |
291 | 747 len=MP3_DecodeFrame(buf,-1); |
1485
b895f95e7657
AVI demuxer cleanups, fileformat-dependent stuff moved to priv_t
arpi
parents:
1456
diff
changeset
|
748 // len=MP3_DecodeFrame(buf,3); |
291 | 749 break; |
1830 | 750 #ifdef HAVE_OGGVORBIS |
1828 | 751 case AFM_VORBIS: { // OggVorbis |
752 /* note: good minlen would be 4k or 8k IMHO - atmos */ | |
753 int ret; | |
754 char *buffer; | |
755 int bytes; | |
756 int samples; | |
757 float **pcm; | |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
758 //ogg_int16_t convbuffer[4096]; |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
759 // int convsize; |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
760 int readlen=1024; |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
761 len=0; |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
762 // convsize=minlen/sh_audio->ov->vi.channels; |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
763 |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
764 while(len < minlen) { /* double loop allows for break in inner loop */ |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
765 while(len < minlen) { /* without aborting the outer loop - atmos */ |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
766 ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
767 if(ret==0) { |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
768 int xxx=0; |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
769 //printf("OggVorbis: Packetout: need more data, paging!\n"); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
770 while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) { |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
771 if(ret == -1) |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
772 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n"); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
773 else |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
774 if(ret == 0) { |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
775 //printf("OggVorbis: Pageout: need more data to verify page, reading more data.\n"); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
776 /* submit a readlen k block to libvorbis' Ogg layer */ |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
777 buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
778 bytes=demux_read_data(sh_audio->ds,buffer,readlen); |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
779 xxx+=bytes; |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
780 ogg_sync_wrote(&sh_audio->ov->oy,bytes); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
781 if(bytes==0) |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
782 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n"); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
783 } |
1828 | 784 } |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
785 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
786 //printf("OggVorbis: Pageout: successfull, pagin in.\n"); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
787 if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0) |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
788 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n"); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
789 break; |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
790 } else if(ret<0) { |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
791 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Packetout: missing or corrupt data, skipping packet!\n"); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
792 break; |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
atmos4
parents:
1831
diff
changeset
|
793 } else { |
1828 | 794 |
795 /* we have a packet. Decode it */ | |
796 | |
797 if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */ | |
798 vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->vb); | |
799 | |
800 /* **pcm is a multichannel float vector. In stereo, for | |
801 example, pcm[0] is left, and pcm[1] is right. samples is | |
802 the size of each channel. Convert the float values | |
803 (-1.<=range<=1.) to whatever PCM format and write it out */ | |
804 | |
805 while((samples=vorbis_synthesis_pcmout(&sh_audio->ov->vd,&pcm))>0){ | |
806 int i,j; | |
807 int clipflag=0; | |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
808 int convsize=(maxlen-len)/(2*sh_audio->ov->vi.channels); // max size! |
1828 | 809 int bout=(samples<convsize?samples:convsize); |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
810 |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
811 if(bout<=0) break; |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
812 |
1828 | 813 /* convert floats to 16 bit signed ints (host order) and |
814 interleave */ | |
815 for(i=0;i<sh_audio->ov->vi.channels;i++){ | |
1837
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
816 ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); |
1828 | 817 ogg_int16_t *ptr=convbuffer+i; |
818 float *mono=pcm[i]; | |
819 for(j=0;j<bout;j++){ | |
820 #if 1 | |
821 int val=mono[j]*32767.f; | |
822 #else /* optional dither */ | |
823 int val=mono[j]*32767.f+drand48()-0.5f; | |
824 #endif | |
825 /* might as well guard against clipping */ | |
826 if(val>32767){ | |
827 val=32767; | |
828 clipflag=1; | |
829 } | |
830 if(val<-32768){ | |
831 val=-32768; | |
832 clipflag=1; | |
833 } | |
834 *ptr=val; | |
835 ptr+=sh_audio->ov->vi.channels; | |
836 } | |
837 } | |
838 | |
839 if(clipflag) | |
1837
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840 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence)); |
1828 | 841 |
842 //fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging | |
1837
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843 //memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout); |
1834
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844 len+=2*sh_audio->ov->vi.channels*bout; |
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845 |
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846 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples); |
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847 |
1828 | 848 vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how |
849 many samples we | |
850 actually consumed */ | |
851 } | |
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852 } // from else, packetout ok |
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853 } // while len |
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854 } // outer while len |
1828 | 855 if(ogg_page_eos(&sh_audio->ov->og)) |
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856 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos |
1828 | 857 |
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858 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len); |
1828 | 859 |
860 break; | |
1831 | 861 } |
1830 | 862 #endif |
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863 case AFM_PCM: // AVI PCM |
1058 | 864 len=demux_read_data(sh_audio->ds,buf,minlen); |
401 | 865 break; |
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866 case AFM_DVDPCM: // DVD PCM |
401 | 867 { int j; |
746 | 868 len=demux_read_data(sh_audio->ds,buf,minlen); |
291 | 869 //if(i&1){ printf("Warning! pcm_audio_size&1 !=0 (%d)\n",i);i&=~1; } |
746 | 870 // swap endian: |
291 | 871 for(j=0;j<len;j+=2){ |
872 char x=buf[j]; | |
873 buf[j]=buf[j+1]; | |
874 buf[j+1]=x; | |
875 } | |
876 break; | |
877 } | |
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878 case AFM_ALAW: // aLaw decoder |
746 | 879 { int l=demux_read_data(sh_audio->ds,buf,minlen/2); |
291 | 880 unsigned short *d=(unsigned short *) buf; |
881 unsigned char *s=buf; | |
882 len=2*l; | |
758 | 883 if(sh_audio->format==6){ |
884 // aLaw | |
883 | 885 while(l>0){ --l; d[l]=alaw2short[s[l]]; } |
758 | 886 } else { |
887 // uLaw | |
883 | 888 while(l>0){ --l; d[l]=ulaw2short[s[l]]; } |
291 | 889 } |
890 break; | |
891 } | |
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892 case AFM_GSM: // MS-GSM decoder |
2420 | 893 { unsigned char ibuf[65]; // 65 bytes / frame |
894 if(demux_read_data(sh_audio->ds,ibuf,65)!=65) break; // EOF | |
895 XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short | |
746 | 896 // XA_GSM_Decoder(buf,(unsigned short *) &sh_audio->a_buffer[sh_audio->a_buffer_len]); // decodes 33 byte -> 160 short |
897 len=2*320; | |
291 | 898 break; |
899 } | |
3426 | 900 case AFM_G72X: // GSM 723 decoder |
901 { if(demux_read_data(sh_audio->ds,g72x_data.block, g72x_data.blocksize)!=g72x_data.blocksize) break; // EOF | |
902 g72x_decode_block(&g72x_data); | |
903 len=2*g72x_data.samplesperblock; | |
904 memcpy(buf,g72x_data.samples,len); | |
905 break; | |
906 } | |
2420 | 907 case AFM_IMA4: // IMA-ADPCM 4:1 audio codec: |
908 { unsigned char ibuf[IMA4_BLOCK_SIZE]; // bytes / frame | |
909 if(demux_read_data(sh_audio->ds,ibuf,IMA4_BLOCK_SIZE)!=IMA4_BLOCK_SIZE) break; // EOF | |
2424 | 910 len=2*ima4_decode_block((unsigned short*)buf,ibuf,2*IMA4_SAMPLES_PER_BLOCK); |
2420 | 911 break; |
912 } | |
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913 case AFM_AC3: // AC3 decoder |
291 | 914 //printf("{1:%d}",avi_header.idx_pos);fflush(stdout); |
915 if(!sh_audio->ac3_frame) sh_audio->ac3_frame=ac3_decode_frame(); | |
916 //printf("{2:%d}",avi_header.idx_pos);fflush(stdout); | |
917 if(sh_audio->ac3_frame){ | |
296 | 918 len = 256 * 6 *sh_audio->channels*sh_audio->samplesize; |
587
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919 memcpy(buf,((ac3_frame_t*)sh_audio->ac3_frame)->audio_data,len); |
291 | 920 sh_audio->ac3_frame=NULL; |
921 } | |
922 //printf("{3:%d}",avi_header.idx_pos);fflush(stdout); | |
923 break; | |
3413 | 924 case AFM_A52: { // AC3 decoder |
925 int flags=0; | |
926 int i; | |
927 sample_t level=1, bias=384; | |
928 if(!sh_audio->a_in_buffer_len) | |
929 if(a52_fillbuff(sh_audio)<0) break; // EOF | |
930 switch(sh_audio->channels){ | |
931 case 1: flags=A52_MONO; break; | |
932 // case 2: flags=A52_STEREO; break; | |
933 case 2: flags=A52_DOLBY; break; | |
934 // case 3: flags=A52_3F; break; | |
935 case 3: flags=A52_2F1R; break; | |
936 case 4: flags=A52_2F2R; break; // 2+2 | |
937 case 5: flags=A52_3F2R; break; | |
938 case 6: flags=A52_3F2R|A52_LFE; break; // 5.1 | |
939 } | |
940 flags|=A52_ADJUST_LEVEL; | |
941 sh_audio->a_in_buffer_len=0; | |
942 if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ | |
943 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n"); | |
944 break; | |
945 } | |
946 // a52_dynrng (&state, NULL, NULL); // disable dynamic range compensation | |
947 | |
948 // frame decoded, let's resample: | |
949 a52_resample_init(flags,sh_audio->channels); | |
950 len=0; | |
951 for (i = 0; i < 6; i++) { | |
952 if (a52_block (&a52_state, a52_samples)){ | |
953 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n"); | |
954 break; | |
955 } | |
956 len+=2*a52_resample(a52_samples,&buf[len]); | |
957 } | |
3417 | 958 // printf("len = %d \n",len); // 6144 on all vobs I tried so far... (5.1 and 2.0) ::atmos |
3413 | 959 break; |
960 } | |
1528
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961 case AFM_HWAC3: // AC3 through SPDIF |
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962 if(demux_read_data(sh_audio->ds,sh_audio->ac3_frame, 6144) != 6144) |
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963 break; //EOF |
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964 ac3_iec958_build_burst(1536, 0x1F, 1, buf, sh_audio->ac3_frame); |
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965 len = 6144; |
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966 break; |
1517
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967 #ifdef USE_WIN32DLL |
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968 case AFM_ACM: |
758 | 969 // len=sh_audio->audio_out_minsize; // optimal decoded fragment size |
970 // if(len<minlen) len=minlen; else | |
971 // if(len>maxlen) len=maxlen; | |
972 // len=acm_decode_audio(sh_audio,buf,len); | |
973 len=acm_decode_audio(sh_audio,buf,minlen,maxlen); | |
291 | 974 break; |
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975 #endif |
746 | 976 |
291 | 977 #ifdef USE_DIRECTSHOW |
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978 case AFM_DSHOW: // DirectShow |
1327
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|
979 { int size_in=0; |
291 | 980 int size_out=0; |
3442 | 981 int srcsize=DS_AudioDecoder_GetSrcSize(ds_adec, maxlen); |
1567 | 982 mp_msg(MSGT_DECAUDIO,MSGL_DBG3,"DShow says: srcsize=%d (buffsize=%d) out_size=%d\n",srcsize,sh_audio->a_in_buffer_size,maxlen); |
291 | 983 if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!! |
984 if(sh_audio->a_in_buffer_len<srcsize){ | |
985 sh_audio->a_in_buffer_len+= | |
986 demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len], | |
987 srcsize-sh_audio->a_in_buffer_len); | |
988 } | |
3442 | 989 DS_AudioDecoder_Convert(ds_adec, sh_audio->a_in_buffer,sh_audio->a_in_buffer_len, |
291 | 990 buf,maxlen, &size_in,&size_out); |
1567 | 991 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"DShow: audio %d -> %d converted (in_buf_len=%d of %d) %d\n",size_in,size_out,sh_audio->a_in_buffer_len,sh_audio->a_in_buffer_size,ds_tell_pts(sh_audio->ds)); |
291 | 992 if(size_in>=sh_audio->a_in_buffer_len){ |
993 sh_audio->a_in_buffer_len=0; | |
994 } else { | |
995 sh_audio->a_in_buffer_len-=size_in; | |
996 memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len); | |
997 } | |
998 len=size_out; | |
999 break; | |
1000 } | |
1001 #endif | |
2415 | 1002 |
1003 #ifdef USE_LIBMAD | |
1004 case AFM_MAD: | |
1005 { | |
1006 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
1007 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
1008 if(mad_frame_decode(&mad_frame, &mad_stream) == 0) | |
1009 { | |
1010 mad_synth_frame(&mad_synth, &mad_frame); | |
1011 mad_postprocess_buffer(sh_audio, &mad_stream); | |
1012 | |
1013 /* and fill buffer */ | |
1014 | |
1015 { | |
1016 int i; | |
1017 int end_size = mad_synth.pcm.length; | |
1018 signed short* samples = (signed short*)buf; | |
1019 if(end_size > maxlen/4) | |
1020 end_size=maxlen/4; | |
1021 | |
1022 for(i=0; i<mad_synth.pcm.length; ++i) { | |
1023 *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); | |
1024 *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); | |
1025 // *buf++ = mad_scale(mad_synth.pcm.sampAles[1][i]); | |
1026 } | |
1027 len = end_size*4; | |
1028 } | |
1029 } | |
1030 else | |
1031 { | |
1032 printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__); | |
1033 } | |
1034 | |
1035 break; | |
1036 } | |
1037 #endif | |
291 | 1038 } |
1039 return len; | |
1040 } | |
1041 | |
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1042 void resync_audio_stream(sh_audio_t *sh_audio){ |
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1043 switch(sh_audio->codec->driver){ |
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1044 case AFM_MPEG: |
1327
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1045 MP3_DecodeFrame(NULL,-2); // resync |
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1046 MP3_DecodeFrame(NULL,-2); // resync |
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1047 MP3_DecodeFrame(NULL,-2); // resync |
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1048 break; |
1849
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1049 #ifdef HAVE_OGGVORBIS |
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1050 case AFM_VORBIS: |
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1051 //printf("OggVorbis: resetting stream.\n"); |
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1052 ogg_sync_reset(&sh_audio->ov->oy); |
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1053 ogg_stream_reset(&sh_audio->ov->os); |
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1054 break; |
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1055 #endif |
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1056 case AFM_AC3: |
1327
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1057 ac3_bitstream_reset(); // reset AC3 bitstream buffer |
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1058 // if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);} |
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1059 sh_audio->ac3_frame=ac3_decode_frame(); // resync |
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1060 // if(verbose) printf(" OK!\n"); |
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1061 break; |
3413 | 1062 case AFM_A52: |
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1063 case AFM_ACM: |
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1064 case AFM_DSHOW: |
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1065 case AFM_HWAC3: |
1327
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1066 sh_audio->a_in_buffer_len=0; // reset ACM/DShow audio buffer |
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1067 break; |
2415 | 1068 |
1069 #ifdef USE_LIBMAD | |
1070 case AFM_MAD: | |
1071 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
1072 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
1073 mad_stream_sync(&mad_stream); | |
1074 mad_postprocess_buffer(sh_audio, &mad_stream); | |
1075 break; | |
2418 | 1076 #endif |
1327
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1077 } |
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1078 } |
291 | 1079 |
1327
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1080 void skip_audio_frame(sh_audio_t *sh_audio){ |
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1081 switch(sh_audio->codec->driver){ |
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1082 case AFM_MPEG: MP3_DecodeFrame(NULL,-2);break; // skip MPEG frame |
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1083 case AFM_AC3: sh_audio->ac3_frame=ac3_decode_frame();break; // skip AC3 frame |
3413 | 1084 case AFM_A52: a52_fillbuff(sh_audio);break; // skip AC3 frame |
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1085 case AFM_ACM: |
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1086 case AFM_DSHOW: { |
1408 | 1087 int skip=sh_audio->wf->nBlockAlign; |
1088 if(skip<16){ | |
1089 skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7); | |
1090 if(skip<16) skip=16; | |
1091 } | |
1092 demux_read_data(sh_audio->ds,NULL,skip); | |
1093 break; | |
1094 } | |
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1095 case AFM_PCM: |
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|
1096 case AFM_DVDPCM: |
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1097 case AFM_ALAW: { |
1408 | 1098 int skip=sh_audio->i_bps/16; |
1099 skip=skip&(~3); | |
1100 demux_read_data(sh_audio->ds,NULL,skip); | |
1101 break; | |
1102 } | |
2415 | 1103 #ifdef USE_LIBMAD |
1104 case AFM_MAD: | |
1105 { | |
1106 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
1107 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
1108 mad_stream_skip(&mad_stream, 2); | |
1109 mad_stream_sync(&mad_stream); | |
1110 mad_postprocess_buffer(sh_audio, &mad_stream); | |
1111 break; | |
1112 } | |
1113 #endif | |
1114 | |
1327
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
1320
diff
changeset
|
1115 default: ds_fill_buffer(sh_audio->ds); // skip PCM frame |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
1320
diff
changeset
|
1116 } |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
1320
diff
changeset
|
1117 } |