From 7b96e2daa65af18310cc998de053c5188c32cbe1 Mon Sep 17 00:00:00 2001 From: Dave Chapman Date: Wed, 16 Feb 2005 12:56:00 +0000 Subject: 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 --- apps/plugins/Makefile | 2 +- apps/plugins/SOURCES | 1 + apps/plugins/a52towav.c | 455 ++++++++++++++++++++++++++++++++++++++++++++ apps/plugins/viewers.config | 2 + 4 files changed, 459 insertions(+), 1 deletion(-) create mode 100644 apps/plugins/a52towav.c (limited to 'apps') diff --git a/apps/plugins/Makefile b/apps/plugins/Makefile index 12126a0d3d..8b7243b127 100644 --- a/apps/plugins/Makefile +++ b/apps/plugins/Makefile @@ -17,7 +17,7 @@ ifdef APPEXTRA endif ifdef SOFTWARECODECS - CODECLIBS = -lmad + CODECLIBS = -lmad -la52 endif LDS := plugin.lds diff --git a/apps/plugins/SOURCES b/apps/plugins/SOURCES index 9dd042ff31..caa77f080f 100644 --- a/apps/plugins/SOURCES +++ b/apps/plugins/SOURCES @@ -65,4 +65,5 @@ alpine_cdc.c #if CONFIG_HWCODEC == MASNONE /* software codec platforms */ mpa2wav.c +a52towav.c #endif diff --git a/apps/plugins/a52towav.c b/apps/plugins/a52towav.c new file mode 100644 index 0000000000..17b6c91e51 --- /dev/null +++ b/apps/plugins/a52towav.c @@ -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 /* Needed by a52.h */ + +#include +#include + +/* 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;i0) { *(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 */ diff --git a/apps/plugins/viewers.config b/apps/plugins/viewers.config index bf2488fea4..bb5752c9db 100644 --- a/apps/plugins/viewers.config +++ b/apps/plugins/viewers.config @@ -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 -- cgit v1.2.3