1
0
mirror of https://github.com/mpv-player/mpv synced 2024-12-22 06:42:03 +00:00
mpv/libmpcodecs/ad_liba52.c
arpi 66e6173c0c AltiVec detection code ("borrowed" from FFmpeg and
libmpeg2) & enough code to enable the AltiVec IMDCT
    in liba52 and the DCT64 in mp3lib.
patch by Romain Dolbeau <dolbeau@irisa.fr>


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9004 b3059339-0415-0410-9bf9-f77b7e298cf2
2003-01-18 19:29:46 +00:00

196 lines
5.6 KiB
C

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#ifdef USE_LIBA52
#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;
if(gCpuCaps.hasAltiVec) a52_accel|=MM_ACCEL_PPC_ALTIVEC;
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;
}
#endif