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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "plugin.h"
#if (CONFIG_HWCODEC == MASNONE)
/* software codec platforms */
#include <inttypes.h> /* Needed by a52.h */
#include <codecs/liba52/config.h>
#include <codecs/liba52/a52.h>
#include "lib/xxx2wav.h" /* Helper functions common to test decoders */
static struct plugin_api* rb;
/* FIX: We can remove this warning when the build system has a
mechanism for auto-detecting the endianness of the target CPU -
WORDS_BIGENDIAN is defined in liba52/config.h and is also used
internally by liba52.
*/
#ifdef WORDS_BIGENDIAN
#warning ************************************* BIG ENDIAN
#define LE_S16(x) ( (uint16_t) ( ((uint16_t)(x) >> 8) | ((uint16_t)(x) << 8) ) )
#else
#define LE_S16(x) (x)
#endif
static float gain = 1;
static a52_state_t * state;
static inline int16_t convert (int32_t i)
{
i >>= 15;
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}
void ao_play(file_info_struct* file_info,sample_t* samples,int flags) {
int i;
static int16_t int16_samples[256*2];
flags &= A52_CHANNEL_MASK | A52_LFE;
if (flags==A52_STEREO) {
for (i = 0; i < 256; i++) {
int16_samples[2*i] = LE_S16(convert (samples[i]));
int16_samples[2*i+1] = LE_S16(convert (samples[i+256]));
}
} else {
#ifdef SIMULATOR
printf("ERROR: unsupported format: %d\n",flags);
#endif
}
/* FIX: Buffer the disk write to write larger amounts at one */
i=rb->write(file_info->outfile,int16_samples,256*2*2);
}
void a52_decode_data (file_info_struct* file_info, uint8_t * start, uint8_t * end)
{
static uint8_t buf[3840];
static uint8_t * bufptr = buf;
static uint8_t * bufpos = buf + 7;
/*
* sample_rate and flags are static because this routine could
* exit between the a52_syncinfo() and the ao_setup(), and we want
* to have the same values when we get back !
*/
static int sample_rate;
static int flags;
int bit_rate;
int len;
while (1) {
len = end - start;
if (!len)
break;
if (len > bufpos - bufptr)
len = bufpos - bufptr;
memcpy (bufptr, start, len);
bufptr += len;
start += len;
if (bufptr == bufpos) {
if (bufpos == buf + 7) {
int length;
length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate);
if (!length) {
#ifdef SIMULATOR
printf("skip\n");
#endif
for (bufptr = buf; bufptr < buf + 6; bufptr++)
bufptr[0] = bufptr[1];
continue;
}
bufpos = buf + length;
} else {
// The following two defaults are taken from audio_out_oss.c:
level_t level;
sample_t bias;
int i;
/* This is the configuration for the downmixing: */
flags=A52_STEREO|A52_ADJUST_LEVEL|A52_LFE;
level=(1 << 26);
bias=0;
level = (level_t) (level * gain);
if (a52_frame (state, buf, &flags, &level, bias))
goto error;
file_info->frames_decoded++;
/* We assume this never changes */
file_info->samplerate=sample_rate;
// An A52 frame consists of 6 blocks of 256 samples
// So we decode and output them one block at a time
for (i = 0; i < 6; i++) {
if (a52_block (state)) {
goto error;
}
ao_play (file_info, a52_samples (state),flags);
file_info->current_sample+=256;
}
bufptr = buf;
bufpos = buf + 7;
continue;
error:
#ifdef SIMULATOR
printf ("error\n");
#endif
bufptr = buf;
bufpos = buf + 7;
}
}
}
}
#define BUFFER_SIZE 4096
/* this is the plugin entry point */
enum plugin_status plugin_start(struct plugin_api* api, void* file)
{
file_info_struct file_info;
/* Generic plugin initialisation */
TEST_PLUGIN_API(api);
rb = api;
/* This function sets up the buffers and reads the file into RAM */
if (local_init(file,"/ac3test.wav",&file_info,api)) {
return PLUGIN_ERROR;
}
/* Intialise the A52 decoder and check for success */
state = a52_init (0); // Parameter is "accel"
if (state == NULL) {
rb->splash(HZ*2, true, "a52_init failed");
return PLUGIN_ERROR;
}
/* The main decoding loop */
file_info.start_tick=*(rb->current_tick);
rb->button_clear_queue();
while (file_info.curpos < file_info.filesize) {
if ((file_info.curpos+BUFFER_SIZE) < file_info.filesize) {
a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.curpos+BUFFER_SIZE]);
file_info.curpos+=BUFFER_SIZE;
} else {
a52_decode_data(&file_info,&filebuf[file_info.curpos],&filebuf[file_info.filesize-1]);
file_info.curpos=file_info.filesize;
}
display_status(&file_info);
if (rb->button_get(false)!=BUTTON_NONE) {
close_wav(&file_info);
return PLUGIN_OK;
}
}
close_wav(&file_info);
/* Cleanly close and exit */
//NOT NEEDED: a52_free (state);
rb->splash(HZ*2, true, "FINISHED!");
return PLUGIN_OK;
}
#endif /* CONFIG_HWCODEC == MASNONE */
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