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mpv/libmpcodecs/ad_liba52.c

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C
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#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <math.h>
#include <assert.h>
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "mpbswap.h"
#include "ad_internal.h"
#include "cpudetect.h"
#include "libaf/af_format.h"
#include "liba52/a52.h"
#include "liba52/mm_accel.h"
static a52_state_t *a52_state;
static uint32_t a52_flags=0;
/** Used by a52_resample_float, it defines the mapping between liba52
* channels and output channels. The ith nibble from the right in the
* hex representation of channel_map is the index of the source
* channel corresponding to the ith output channel. Source channels are
* indexed 1-6. Silent output channels are marked by 0xf. */
static uint32_t channel_map;
#define DRC_NO_ACTION 0
#define DRC_NO_COMPRESSION 1
#define DRC_CALLBACK 2
/** The output is multiplied by this var. Used for volume control */
static sample_t a52_level = 1;
/** The value of the -a52drc switch. */
float a52_drc_level = 1.0;
static int a52_drc_action = DRC_NO_ACTION;
#include "mpbswap.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<8){
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;
}
if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
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*/
if(sh_audio->format!=0x2000) swab(sh_audio->a_in_buffer,sh_audio->a_in_buffer,8);
memmove(sh_audio->a_in_buffer,sh_audio->a_in_buffer+1,7);
--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;
sh_audio->samplesize=sh_audio->sample_format==AF_FORMAT_FLOAT_NE ? 4 : 2;
demux_read_data(sh_audio->ds,sh_audio->a_in_buffer+8,length-8);
if(sh_audio->format!=0x2000)
swab(sh_audio->a_in_buffer+8,sh_audio->a_in_buffer+8,length-8);
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_V,"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;
}
sample_t dynrng_call (sample_t c, void *data) {
// fprintf(stderr, "(%lf, %lf): %lf\n", (double)c, (double)a52_drc_level, (double)pow((double)c, a52_drc_level));
return pow((double)c, a52_drc_level);
}
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 */
if (sh->samplesize < 2) sh->samplesize = 2;
sh->audio_out_minsize=audio_output_channels*sh->samplesize*256*6;
sh->audio_in_minsize=3840;
a52_level = 1.0;
return 1;
}
/**
* \brief Function to convert the "planar" float format used by liba52
* into the interleaved float format used by libaf/libao2.
* \param in the input buffer containing the planar samples.
* \param out the output buffer where the interleaved result is stored.
*/
static int a52_resample_float(float *in, int16_t *out)
{
unsigned long i;
float *p = (float*) out;
for (i = 0; i != 256; i++) {
unsigned long map = channel_map;
do {
unsigned long ch = map & 15;
if (ch == 15)
*p = 0;
else
*p = in[i + ((ch-1)<<8)];
p++;
} while ((map >>= 4));
}
return (int16_t*) p - out;
}
static int init(sh_audio_t *sh_audio)
{
uint32_t a52_accel=0;
sample_t level=a52_level, 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_state=a52_init (a52_accel);
if (a52_state == 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;
}
/* Init a52 dynrng */
if (a52_drc_level < 0.001) {
/* level == 0 --> no compression, init library without callback */
a52_drc_action = DRC_NO_COMPRESSION;
} else if (a52_drc_level > 0.999) {
/* level == 1 --> full compression, do nothing at all (library default = full compression) */
a52_drc_action = DRC_NO_ACTION;
} else {
a52_drc_action = DRC_CALLBACK;
}
/* Library init for dynrng has to be done for each frame, see decode_audio() */
/* '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:*/
channel_map = 0;
if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
if (!(flags & A52_LFE)) {
switch ((flags<<3) | sh_audio->channels) {
case (A52_MONO << 3) | 1: channel_map = 0x1; break;
case (A52_CHANNEL << 3) | 2:
case (A52_STEREO << 3) | 2:
case (A52_DOLBY << 3) | 2: channel_map = 0x21; break;
case (A52_2F1R << 3) | 3: channel_map = 0x321; break;
case (A52_2F2R << 3) | 4: channel_map = 0x4321; break;
case (A52_3F << 3) | 5: channel_map = 0x2ff31; break;
case (A52_3F2R << 3) | 5: channel_map = 0x25431; break;
}
} else if (sh_audio->channels == 6) {
switch (flags & ~A52_LFE) {
case A52_MONO : channel_map = 0x12ffff; break;
case A52_CHANNEL:
case A52_STEREO :
case A52_DOLBY : channel_map = 0x1fff32; break;
case A52_3F : channel_map = 0x13ff42; break;
case A52_2F1R : channel_map = 0x1f4432; break;
case A52_2F2R : channel_map = 0x1f5432; break;
case A52_3F2R : channel_map = 0x136542; break;
}
}
if (channel_map) {
a52_resample = a52_resample_float;
break;
}
} else
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_RESYNC_STREAM:
case ADCTRL_SKIP_FRAME:
a52_fillbuff(sh);
return CONTROL_TRUE;
case ADCTRL_SET_VOLUME: {
float vol = *(float*)arg;
if (vol > 60.0) vol = 60.0;
a52_level = vol <= -200.0 ? 0 : pow(10.0,vol/20.0);
return CONTROL_TRUE;
}
case ADCTRL_QUERY_FORMAT:
if (*(int*)arg == AF_FORMAT_S16_NE ||
*(int*)arg == AF_FORMAT_FLOAT_NE)
return CONTROL_TRUE;
return CONTROL_FALSE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
sample_t level=a52_level, bias=384;
int flags=a52_flags|A52_ADJUST_LEVEL;
int i,len=-1;
if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
bias = 0;
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;
}
/* handle dynrng */
if (a52_drc_action != DRC_NO_ACTION) {
if (a52_drc_action == DRC_NO_COMPRESSION)
a52_dynrng(a52_state, NULL, NULL);
else
a52_dynrng(a52_state, dynrng_call, NULL);
}
len=0;
for (i = 0; i < 6; i++) {
if (a52_block (a52_state)){
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"a52: error at resampling\n");
break;
}
len+=2*a52_resample(a52_samples(a52_state),(int16_t *)&buf[len]);
}
assert(len <= maxlen);
return len;
}
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