Mercurial > mplayer.hg
annotate dec_audio.c @ 2637:8101229be3a5
code cleanup
author | nick |
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date | Fri, 02 Nov 2001 18:24:25 +0000 |
parents | fc124f9db88e |
children | 78e11396f431 |
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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" |
1567 | 9 |
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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 |
883 | 29 #include "alaw.h" |
30 | |
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31 #include "xa/xa_gsm.h" |
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32 |
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33 #include "ac3-iec958.h" |
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34 |
2420 | 35 #include "ima4.h" |
36 | |
2563 | 37 #ifdef USE_FAKE_MONO |
38 int fakemono=0; | |
39 #endif | |
40 | |
1291 | 41 #ifdef USE_DIRECTSHOW |
604 | 42 #include "loader/DirectShow/DS_AudioDec.h" |
1291 | 43 #endif |
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44 |
1828 | 45 #ifdef HAVE_OGGVORBIS |
46 /* XXX is math.h really needed? - atmos */ | |
47 #include <math.h> | |
48 #include <vorbis/codec.h> | |
2341 | 49 |
50 typedef struct ov_struct_st { | |
51 ogg_sync_state oy; /* sync and verify incoming physical bitstream */ | |
52 ogg_stream_state os; /* take physical pages, weld into a logical | |
53 stream of packets */ | |
54 ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */ | |
55 ogg_packet op; /* one raw packet of data for decode */ | |
56 | |
57 vorbis_info vi; /* struct that stores all the static vorbis bitstream | |
58 settings */ | |
59 vorbis_comment vc; /* struct that stores all the bitstream user comments */ | |
60 vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ | |
61 vorbis_block vb; /* local working space for packet->PCM decode */ | |
62 } ov_struct_t; | |
1828 | 63 #endif |
64 | |
1927 | 65 #ifdef USE_LIBAVCODEC |
2496 | 66 #ifdef USE_LIBAVCODEC_SO |
67 #include <libffmpeg/avcodec.h> | |
68 #else | |
1927 | 69 #include "libavcodec/avcodec.h" |
2496 | 70 #endif |
1927 | 71 static AVCodec *lavc_codec=NULL; |
72 static AVCodecContext lavc_context; | |
73 extern int avcodec_inited; | |
74 #endif | |
75 | |
2415 | 76 |
77 | |
78 #ifdef USE_LIBMAD | |
79 #include <mad.h> | |
80 static struct mad_stream mad_stream; | |
81 static struct mad_frame mad_frame; | |
82 static struct mad_synth mad_synth; | |
83 | |
84 | |
85 // ensure buffer is filled with some data | |
86 static void mad_prepare_buffer(sh_audio_t* sh_audio, struct mad_stream* ms, int length) | |
87 { | |
88 if(sh_audio->a_in_buffer_len < length) { | |
89 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); | |
90 sh_audio->a_in_buffer_len += len; | |
91 } | |
92 } | |
93 | |
94 static void mad_postprocess_buffer(sh_audio_t* sh_audio, struct mad_stream* ms) | |
95 { | |
96 int delta = (unsigned char*)ms->next_frame - (unsigned char *)sh_audio->a_in_buffer; | |
97 if(delta != 0) { | |
98 sh_audio->a_in_buffer_len -= delta; | |
99 memcpy(sh_audio->a_in_buffer, ms->next_frame, sh_audio->a_in_buffer_len); | |
100 } | |
101 } | |
102 | |
103 | |
104 static inline | |
105 signed short mad_scale(mad_fixed_t sample) | |
106 { | |
107 /* round */ | |
108 sample += (1L << (MAD_F_FRACBITS - 16)); | |
109 | |
110 /* clip */ | |
111 if (sample >= MAD_F_ONE) | |
112 sample = MAD_F_ONE - 1; | |
113 else if (sample < -MAD_F_ONE) | |
114 sample = -MAD_F_ONE; | |
115 | |
116 /* quantize */ | |
117 return sample >> (MAD_F_FRACBITS + 1 - 16); | |
118 } | |
119 #endif | |
120 | |
121 | |
122 | |
123 | |
124 | |
1927 | 125 int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen); |
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126 |
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127 |
1289 | 128 static sh_audio_t* dec_audio_sh=NULL; |
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129 |
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130 // AC3 decoder buffer callback: |
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131 static void ac3_fill_buffer(uint8_t **start,uint8_t **end){ |
1289 | 132 int len=ds_get_packet(dec_audio_sh->ds,start); |
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133 //printf("<ac3:%d>\n",len); |
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134 if(len<0) |
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135 *start = *end = NULL; |
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136 else |
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137 *end = *start + len; |
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138 } |
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139 |
1289 | 140 // MP3 decoder buffer callback: |
141 int mplayer_audio_read(char *buf,int size){ | |
142 int len; | |
143 len=demux_read_data(dec_audio_sh->ds,buf,size); | |
144 return len; | |
145 } | |
146 | |
296 | 147 int init_audio(sh_audio_t *sh_audio){ |
303 | 148 int driver=sh_audio->codec->driver; |
296 | 149 |
150 sh_audio->samplesize=2; | |
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151 #if WORDS_BIGENDIAN |
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152 sh_audio->sample_format=AFMT_S16_BE; |
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153 #else |
758 | 154 sh_audio->sample_format=AFMT_S16_LE; |
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155 #endif |
303 | 156 sh_audio->samplerate=0; |
401 | 157 //sh_audio->pcm_bswap=0; |
1529 | 158 sh_audio->o_bps=0; |
303 | 159 |
758 | 160 sh_audio->a_buffer_size=0; |
303 | 161 sh_audio->a_buffer=NULL; |
296 | 162 |
746 | 163 sh_audio->a_in_buffer_len=0; |
164 | |
758 | 165 // setup required min. in/out buffer size: |
166 sh_audio->audio_out_minsize=8192;// default size, maybe not enough for Win32/ACM | |
167 | |
168 switch(driver){ | |
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169 case AFM_ACM: |
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170 #ifndef USE_WIN32DLL |
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171 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoACMSupport); |
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172 driver=0; |
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173 #else |
296 | 174 // Win32 ACM audio codec: |
175 if(init_acm_audio_codec(sh_audio)){ | |
746 | 176 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
296 | 177 sh_audio->channels=sh_audio->o_wf.nChannels; |
178 sh_audio->samplerate=sh_audio->o_wf.nSamplesPerSec; | |
758 | 179 // if(sh_audio->audio_out_minsize>16384) sh_audio->audio_out_minsize=16384; |
180 // sh_audio->a_buffer_size=sh_audio->audio_out_minsize; | |
181 // if(sh_audio->a_buffer_size<sh_audio->audio_out_minsize+MAX_OUTBURST) | |
182 // sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; | |
296 | 183 } else { |
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184 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_ACMiniterror); |
303 | 185 driver=0; |
296 | 186 } |
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187 #endif |
758 | 188 break; |
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189 case AFM_DSHOW: |
296 | 190 #ifndef USE_DIRECTSHOW |
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191 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoDShowAudio); |
303 | 192 driver=0; |
296 | 193 #else |
194 // Win32 DShow audio codec: | |
340 | 195 // printf("DShow_audio: channs=%d rate=%d\n",sh_audio->channels,sh_audio->samplerate); |
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196 if(DS_AudioDecoder_Open(sh_audio->codec->dll,&sh_audio->codec->guid,sh_audio->wf)){ |
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197 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingDLLcodec,sh_audio->codec->dll); |
303 | 198 driver=0; |
199 } else { | |
746 | 200 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; |
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201 sh_audio->channels=sh_audio->wf->nChannels; |
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202 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
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203 sh_audio->audio_in_minsize=2*sh_audio->wf->nBlockAlign; |
296 | 204 if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; |
205 sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; | |
206 sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); | |
207 sh_audio->a_in_buffer_len=0; | |
758 | 208 sh_audio->audio_out_minsize=16384; |
296 | 209 } |
210 #endif | |
758 | 211 break; |
1828 | 212 case AFM_VORBIS: |
213 #ifndef HAVE_OGGVORBIS | |
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214 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoOggVorbis); |
1828 | 215 driver=0; |
216 #else | |
217 /* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */ | |
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218 sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame |
1828 | 219 #endif |
220 break; | |
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221 case AFM_PCM: |
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222 case AFM_DVDPCM: |
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223 case AFM_ALAW: |
758 | 224 // PCM, aLaw |
225 sh_audio->audio_out_minsize=2048; | |
226 break; | |
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227 case AFM_AC3: |
758 | 228 // Dolby AC3 audio: |
229 sh_audio->audio_out_minsize=4*256*6; | |
230 break; | |
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231 case AFM_HWAC3: |
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232 // Dolby AC3 audio: |
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233 sh_audio->audio_out_minsize=4*256*6; |
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234 sh_audio->sample_format = AFMT_AC3; |
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235 break; |
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236 case AFM_GSM: |
758 | 237 // MS-GSM audio codec: |
238 sh_audio->audio_out_minsize=4*320; | |
239 break; | |
2420 | 240 case AFM_IMA4: |
241 // IMA-ADPCM 4:1 audio codec: | |
242 sh_audio->audio_out_minsize=4096; //4*IMA4_SAMPLES_PER_BLOCK; | |
243 sh_audio->ds->ss_div=IMA4_SAMPLES_PER_BLOCK; | |
244 sh_audio->ds->ss_mul=IMA4_BLOCK_SIZE; | |
245 break; | |
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246 case AFM_MPEG: |
758 | 247 // MPEG Audio: |
248 sh_audio->audio_out_minsize=4608; | |
249 break; | |
1927 | 250 case AFM_FFMPEG: |
251 #ifndef USE_LIBAVCODEC | |
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252 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_NoLAVCsupport); |
1927 | 253 return 0; |
254 #else | |
255 // FFmpeg Audio: | |
256 sh_audio->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; | |
257 break; | |
258 #endif | |
2415 | 259 |
260 #ifdef USE_LIBMAD | |
261 case AFM_MAD: | |
262 printf(__FILE__ ":%d:mad: setting minimum outputsize\n", __LINE__); | |
263 sh_audio->audio_out_minsize=4608; | |
264 if(sh_audio->audio_in_minsize<8192) sh_audio->audio_in_minsize=8192; | |
265 sh_audio->a_in_buffer_size=sh_audio->audio_in_minsize; | |
266 sh_audio->a_in_buffer=malloc(sh_audio->a_in_buffer_size); | |
267 sh_audio->a_in_buffer_len=0; | |
268 break; | |
269 #endif | |
296 | 270 } |
271 | |
303 | 272 if(!driver) return 0; |
296 | 273 |
274 // allocate audio out buffer: | |
758 | 275 sh_audio->a_buffer_size=sh_audio->audio_out_minsize+MAX_OUTBURST; // worst case calc. |
276 | |
1567 | 277 mp_msg(MSGT_DECAUDIO,MSGL_V,"dec_audio: Allocating %d + %d = %d bytes for output buffer\n", |
758 | 278 sh_audio->audio_out_minsize,MAX_OUTBURST,sh_audio->a_buffer_size); |
279 | |
296 | 280 sh_audio->a_buffer=malloc(sh_audio->a_buffer_size); |
758 | 281 if(!sh_audio->a_buffer){ |
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282 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_CantAllocAudioBuf); |
758 | 283 return 0; |
284 } | |
296 | 285 memset(sh_audio->a_buffer,0,sh_audio->a_buffer_size); |
286 sh_audio->a_buffer_len=0; | |
287 | |
303 | 288 switch(driver){ |
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289 #ifdef USE_WIN32DLL |
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290 case AFM_ACM: { |
758 | 291 int ret=acm_decode_audio(sh_audio,sh_audio->a_buffer,4096,sh_audio->a_buffer_size); |
296 | 292 if(ret<0){ |
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293 mp_msg(MSGT_DECAUDIO,MSGL_INFO,"ACM decoding error: %d\n",ret); |
303 | 294 driver=0; |
296 | 295 } |
640 | 296 sh_audio->a_buffer_len=ret; |
303 | 297 break; |
296 | 298 } |
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299 #endif |
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300 case AFM_PCM: { |
296 | 301 // AVI PCM Audio: |
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302 WAVEFORMATEX *h=sh_audio->wf; |
746 | 303 sh_audio->i_bps=h->nAvgBytesPerSec; |
296 | 304 sh_audio->channels=h->nChannels; |
305 sh_audio->samplerate=h->nSamplesPerSec; | |
306 sh_audio->samplesize=(h->wBitsPerSample+7)/8; | |
758 | 307 switch(sh_audio->format){ // hardware formats: |
308 case 0x6: sh_audio->sample_format=AFMT_A_LAW;break; | |
309 case 0x7: sh_audio->sample_format=AFMT_MU_LAW;break; | |
310 case 0x11: sh_audio->sample_format=AFMT_IMA_ADPCM;break; | |
311 case 0x50: sh_audio->sample_format=AFMT_MPEG;break; | |
2547 | 312 case 0x736F7774: sh_audio->sample_format=AFMT_S16_LE;sh_audio->codec->driver=AFM_DVDPCM;break; |
758 | 313 // case 0x2000: sh_audio->sample_format=AFMT_AC3; |
314 default: sh_audio->sample_format=(sh_audio->samplesize==2)?AFMT_S16_LE:AFMT_U8; | |
315 } | |
401 | 316 break; |
317 } | |
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318 case AFM_DVDPCM: { |
401 | 319 // DVD PCM Audio: |
320 sh_audio->channels=2; | |
321 sh_audio->samplerate=48000; | |
746 | 322 sh_audio->i_bps=2*2*48000; |
296 | 323 // sh_audio->pcm_bswap=1; |
303 | 324 break; |
325 } | |
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326 case AFM_AC3: { |
296 | 327 // Dolby AC3 audio: |
1289 | 328 dec_audio_sh=sh_audio; // save sh_audio for the callback: |
296 | 329 ac3_config.fill_buffer_callback = ac3_fill_buffer; |
330 ac3_config.num_output_ch = 2; | |
331 ac3_config.flags = 0; | |
332 #ifdef HAVE_MMX | |
333 ac3_config.flags |= AC3_MMX_ENABLE; | |
334 #endif | |
335 #ifdef HAVE_3DNOW | |
336 ac3_config.flags |= AC3_3DNOW_ENABLE; | |
337 #endif | |
338 ac3_init(); | |
339 sh_audio->ac3_frame = ac3_decode_frame(); | |
340 if(sh_audio->ac3_frame){ | |
746 | 341 ac3_frame_t* fr=(ac3_frame_t*)sh_audio->ac3_frame; |
342 sh_audio->samplerate=fr->sampling_rate; | |
296 | 343 sh_audio->channels=2; |
746 | 344 // 1 frame: 6*256 samples 1 sec: sh_audio->samplerate samples |
345 //sh_audio->i_bps=fr->frame_size*fr->sampling_rate/(6*256); | |
346 sh_audio->i_bps=fr->bit_rate*(1000/8); | |
303 | 347 } else { |
348 driver=0; // bad frame -> disable audio | |
349 } | |
350 break; | |
351 } | |
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352 case AFM_HWAC3: { |
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353 unsigned char *buffer; |
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354 struct hwac3info ai; |
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355 int len, skipped; |
1529 | 356 len = ds_get_packet(sh_audio->ds, &buffer); // maybe 1 packet is not enough, |
357 // at least for mpeg, PS packets contain about max. 2000 bytes of data. | |
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358 if(ac3_iec958_parse_syncinfo(buffer, len, &ai, &skipped) < 0) { |
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359 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_AC3notvalid); |
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360 driver = 0; |
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361 break; |
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362 } |
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363 if(ai.samplerate != 48000) { |
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364 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_AC3only48k); |
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365 driver = 0; |
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366 break; |
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367 } |
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368 sh_audio->samplerate=ai.samplerate; |
1529 | 369 sh_audio->samplesize=ai.framesize; |
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370 sh_audio->channels=1; |
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371 sh_audio->i_bps=ai.bitrate*(1000/8); |
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372 sh_audio->ac3_frame=malloc(6144); |
1529 | 373 sh_audio->o_bps=sh_audio->i_bps; // XXX FIXME!!! XXX |
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374 break; |
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375 } |
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376 case AFM_ALAW: { |
296 | 377 // aLaw audio codec: |
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378 sh_audio->channels=sh_audio->wf->nChannels; |
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379 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
746 | 380 sh_audio->i_bps=sh_audio->channels*sh_audio->samplerate; |
303 | 381 break; |
382 } | |
1927 | 383 #ifdef USE_LIBAVCODEC |
384 case AFM_FFMPEG: { | |
385 int x; | |
386 mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); | |
387 if(!avcodec_inited){ | |
388 avcodec_init(); | |
389 avcodec_register_all(); | |
390 avcodec_inited=1; | |
391 } | |
392 lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll); | |
393 if(!lavc_codec){ | |
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394 mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); |
1927 | 395 return 0; |
396 } | |
397 memset(&lavc_context, 0, sizeof(lavc_context)); | |
398 /* open it */ | |
399 if (avcodec_open(&lavc_context, lavc_codec) < 0) { | |
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400 mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); |
1927 | 401 return 0; |
402 } | |
403 mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n"); | |
404 | |
405 // Decode at least 1 byte: (to get header filled) | |
406 x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); | |
407 if(x>0) sh_audio->a_buffer_len=x; | |
408 | |
409 #if 1 | |
410 sh_audio->channels=lavc_context.channels; | |
411 sh_audio->samplerate=lavc_context.sample_rate; | |
412 sh_audio->i_bps=lavc_context.bit_rate/8; | |
413 #else | |
414 sh_audio->channels=sh_audio->wf->nChannels; | |
415 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
416 sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; | |
417 #endif | |
418 break; | |
419 } | |
420 #endif | |
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421 case AFM_GSM: { |
296 | 422 // MS-GSM audio codec: |
423 GSM_Init(); | |
432
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424 sh_audio->channels=sh_audio->wf->nChannels; |
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425 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; |
746 | 426 // decodes 65 byte -> 320 short |
427 // 1 sec: sh_audio->channels*sh_audio->samplerate samples | |
428 // 1 frame: 320 samples | |
429 sh_audio->i_bps=65*(sh_audio->channels*sh_audio->samplerate)/320; // 1:10 | |
303 | 430 break; |
296 | 431 } |
2420 | 432 case AFM_IMA4: { |
433 // IMA-ADPCM 4:1 audio codec: | |
434 sh_audio->channels=sh_audio->wf->nChannels; | |
435 sh_audio->samplerate=sh_audio->wf->nSamplesPerSec; | |
436 // decodes 34 byte -> 64 short | |
437 sh_audio->i_bps=IMA4_BLOCK_SIZE*(sh_audio->channels*sh_audio->samplerate)/IMA4_SAMPLES_PER_BLOCK; // 1:4 | |
438 break; | |
439 } | |
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440 case AFM_MPEG: { |
296 | 441 // MPEG Audio: |
1289 | 442 dec_audio_sh=sh_audio; // save sh_audio for the callback: |
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443 #ifdef USE_FAKE_MONO |
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444 MP3_Init(fakemono); |
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445 #else |
296 | 446 MP3_Init(); |
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447 #endif |
296 | 448 MP3_samplerate=MP3_channels=0; |
449 // printf("[\n"); | |
450 sh_audio->a_buffer_len=MP3_DecodeFrame(sh_audio->a_buffer,-1); | |
451 // printf("]\n"); | |
452 sh_audio->channels=2; // hack | |
453 sh_audio->samplerate=MP3_samplerate; | |
746 | 454 sh_audio->i_bps=MP3_bitrate*(1000/8); |
303 | 455 break; |
456 } | |
1828 | 457 #ifdef HAVE_OGGVORBIS |
458 case AFM_VORBIS: { | |
459 // OggVorbis Audio: | |
460 #if 0 /* just here for reference - atmos */ | |
461 ogg_sync_state oy; /* sync and verify incoming physical bitstream */ | |
462 ogg_stream_state os; /* take physical pages, weld into a logical | |
463 stream of packets */ | |
464 ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */ | |
465 ogg_packet op; /* one raw packet of data for decode */ | |
466 | |
467 vorbis_info vi; /* struct that stores all the static vorbis bitstream | |
468 settings */ | |
469 vorbis_comment vc; /* struct that stores all the bitstream user comments */ | |
470 vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ | |
471 vorbis_block vb; /* local working space for packet->PCM decode */ | |
472 #else | |
473 /* nix, nada, rien, nothing, nem, nüx */ | |
474 #endif | |
475 | |
476 uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header, | |
477 maybe use ogg_uint32_t */ | |
478 //int i; | |
479 int ret; | |
480 char *buffer; | |
481 ogg_packet hdr; | |
482 //ov_struct_t *s=&sh_audio->ov; | |
483 sh_audio->ov=malloc(sizeof(ov_struct_t)); | |
484 //s=&sh_audio->ov; | |
485 | |
486 vorbis_info_init(&sh_audio->ov->vi); | |
487 vorbis_comment_init(&sh_audio->ov->vc); | |
488 | |
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489 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize); |
1828 | 490 memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t)); |
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491 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]); |
1828 | 492 /*for(i=12; i <= 40; i+=2) { // header bruteforce :) |
493 memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t)); | |
494 printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]); | |
495 }*/ | |
496 | |
497 /* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos | |
498 hdr.packet=NULL; | |
499 hdr.b_o_s = 1; /* beginning of stream for first packet */ | |
500 hdr.bytes = hdrsizes[0]; | |
501 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
502 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes); | |
503 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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504 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n"); |
1828 | 505 hdr.b_o_s = 0; |
506 hdr.bytes = hdrsizes[1]; | |
507 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
508 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes); | |
509 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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510 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n"); |
1828 | 511 hdr.bytes = hdrsizes[2]; |
512 hdr.packet = realloc(hdr.packet,hdr.bytes); | |
513 memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes); | |
514 if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0) | |
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515 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n"); |
1828 | 516 hdr.bytes=0; |
517 hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */ | |
518 /* done with the headers */ | |
519 | |
520 | |
521 /* Throw the comments plus a few lines about the bitstream we're | |
522 decoding */ | |
523 { | |
524 char **ptr=sh_audio->ov->vc.user_comments; | |
525 while(*ptr){ | |
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526 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr); |
1828 | 527 ++ptr; |
528 } | |
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529 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'); |
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530 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor); |
1828 | 531 } |
532 sh_audio->channels=sh_audio->ov->vi.channels; | |
533 sh_audio->samplerate=sh_audio->ov->vi.rate; | |
534 sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8; | |
535 | |
536 // printf("[\n"); | |
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537 // sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels; |
1828 | 538 // printf("]\n"); |
539 | |
540 /* OK, got and parsed all three headers. Initialize the Vorbis | |
541 packet->PCM decoder. */ | |
542 vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */ | |
543 vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode | |
544 so multiple block decodes can | |
545 proceed in parallel. We could init | |
546 multiple vorbis_block structures | |
547 for vd here */ | |
548 //printf("OggVorbis: synthesis and block init done.\n"); | |
549 ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */ | |
550 | |
551 while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) { | |
552 if(ret == -1) | |
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553 mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n"); |
1828 | 554 else |
555 if(ret == 0) { | |
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556 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n"); |
1828 | 557 /* submit a a_buffer_len block to libvorbis' Ogg layer */ |
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558 buffer=ogg_sync_buffer(&sh_audio->ov->oy,256); |
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559 ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256)); |
1828 | 560 } |
561 } | |
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562 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n"); |
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563 ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here |
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564 as they'll also become apparent |
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565 at packetout */ |
1828 | 566 |
567 /* Get the serial number and set up the rest of decode. */ | |
568 /* serialno first; use it to set up a logical stream */ | |
569 ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og)); | |
570 | |
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571 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n"); |
1828 | 572 |
573 break; | |
574 } | |
575 #endif | |
2415 | 576 |
577 #ifdef USE_LIBMAD | |
578 case AFM_MAD: | |
579 { | |
580 printf(__FILE__ ":%d:mad: initialising\n", __LINE__); | |
581 mad_frame_init(&mad_frame); | |
582 mad_stream_init(&mad_stream); | |
583 | |
584 printf(__FILE__ ":%d:mad: preparing buffer\n", __LINE__); | |
585 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
586 mad_stream_buffer(&mad_stream, (unsigned char*)(sh_audio->a_in_buffer), sh_audio->a_in_buffer_len); | |
587 mad_stream_sync(&mad_stream); | |
588 mad_synth_init(&mad_synth); | |
589 | |
590 if(mad_frame_decode(&mad_frame, &mad_stream) == 0) | |
591 { | |
592 printf(__FILE__ ":%d:mad: post processing buffer\n", __LINE__); | |
593 mad_postprocess_buffer(sh_audio, &mad_stream); | |
594 } | |
595 else | |
596 { | |
597 printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__); | |
598 } | |
599 | |
2434
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audio channels nr is identified from libmad's channel mode
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|
600 switch (mad_frame.header.mode) |
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|
601 { |
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602 case MAD_MODE_SINGLE_CHANNEL: |
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|
603 sh_audio->channels=1; |
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|
604 break; |
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605 case MAD_MODE_DUAL_CHANNEL: |
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|
606 case MAD_MODE_JOINT_STEREO: |
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|
607 case MAD_MODE_STEREO: |
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|
608 sh_audio->channels=2; |
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|
609 break; |
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|
610 default: |
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|
611 mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "mad: unknown number of channels\n"); |
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audio channels nr is identified from libmad's channel mode
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|
612 } |
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|
613 mp_msg(MSGT_DECAUDIO, MSGL_HINT, "mad: channels: %d (mad channel mode: %d)\n", |
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614 sh_audio->channels, mad_frame.header.mode); |
2432 | 615 /* var. name changed in 0.13.0 (beta) (libmad/CHANGES) -- alex */ |
616 #if (MAD_VERSION_MAJOR >= 0) && (MAD_VERSION_MINOR >= 13) | |
617 sh_audio->samplerate=mad_frame.header.samplerate; | |
618 #else | |
2415 | 619 sh_audio->samplerate=mad_frame.header.sfreq; |
2432 | 620 #endif |
2415 | 621 sh_audio->i_bps=mad_frame.header.bitrate; |
622 printf(__FILE__ ":%d:mad: continuing\n", __LINE__); | |
623 break; | |
624 } | |
625 #endif | |
296 | 626 } |
627 | |
628 if(!sh_audio->channels || !sh_audio->samplerate){ | |
1973
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all error/warn/info messages moved to help_mp-en.h for translation
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1927
diff
changeset
|
629 mp_msg(MSGT_DECAUDIO,MSGL_WARN,MSGTR_UnknownAudio); |
303 | 630 driver=0; |
296 | 631 } |
632 | |
303 | 633 if(!driver){ |
634 if(sh_audio->a_buffer) free(sh_audio->a_buffer); | |
1313
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sh_audio->a_buffer is freed twice (here and in mplayer.c), can result to
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changeset
|
635 sh_audio->a_buffer=NULL; |
303 | 636 return 0; |
637 } | |
296 | 638 |
1529 | 639 if(!sh_audio->o_bps) |
303 | 640 sh_audio->o_bps=sh_audio->channels*sh_audio->samplerate*sh_audio->samplesize; |
641 return driver; | |
296 | 642 } |
291 | 643 |
746 | 644 // Audio decoding: |
291 | 645 |
746 | 646 // Decode a single frame (mp3,acm etc) or 'minlen' bytes (pcm/alaw etc) |
647 // buffer length is 'maxlen' bytes, it shouldn't be exceeded... | |
648 | |
649 int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){ | |
291 | 650 int len=-1; |
303 | 651 switch(sh_audio->codec->driver){ |
1927 | 652 #ifdef USE_LIBAVCODEC |
653 case AFM_FFMPEG: { | |
654 unsigned char *start=NULL; | |
655 int y; | |
656 while(len<minlen){ | |
657 int len2=0; | |
658 int x=ds_get_packet(sh_audio->ds,&start); | |
659 if(x<=0) break; // error | |
2049
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VfW stuff moved to dll_init, warnings fixed, using dll_init.h
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diff
changeset
|
660 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
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1927
diff
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|
661 if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } |
1927 | 662 if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) |
663 if(len2>0){ | |
664 //len=len2;break; | |
665 if(len<0) len=len2; else len+=len2; | |
666 buf+=len2; | |
667 } | |
668 mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); | |
669 } | |
670 } | |
671 break; | |
672 #endif | |
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673 case AFM_MPEG: // MPEG layer 2 or 3 |
291 | 674 len=MP3_DecodeFrame(buf,-1); |
1485
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675 // len=MP3_DecodeFrame(buf,3); |
291 | 676 break; |
1830 | 677 #ifdef HAVE_OGGVORBIS |
1828 | 678 case AFM_VORBIS: { // OggVorbis |
679 /* note: good minlen would be 4k or 8k IMHO - atmos */ | |
680 int ret; | |
681 char *buffer; | |
682 int bytes; | |
683 int samples; | |
684 float **pcm; | |
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685 //ogg_int16_t convbuffer[4096]; |
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686 // int convsize; |
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687 int readlen=1024; |
1834
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688 len=0; |
1837
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689 // convsize=minlen/sh_audio->ov->vi.channels; |
1834
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Now oggvorbis decoding properly works, but a/v sync is bad.
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690 |
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691 while(len < minlen) { /* double loop allows for break in inner loop */ |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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692 while(len < minlen) { /* without aborting the outer loop - atmos */ |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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693 ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op); |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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694 if(ret==0) { |
1837
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695 int xxx=0; |
1834
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696 //printf("OggVorbis: Packetout: need more data, paging!\n"); |
3f8da9408f04
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697 while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) { |
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698 if(ret == -1) |
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699 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n"); |
1834
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Now oggvorbis decoding properly works, but a/v sync is bad.
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700 else |
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701 if(ret == 0) { |
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702 //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.
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703 /* submit a readlen k block to libvorbis' Ogg layer */ |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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704 buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen); |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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705 bytes=demux_read_data(sh_audio->ds,buffer,readlen); |
1837
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706 xxx+=bytes; |
1834
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707 ogg_sync_wrote(&sh_audio->ov->oy,bytes); |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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708 if(bytes==0) |
1837
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709 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n"); |
1834
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Now oggvorbis decoding properly works, but a/v sync is bad.
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710 } |
1828 | 711 } |
1837
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712 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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713 //printf("OggVorbis: Pageout: successfull, pagin in.\n"); |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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714 if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0) |
1837
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715 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n"); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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716 break; |
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Now oggvorbis decoding properly works, but a/v sync is bad.
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717 } else if(ret<0) { |
1837
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parents:
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|
718 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
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diff
changeset
|
719 break; |
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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|
720 } else { |
1828 | 721 |
722 /* we have a packet. Decode it */ | |
723 | |
724 if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */ | |
725 vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->vb); | |
726 | |
727 /* **pcm is a multichannel float vector. In stereo, for | |
728 example, pcm[0] is left, and pcm[1] is right. samples is | |
729 the size of each channel. Convert the float values | |
730 (-1.<=range<=1.) to whatever PCM format and write it out */ | |
731 | |
732 while((samples=vorbis_synthesis_pcmout(&sh_audio->ov->vd,&pcm))>0){ | |
733 int i,j; | |
734 int clipflag=0; | |
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|
735 int convsize=(maxlen-len)/(2*sh_audio->ov->vi.channels); // max size! |
1828 | 736 int bout=(samples<convsize?samples:convsize); |
1837
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|
737 |
9fd8b5e0b864
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|
738 if(bout<=0) break; |
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|
739 |
1828 | 740 /* convert floats to 16 bit signed ints (host order) and |
741 interleave */ | |
742 for(i=0;i<sh_audio->ov->vi.channels;i++){ | |
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|
743 ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); |
1828 | 744 ogg_int16_t *ptr=convbuffer+i; |
745 float *mono=pcm[i]; | |
746 for(j=0;j<bout;j++){ | |
747 #if 1 | |
748 int val=mono[j]*32767.f; | |
749 #else /* optional dither */ | |
750 int val=mono[j]*32767.f+drand48()-0.5f; | |
751 #endif | |
752 /* might as well guard against clipping */ | |
753 if(val>32767){ | |
754 val=32767; | |
755 clipflag=1; | |
756 } | |
757 if(val<-32768){ | |
758 val=-32768; | |
759 clipflag=1; | |
760 } | |
761 *ptr=val; | |
762 ptr+=sh_audio->ov->vi.channels; | |
763 } | |
764 } | |
765 | |
766 if(clipflag) | |
1837
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1834
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changeset
|
767 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence)); |
1828 | 768 |
769 //fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging | |
1837
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diff
changeset
|
770 //memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout); |
1834
3f8da9408f04
Now oggvorbis decoding properly works, but a/v sync is bad.
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changeset
|
771 len+=2*sh_audio->ov->vi.channels*bout; |
1837
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changeset
|
772 |
9fd8b5e0b864
ogg/vorbis stuff cleanup, buffering fixes, printf->mp_msg
arpi
parents:
1834
diff
changeset
|
773 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples); |
9fd8b5e0b864
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1834
diff
changeset
|
774 |
1828 | 775 vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how |
776 many samples we | |
777 actually consumed */ | |
778 } | |
1834
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parents:
1831
diff
changeset
|
779 } // from else, packetout ok |
3f8da9408f04
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diff
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|
780 } // while len |
3f8da9408f04
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781 } // outer while len |
1828 | 782 if(ogg_page_eos(&sh_audio->ov->og)) |
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|
783 mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos |
1828 | 784 |
1837
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|
785 mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len); |
1828 | 786 |
787 break; | |
1831 | 788 } |
1830 | 789 #endif |
1410
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using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
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parents:
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|
790 case AFM_PCM: // AVI PCM |
1058 | 791 len=demux_read_data(sh_audio->ds,buf,minlen); |
401 | 792 break; |
1410
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using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
793 case AFM_DVDPCM: // DVD PCM |
401 | 794 { int j; |
746 | 795 len=demux_read_data(sh_audio->ds,buf,minlen); |
291 | 796 //if(i&1){ printf("Warning! pcm_audio_size&1 !=0 (%d)\n",i);i&=~1; } |
746 | 797 // swap endian: |
291 | 798 for(j=0;j<len;j+=2){ |
799 char x=buf[j]; | |
800 buf[j]=buf[j+1]; | |
801 buf[j+1]=x; | |
802 } | |
803 break; | |
804 } | |
1410
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using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
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diff
changeset
|
805 case AFM_ALAW: // aLaw decoder |
746 | 806 { int l=demux_read_data(sh_audio->ds,buf,minlen/2); |
291 | 807 unsigned short *d=(unsigned short *) buf; |
808 unsigned char *s=buf; | |
809 len=2*l; | |
758 | 810 if(sh_audio->format==6){ |
811 // aLaw | |
883 | 812 while(l>0){ --l; d[l]=alaw2short[s[l]]; } |
758 | 813 } else { |
814 // uLaw | |
883 | 815 while(l>0){ --l; d[l]=ulaw2short[s[l]]; } |
291 | 816 } |
817 break; | |
818 } | |
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changeset
|
819 case AFM_GSM: // MS-GSM decoder |
2420 | 820 { unsigned char ibuf[65]; // 65 bytes / frame |
821 if(demux_read_data(sh_audio->ds,ibuf,65)!=65) break; // EOF | |
822 XA_MSGSM_Decoder(ibuf,(unsigned short *) buf); // decodes 65 byte -> 320 short | |
746 | 823 // XA_GSM_Decoder(buf,(unsigned short *) &sh_audio->a_buffer[sh_audio->a_buffer_len]); // decodes 33 byte -> 160 short |
824 len=2*320; | |
291 | 825 break; |
826 } | |
2420 | 827 case AFM_IMA4: // IMA-ADPCM 4:1 audio codec: |
828 { unsigned char ibuf[IMA4_BLOCK_SIZE]; // bytes / frame | |
829 if(demux_read_data(sh_audio->ds,ibuf,IMA4_BLOCK_SIZE)!=IMA4_BLOCK_SIZE) break; // EOF | |
2424 | 830 len=2*ima4_decode_block((unsigned short*)buf,ibuf,2*IMA4_SAMPLES_PER_BLOCK); |
2420 | 831 break; |
832 } | |
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1408
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|
833 case AFM_AC3: // AC3 decoder |
291 | 834 //printf("{1:%d}",avi_header.idx_pos);fflush(stdout); |
835 if(!sh_audio->ac3_frame) sh_audio->ac3_frame=ac3_decode_frame(); | |
836 //printf("{2:%d}",avi_header.idx_pos);fflush(stdout); | |
837 if(sh_audio->ac3_frame){ | |
296 | 838 len = 256 * 6 *sh_audio->channels*sh_audio->samplesize; |
587
8511095c5283
stage#1 completed: c files no more included from mplayer.c
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diff
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|
839 memcpy(buf,((ac3_frame_t*)sh_audio->ac3_frame)->audio_data,len); |
291 | 840 sh_audio->ac3_frame=NULL; |
841 } | |
842 //printf("{3:%d}",avi_header.idx_pos);fflush(stdout); | |
843 break; | |
1528
a444bd456fcc
ac3/spdif patch by German Gomez Garcia <german@piraos.com>
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parents:
1517
diff
changeset
|
844 case AFM_HWAC3: // AC3 through SPDIF |
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diff
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|
845 if(demux_read_data(sh_audio->ds,sh_audio->ac3_frame, 6144) != 6144) |
a444bd456fcc
ac3/spdif patch by German Gomez Garcia <german@piraos.com>
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|
846 break; //EOF |
a444bd456fcc
ac3/spdif patch by German Gomez Garcia <german@piraos.com>
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diff
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|
847 ac3_iec958_build_burst(1536, 0x1F, 1, buf, sh_audio->ac3_frame); |
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parents:
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diff
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|
848 len = 6144; |
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ac3/spdif patch by German Gomez Garcia <german@piraos.com>
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parents:
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diff
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|
849 break; |
1517
0e9c29538a86
Use USE_WIN32DLL define instead of ARCH_X86 to decide whether or not to compile
jkeil
parents:
1485
diff
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|
850 #ifdef USE_WIN32DLL |
1410
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using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
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|
851 case AFM_ACM: |
758 | 852 // len=sh_audio->audio_out_minsize; // optimal decoded fragment size |
853 // if(len<minlen) len=minlen; else | |
854 // if(len>maxlen) len=maxlen; | |
855 // len=acm_decode_audio(sh_audio,buf,len); | |
856 len=acm_decode_audio(sh_audio,buf,minlen,maxlen); | |
291 | 857 break; |
1309
598e3047ce13
Add some preliminary support for non-x86 architectures to mplayer
jkeil
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1291
diff
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|
858 #endif |
746 | 859 |
291 | 860 #ifdef USE_DIRECTSHOW |
1410
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using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
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|
861 case AFM_DSHOW: // DirectShow |
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1320
diff
changeset
|
862 { int size_in=0; |
291 | 863 int size_out=0; |
864 int srcsize=DS_AudioDecoder_GetSrcSize(maxlen); | |
1567 | 865 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 | 866 if(srcsize>sh_audio->a_in_buffer_size) srcsize=sh_audio->a_in_buffer_size; // !!!!!! |
867 if(sh_audio->a_in_buffer_len<srcsize){ | |
868 sh_audio->a_in_buffer_len+= | |
869 demux_read_data(sh_audio->ds,&sh_audio->a_in_buffer[sh_audio->a_in_buffer_len], | |
870 srcsize-sh_audio->a_in_buffer_len); | |
871 } | |
872 DS_AudioDecoder_Convert(sh_audio->a_in_buffer,sh_audio->a_in_buffer_len, | |
873 buf,maxlen, &size_in,&size_out); | |
1567 | 874 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 | 875 if(size_in>=sh_audio->a_in_buffer_len){ |
876 sh_audio->a_in_buffer_len=0; | |
877 } else { | |
878 sh_audio->a_in_buffer_len-=size_in; | |
879 memcpy(sh_audio->a_in_buffer,&sh_audio->a_in_buffer[size_in],sh_audio->a_in_buffer_len); | |
880 } | |
881 len=size_out; | |
882 break; | |
883 } | |
884 #endif | |
2415 | 885 |
886 #ifdef USE_LIBMAD | |
887 case AFM_MAD: | |
888 { | |
889 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
890 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
891 if(mad_frame_decode(&mad_frame, &mad_stream) == 0) | |
892 { | |
893 mad_synth_frame(&mad_synth, &mad_frame); | |
894 mad_postprocess_buffer(sh_audio, &mad_stream); | |
895 | |
896 /* and fill buffer */ | |
897 | |
898 { | |
899 int i; | |
900 int end_size = mad_synth.pcm.length; | |
901 signed short* samples = (signed short*)buf; | |
902 if(end_size > maxlen/4) | |
903 end_size=maxlen/4; | |
904 | |
905 for(i=0; i<mad_synth.pcm.length; ++i) { | |
906 *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); | |
907 *samples++ = mad_scale(mad_synth.pcm.samples[0][i]); | |
908 // *buf++ = mad_scale(mad_synth.pcm.sampAles[1][i]); | |
909 } | |
910 len = end_size*4; | |
911 } | |
912 } | |
913 else | |
914 { | |
915 printf(__FILE__ ":%d:mad: frame decoding failed\n", __LINE__); | |
916 } | |
917 | |
918 break; | |
919 } | |
920 #endif | |
291 | 921 } |
922 return len; | |
923 } | |
924 | |
1327
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925 void resync_audio_stream(sh_audio_t *sh_audio){ |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
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diff
changeset
|
926 switch(sh_audio->codec->driver){ |
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parents:
1408
diff
changeset
|
927 case AFM_MPEG: |
1327
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some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
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diff
changeset
|
928 MP3_DecodeFrame(NULL,-2); // resync |
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some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
1320
diff
changeset
|
929 MP3_DecodeFrame(NULL,-2); // resync |
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arpi
parents:
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diff
changeset
|
930 MP3_DecodeFrame(NULL,-2); // resync |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
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diff
changeset
|
931 break; |
1849
5f246fcc2971
Resetting stream and sync for seek, no correct test files yet for proper testing but should be ok.
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1837
diff
changeset
|
932 #ifdef HAVE_OGGVORBIS |
5f246fcc2971
Resetting stream and sync for seek, no correct test files yet for proper testing but should be ok.
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parents:
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diff
changeset
|
933 case AFM_VORBIS: |
5f246fcc2971
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diff
changeset
|
934 //printf("OggVorbis: resetting stream.\n"); |
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parents:
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diff
changeset
|
935 ogg_sync_reset(&sh_audio->ov->oy); |
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atmos4
parents:
1837
diff
changeset
|
936 ogg_stream_reset(&sh_audio->ov->os); |
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Resetting stream and sync for seek, no correct test files yet for proper testing but should be ok.
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parents:
1837
diff
changeset
|
937 break; |
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Resetting stream and sync for seek, no correct test files yet for proper testing but should be ok.
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parents:
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diff
changeset
|
938 #endif |
1410
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arpi
parents:
1408
diff
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|
939 case AFM_AC3: |
1327
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arpi
parents:
1320
diff
changeset
|
940 ac3_bitstream_reset(); // reset AC3 bitstream buffer |
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some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
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diff
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|
941 // if(verbose){ printf("Resyncing AC3 audio...");fflush(stdout);} |
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diff
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|
942 sh_audio->ac3_frame=ac3_decode_frame(); // resync |
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parents:
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diff
changeset
|
943 // if(verbose) printf(" OK!\n"); |
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parents:
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diff
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|
944 break; |
1410
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parents:
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diff
changeset
|
945 case AFM_ACM: |
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parents:
1408
diff
changeset
|
946 case AFM_DSHOW: |
1528
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ac3/spdif patch by German Gomez Garcia <german@piraos.com>
arpi
parents:
1517
diff
changeset
|
947 case AFM_HWAC3: |
1327
b12e1817bcc2
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arpi
parents:
1320
diff
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|
948 sh_audio->a_in_buffer_len=0; // reset ACM/DShow audio buffer |
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diff
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|
949 break; |
2415 | 950 |
951 #ifdef USE_LIBMAD | |
952 case AFM_MAD: | |
953 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
954 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
955 mad_stream_sync(&mad_stream); | |
956 mad_postprocess_buffer(sh_audio, &mad_stream); | |
957 break; | |
2418 | 958 #endif |
1327
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parents:
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diff
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|
959 } |
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some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
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diff
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|
960 } |
291 | 961 |
1327
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
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diff
changeset
|
962 void skip_audio_frame(sh_audio_t *sh_audio){ |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
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diff
changeset
|
963 switch(sh_audio->codec->driver){ |
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parents:
1408
diff
changeset
|
964 case AFM_MPEG: MP3_DecodeFrame(NULL,-2);break; // skip MPEG frame |
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
965 case AFM_AC3: sh_audio->ac3_frame=ac3_decode_frame();break; // skip AC3 frame |
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
966 case AFM_ACM: |
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
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parents:
1408
diff
changeset
|
967 case AFM_DSHOW: { |
1408 | 968 int skip=sh_audio->wf->nBlockAlign; |
969 if(skip<16){ | |
970 skip=(sh_audio->wf->nAvgBytesPerSec/16)&(~7); | |
971 if(skip<16) skip=16; | |
972 } | |
973 demux_read_data(sh_audio->ds,NULL,skip); | |
974 break; | |
975 } | |
1410
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
976 case AFM_PCM: |
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
977 case AFM_DVDPCM: |
eda16e490ae7
using AFM_/VFM_ macros instead of hardcoded constants (idea by al3x)
arpi
parents:
1408
diff
changeset
|
978 case AFM_ALAW: { |
1408 | 979 int skip=sh_audio->i_bps/16; |
980 skip=skip&(~3); | |
981 demux_read_data(sh_audio->ds,NULL,skip); | |
982 break; | |
983 } | |
2415 | 984 #ifdef USE_LIBMAD |
985 case AFM_MAD: | |
986 { | |
987 mad_prepare_buffer(sh_audio, &mad_stream, sh_audio->a_in_buffer_size); | |
988 mad_stream_buffer(&mad_stream, sh_audio->a_in_buffer, sh_audio->a_in_buffer_len); | |
989 mad_stream_skip(&mad_stream, 2); | |
990 mad_stream_sync(&mad_stream); | |
991 mad_postprocess_buffer(sh_audio, &mad_stream); | |
992 break; | |
993 } | |
994 #endif | |
995 | |
1327
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
arpi
parents:
1320
diff
changeset
|
996 default: ds_fill_buffer(sh_audio->ds); // skip PCM frame |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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parents:
1320
diff
changeset
|
997 } |
b12e1817bcc2
some cleanup - fixed warnings, removed old stuff, moved audio resync to dec_audio
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diff
changeset
|
998 } |