Mercurial > mplayer.hg
view libmpcodecs/ad_liba52.c @ 7732:328bbac6224c
Fixes:
- missing check in init
- missing brackets causing failure
- nas_aformat_to_auformat not working properly
- fix hang that was finally reproducible with high disk activity
- don't cut of audio on uninit(), wait for buffer to empty
It also simplifies the event_handler, making it more readable and
implements Sidik Isani's suggestion to make the buffer size dependent on
bytes per second. I've been using it for two days and found no further
problems.
patch by Tobias Diedrich <td@sim.uni-hannover.de>
author | arpi |
---|---|
date | Sun, 13 Oct 2002 22:00:15 +0000 |
parents | 1eadce15446c |
children | b465ba5897a3 |
line wrap: on
line source
#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "cpudetect.h" #include "../liba52/a52.h" #include "../liba52/mm_accel.h" static sample_t * a52_samples; static a52_state_t a52_state; static uint32_t a52_flags=0; #include "bswap.h" static ad_info_t info = { "AC3 decoding with liba52", "liba52", "Nick Kurshev", "Michel LESPINASSE", "" }; LIBAD_EXTERN(liba52) extern int audio_output_channels; int a52_fillbuff(sh_audio_t *sh_audio){ int length=0; int flags=0; int sample_rate=0; int bit_rate=0; sh_audio->a_in_buffer_len=0; /* sync frame:*/ while(1){ while(sh_audio->a_in_buffer_len<7){ int c=demux_getc(sh_audio->ds); if(c<0) return -1; /* EOF*/ sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++]=c; } length = a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); if(length>=7 && length<=3840) break; /* we're done.*/ /* bad file => resync*/ memcpy(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,6); --sh_audio->a_in_buffer_len; } mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"a52: len=%d flags=0x%X %d Hz %d bit/s\n",length,flags,sample_rate,bit_rate); sh_audio->samplerate=sample_rate; sh_audio->i_bps=bit_rate/8; demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+7,length-7); if(crc16_block(sh_audio->a_in_buffer+2,length-2)!=0) mp_msg(MSGT_DECAUDIO,MSGL_STATUS,"a52: CRC check failed! \n"); return length; } /* returns: number of available channels*/ static int a52_printinfo(sh_audio_t *sh_audio){ int flags, sample_rate, bit_rate; char* mode="unknown"; int channels=0; a52_syncinfo (sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate); switch(flags&A52_CHANNEL_MASK){ case A52_CHANNEL: mode="channel"; channels=2; break; case A52_MONO: mode="mono"; channels=1; break; case A52_STEREO: mode="stereo"; channels=2; break; case A52_3F: mode="3f";channels=3;break; case A52_2F1R: mode="2f+1r";channels=3;break; case A52_3F1R: mode="3f+1r";channels=4;break; case A52_2F2R: mode="2f+2r";channels=4;break; case A52_3F2R: mode="3f+2r";channels=5;break; case A52_CHANNEL1: mode="channel1"; channels=2; break; case A52_CHANNEL2: mode="channel2"; channels=2; break; case A52_DOLBY: mode="dolby"; channels=2; break; } mp_msg(MSGT_DECAUDIO,MSGL_INFO,"AC3: %d.%d (%s%s) %d Hz %3.1f kbit/s\n", channels, (flags&A52_LFE)?1:0, mode, (flags&A52_LFE)?"+lfe":"", sample_rate, bit_rate*0.001f); return (flags&A52_LFE) ? (channels+1) : channels; } static int preinit(sh_audio_t *sh) { /* Dolby AC3 audio: */ /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */ sh->audio_out_minsize=audio_output_channels*2*256*6; sh->audio_in_minsize=3840; return 1; } static int init(sh_audio_t *sh_audio) { uint32_t a52_accel=0; sample_t level=1, bias=384; int flags=0; /* Dolby AC3 audio:*/ if(gCpuCaps.hasSSE) a52_accel|=MM_ACCEL_X86_SSE; if(gCpuCaps.hasMMX) a52_accel|=MM_ACCEL_X86_MMX; if(gCpuCaps.hasMMX2) a52_accel|=MM_ACCEL_X86_MMXEXT; if(gCpuCaps.has3DNow) a52_accel|=MM_ACCEL_X86_3DNOW; if(gCpuCaps.has3DNowExt) a52_accel|=MM_ACCEL_X86_3DNOWEXT; a52_samples=a52_init (a52_accel); if (a52_samples == NULL) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 init failed\n"); return 0; } if(a52_fillbuff(sh_audio)<0){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"A52 sync failed\n"); return 0; } /* 'a52 cannot upmix' hotfix:*/ a52_printinfo(sh_audio); sh_audio->channels=audio_output_channels; while(sh_audio->channels>0){ switch(sh_audio->channels){ case 1: a52_flags=A52_MONO; break; /* case 2: a52_flags=A52_STEREO; break;*/ case 2: a52_flags=A52_DOLBY; break; /* case 3: a52_flags=A52_3F; break;*/ case 3: a52_flags=A52_2F1R; break; case 4: a52_flags=A52_2F2R; break; /* 2+2*/ case 5: a52_flags=A52_3F2R; break; case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/ } /* test:*/ flags=a52_flags|A52_ADJUST_LEVEL; mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags); if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: error decoding frame -> nosound\n"); return 0; } mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags); /* frame decoded, let's init resampler:*/ if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break; --sh_audio->channels; /* try to decrease no. of channels*/ } if(sh_audio->channels<=0){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"a52: no resampler. try different channel setup!\n"); return 0; } return 1; } static void uninit(sh_audio_t *sh) { } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { switch(cmd) { case ADCTRL_SKIP_FRAME: a52_fillbuff(sh); break; // skip AC3 frame return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { sample_t level=1, bias=384; int flags=a52_flags|A52_ADJUST_LEVEL; int i,len=-1; if(!sh_audio->a_in_buffer_len) if(a52_fillbuff(sh_audio)<0) return len; /* EOF */ sh_audio->a_in_buffer_len=0; if (a52_frame (&a52_state, sh_audio->a_in_buffer, &flags, &level, bias)){ mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error decoding frame\n"); return len; } len=0; for (i = 0; i < 6; i++) { if (a52_block (&a52_state, a52_samples)){ mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n"); break; } len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]); } return len; }