summaryrefslogtreecommitdiff
path: root/apps/plugins
diff options
context:
space:
mode:
Diffstat (limited to 'apps/plugins')
-rw-r--r--apps/plugins/Makefile2
-rw-r--r--apps/plugins/SOURCES1
-rw-r--r--apps/plugins/a52towav.c455
-rw-r--r--apps/plugins/viewers.config2
4 files changed, 459 insertions, 1 deletions
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 <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 */
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
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-02 13:40:47 +0000