Initial version of a52towav test viewer plugin for liba52 - output is hardcoded to /ac3test.wav. CUrrently restricted to Stereo AC-3 files, but easy to fix for other types of files (e.g. 5.1)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@5977 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Dave Chapman
parent
fd58842b29
commit
7b96e2daa6
4 changed files with 459 additions and 1 deletions
|
|
@ -17,7 +17,7 @@ ifdef APPEXTRA
|
|||
endif
|
||||
|
||||
ifdef SOFTWARECODECS
|
||||
CODECLIBS = -lmad
|
||||
CODECLIBS = -lmad -la52
|
||||
endif
|
||||
|
||||
LDS := plugin.lds
|
||||
|
|
|
|||
|
|
@ -65,4 +65,5 @@ alpine_cdc.c
|
|||
|
||||
#if CONFIG_HWCODEC == MASNONE /* software codec platforms */
|
||||
mpa2wav.c
|
||||
a52towav.c
|
||||
#endif
|
||||
|
|
|
|||
455
apps/plugins/a52towav.c
Normal file
455
apps/plugins/a52towav.c
Normal file
|
|
@ -0,0 +1,455 @@
|
|||
/***************************************************************************
|
||||
* __________ __ ___.
|
||||
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
|
||||
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
|
||||
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
|
||||
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
|
||||
* \/ \/ \/ \/ \/
|
||||
* $Id$
|
||||
*
|
||||
* Copyright (C) 2002 Björn Stenberg
|
||||
*
|
||||
* 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) && !defined(SIMULATOR)
|
||||
/* software codec platforms, not for simulator */
|
||||
|
||||
#include <inttypes.h> /* Needed by a52.h */
|
||||
|
||||
#include <codecs/liba52/config.h>
|
||||
#include <codecs/liba52/a52.h>
|
||||
|
||||
/* Currently used for WAV output */
|
||||
#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
|
||||
|
||||
typedef struct ao_sample_format {
|
||||
int bits; /* bits per sample */
|
||||
int rate; /* samples per second (in a single channel) */
|
||||
int channels; /* number of audio channels */
|
||||
int byte_format; /* Byte ordering in sample, see constants below */
|
||||
} ao_sample_format;
|
||||
|
||||
#define AO_FMT_LITTLE 1
|
||||
#define AO_FMT_BIG 2
|
||||
#define AO_FMT_NATIVE 4
|
||||
|
||||
/* the main data structure of the program */
|
||||
typedef struct {
|
||||
int infile;
|
||||
int outfile;
|
||||
off_t curpos;
|
||||
off_t filesize;
|
||||
ao_sample_format samfmt; /* bits, rate, channels, byte_format */
|
||||
// ao_device *ao_dev;
|
||||
unsigned long total_samples;
|
||||
unsigned long current_sample;
|
||||
float total_time; /* seconds */
|
||||
float elapsed_time; /* seconds */
|
||||
} file_info_struct;
|
||||
|
||||
file_info_struct file_info;
|
||||
|
||||
#define MALLOC_BUFSIZE (512*1024)
|
||||
|
||||
int mem_ptr;
|
||||
int bufsize;
|
||||
unsigned char* mp3buf; // The actual MP3 buffer from Rockbox
|
||||
unsigned char* mallocbuf; // 512K from the start of MP3 buffer
|
||||
unsigned char* filebuf; // The rest of the MP3 buffer
|
||||
|
||||
|
||||
|
||||
#define BUFFER_SIZE 4096
|
||||
//static uint8_t buffer[BUFFER_SIZE];
|
||||
static float gain = 1;
|
||||
static a52_state_t * state;
|
||||
|
||||
int output;
|
||||
|
||||
// DAVE: I'm not sure what these are for.
|
||||
int disable_accel=0;
|
||||
int disable_adjust=0;
|
||||
int disable_dynrng=0;
|
||||
|
||||
/* welcome to the example rockbox plugin */
|
||||
|
||||
/* here is a global api struct pointer. while not strictly necessary,
|
||||
it's nice not to have to pass the api pointer in all function calls
|
||||
in the plugin */
|
||||
static struct plugin_api* rb;
|
||||
|
||||
void* malloc(size_t size) {
|
||||
void* x;
|
||||
char s[32];
|
||||
|
||||
x=&mallocbuf[mem_ptr];
|
||||
mem_ptr+=size+(size%4); // Keep memory 32-bit aligned (if it was already?)
|
||||
|
||||
rb->snprintf(s,30,"Memory used: %d\r",mem_ptr);
|
||||
rb->lcd_putsxy(0,80,s);
|
||||
rb->lcd_update();
|
||||
return(x);
|
||||
}
|
||||
|
||||
void* calloc(size_t nmemb, size_t size) {
|
||||
void* x;
|
||||
x=malloc(nmemb*size);
|
||||
rb->memset(x,0,nmemb*size);
|
||||
return(x);
|
||||
}
|
||||
|
||||
void free(void* ptr) {
|
||||
(void)ptr;
|
||||
}
|
||||
|
||||
void* realloc(void* ptr, size_t size) {
|
||||
void* x;
|
||||
(void)ptr;
|
||||
x=malloc(size);
|
||||
return(x);
|
||||
}
|
||||
|
||||
void *memcpy(void *dest, const void *src, size_t n) {
|
||||
return(rb->memcpy(dest,src,n));
|
||||
}
|
||||
|
||||
void *memset(void *s, int c, size_t n) {
|
||||
return(rb->memset(s,c,n));
|
||||
}
|
||||
|
||||
int memcmp(const void *s1, const void *s2, size_t n) {
|
||||
return(rb->memcmp(s1,s2,n));
|
||||
}
|
||||
|
||||
void* memmove(const void *s1, const void *s2, size_t n) {
|
||||
char* dest=(char*)s1;
|
||||
char* src=(char*)s2;
|
||||
size_t i;
|
||||
|
||||
for (i=0;i<n;i++) { dest[i]=src[i]; }
|
||||
// while(n>0) { *(dest++)=*(src++); n--; }
|
||||
return(dest);
|
||||
}
|
||||
|
||||
void qsort(void *base, size_t nmemb, size_t size, int(*compar)(const void *, const void *)) {
|
||||
rb->qsort(base,nmemb,size,compar);
|
||||
}
|
||||
|
||||
|
||||
|
||||
|
||||
static unsigned char wav_header[44]={'R','I','F','F', // 0 - ChunkID
|
||||
0,0,0,0, // 4 - ChunkSize (filesize-8)
|
||||
'W','A','V','E', // 8 - Format
|
||||
'f','m','t',' ', // 12 - SubChunkID
|
||||
16,0,0,0, // 16 - SubChunk1ID // 16 for PCM
|
||||
1,0, // 20 - AudioFormat (1=16-bit)
|
||||
2,0, // 22 - NumChannels
|
||||
0,0,0,0, // 24 - SampleRate in Hz
|
||||
0,0,0,0, // 28 - Byte Rate (SampleRate*NumChannels*(BitsPerSample/8)
|
||||
4,0, // 32 - BlockAlign (== NumChannels * BitsPerSample/8)
|
||||
16,0, // 34 - BitsPerSample
|
||||
'd','a','t','a', // 36 - Subchunk2ID
|
||||
0,0,0,0 // 40 - Subchunk2Size
|
||||
};
|
||||
|
||||
void close_wav(file_info_struct* file_info) {
|
||||
int x;
|
||||
int filesize=rb->filesize(file_info->outfile);
|
||||
|
||||
/* We assume 16-bit, Stereo */
|
||||
|
||||
rb->lseek(file_info->outfile,0,SEEK_SET);
|
||||
|
||||
// ChunkSize
|
||||
x=filesize-8;
|
||||
wav_header[4]=(x&0xff);
|
||||
wav_header[5]=(x&0xff00)>>8;
|
||||
wav_header[6]=(x&0xff0000)>>16;
|
||||
wav_header[7]=(x&0xff000000)>>24;
|
||||
|
||||
// Samplerate
|
||||
wav_header[24]=file_info->samfmt.rate&0xff;
|
||||
wav_header[25]=(file_info->samfmt.rate&0xff00)>>8;
|
||||
wav_header[26]=(file_info->samfmt.rate&0xff0000)>>16;
|
||||
wav_header[27]=(file_info->samfmt.rate&0xff000000)>>24;
|
||||
|
||||
// ByteRate
|
||||
x=file_info->samfmt.rate*4;
|
||||
wav_header[28]=(x&0xff);
|
||||
wav_header[29]=(x&0xff00)>>8;
|
||||
wav_header[30]=(x&0xff0000)>>16;
|
||||
wav_header[31]=(x&0xff000000)>>24;
|
||||
|
||||
// Subchunk2Size
|
||||
x=filesize-44;
|
||||
wav_header[40]=(x&0xff);
|
||||
wav_header[41]=(x&0xff00)>>8;
|
||||
wav_header[42]=(x&0xff0000)>>16;
|
||||
wav_header[43]=(x&0xff000000)>>24;
|
||||
|
||||
rb->write(file_info->outfile,wav_header,sizeof(wav_header));
|
||||
rb->close(file_info->outfile);
|
||||
}
|
||||
|
||||
static inline int16_t convert (int32_t i)
|
||||
{
|
||||
i >>= 15;
|
||||
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
|
||||
}
|
||||
|
||||
void convert2s16_2 (sample_t * _f, int16_t * s16)
|
||||
{
|
||||
int i;
|
||||
int32_t * f = (int32_t *) _f;
|
||||
for (i = 0; i < 256; i++) {
|
||||
s16[2*i] = LE_S16(convert (f[i]));
|
||||
s16[2*i+1] = LE_S16(convert (f[i+256]));
|
||||
}
|
||||
}
|
||||
|
||||
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) {
|
||||
// convert2s16_2(samples,int16_samples,flags);
|
||||
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
|
||||
fprintf(stderr,"ERROR: unsupported format: %d\n",flags);
|
||||
#endif
|
||||
}
|
||||
|
||||
i=rb->write(file_info->outfile,int16_samples,256*2*2);
|
||||
|
||||
#ifdef SIMULATOR
|
||||
if (i!=(256*2*2)) {
|
||||
fprintf(stderr,"Attempted to write %d bytes, wrote %d bytes\n",256*2*2,i);
|
||||
}
|
||||
#endif
|
||||
}
|
||||
|
||||
|
||||
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
|
||||
fprintf (stderr, "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;
|
||||
|
||||
if (output==0) {
|
||||
file_info->samfmt.bits=16;
|
||||
file_info->samfmt.rate=sample_rate;
|
||||
output=1;
|
||||
// output=ao_open(&format);
|
||||
}
|
||||
|
||||
// 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;
|
||||
// print_fps (0);
|
||||
continue;
|
||||
error:
|
||||
#ifdef SIMULATOR
|
||||
fprintf (stderr, "error\n");
|
||||
#endif
|
||||
bufptr = buf;
|
||||
bufpos = buf + 7;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* this is the plugin entry point */
|
||||
enum plugin_status plugin_start(struct plugin_api* api, void* file)
|
||||
{
|
||||
int i,n,bytesleft;
|
||||
char s[32];
|
||||
unsigned long ticks_taken;
|
||||
unsigned long start_tick;
|
||||
unsigned long long speed;
|
||||
unsigned long xspeed;
|
||||
int accel=0; // ??? This is the parameter to a52_init().
|
||||
|
||||
/* this macro should be called as the first thing you do in the plugin.
|
||||
it test that the api version and model the plugin was compiled for
|
||||
matches the machine it is running on */
|
||||
TEST_PLUGIN_API(api);
|
||||
|
||||
/* if you are using a global api pointer, don't forget to copy it!
|
||||
otherwise you will get lovely "I04: IllInstr" errors... :-) */
|
||||
rb = api;
|
||||
|
||||
/* now go ahead and have fun! */
|
||||
// rb->splash(HZ*2, true, "Hello world!");
|
||||
|
||||
mem_ptr=0;
|
||||
mp3buf=rb->plugin_get_mp3_buffer(&bufsize);
|
||||
mallocbuf=mp3buf;
|
||||
filebuf=&mp3buf[MALLOC_BUFSIZE];
|
||||
|
||||
rb->snprintf(s,32,"mp3 bufsize: %d\r",bufsize);
|
||||
rb->lcd_putsxy(0,100,s);
|
||||
rb->lcd_update();
|
||||
|
||||
file_info.infile=rb->open(file,O_RDONLY);
|
||||
file_info.outfile=rb->creat("/ac3test.wav",O_WRONLY);
|
||||
rb->write(file_info.outfile,wav_header,sizeof(wav_header));
|
||||
file_info.curpos=0;
|
||||
file_info.filesize=rb->filesize(file_info.infile);
|
||||
|
||||
if (file_info.filesize > (bufsize-MALLOC_BUFSIZE)) {
|
||||
rb->close(file_info.infile);
|
||||
rb->splash(HZ*2, true, "File too large");
|
||||
return PLUGIN_ERROR;
|
||||
}
|
||||
|
||||
rb->snprintf(s,32,"Loading file...");
|
||||
rb->lcd_putsxy(0,0,s);
|
||||
rb->lcd_update();
|
||||
|
||||
bytesleft=file_info.filesize;
|
||||
i=0;
|
||||
while (bytesleft > 0) {
|
||||
n=rb->read(file_info.infile,&filebuf[i],bytesleft);
|
||||
if (n < 0) {
|
||||
rb->close(file_info.infile);
|
||||
rb->splash(HZ*2, true, "ERROR READING FILE");
|
||||
return PLUGIN_ERROR;
|
||||
}
|
||||
i+=n; bytesleft-=n;
|
||||
}
|
||||
rb->close(file_info.infile);
|
||||
|
||||
state = a52_init (accel);
|
||||
if (state == NULL) {
|
||||
//fprintf (stderr, "A52 init failed\n");
|
||||
return PLUGIN_ERROR;
|
||||
}
|
||||
|
||||
i=0;
|
||||
start_tick=*(rb->current_tick);
|
||||
while (file_info.curpos < file_info.filesize) {
|
||||
i++;
|
||||
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;
|
||||
}
|
||||
|
||||
rb->snprintf(s,32,"Bytes read: %d\r",file_info.curpos);
|
||||
rb->lcd_putsxy(0,0,s);
|
||||
rb->snprintf(s,32,"Samples Decoded: %d\r",file_info.current_sample);
|
||||
rb->lcd_putsxy(0,20,s);
|
||||
rb->snprintf(s,32,"Frames Decoded: %d\r",i);
|
||||
rb->lcd_putsxy(0,40,s);
|
||||
|
||||
ticks_taken=*(rb->current_tick)-start_tick;
|
||||
|
||||
/* e.g.:
|
||||
ticks_taken=500
|
||||
sam_fmt.rate=44,100
|
||||
samples_decoded=172,400
|
||||
(samples_decoded/sam_fmt.rate)*100=400 (time it should have taken)
|
||||
% Speed=(400/500)*100=80%
|
||||
|
||||
*/
|
||||
|
||||
if (ticks_taken==0) { ticks_taken=1; } // Avoid fp exception.
|
||||
|
||||
speed=(100*file_info.current_sample)/file_info.samfmt.rate;
|
||||
xspeed=(speed*10000)/ticks_taken;
|
||||
rb->snprintf(s,32,"Speed %ld.%02ld %% Secs: %d",(xspeed/100),(xspeed%100),ticks_taken/100);
|
||||
rb->lcd_putsxy(0,60,s);
|
||||
|
||||
rb->lcd_update();
|
||||
if (rb->button_get(false)!=BUTTON_NONE) {
|
||||
close_wav(&file_info);
|
||||
return PLUGIN_OK;
|
||||
}
|
||||
}
|
||||
close_wav(&file_info);
|
||||
|
||||
//NO NEED: a52_free (state);
|
||||
rb->splash(HZ*2, true, "FINISHED!");
|
||||
return PLUGIN_OK;
|
||||
}
|
||||
#endif /* CONFIG_HWCODEC == MASNONE */
|
||||
|
|
@ -8,3 +8,5 @@ m3u,search.rock,00 00 00 00 00 00
|
|||
txt,sort.rock, 00 00 00 00 00 00
|
||||
mp2,mpa2wav.rock, 00 00 00 00 00 00
|
||||
mp3,mpa2wav.rock, 00 00 00 00 00 00
|
||||
ac3,a52towav.rock, 00 00 00 00 00 00
|
||||
a52,a52towav.rock, 00 00 00 00 00 00
|
||||
|
|
|
|||
Loading…
Reference in a new issue