moved and renamed the codecs, gave the codecs a new extension (.codec),

unified to a single codec-only API, made a new codeclib, disabled the building
of the *2wav plugins


git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6812 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 0 insertions, 520 deletions

diff --git a/apps/plugins/codecmpa.c b/apps/plugins/codecmpa.c
deleted file mode 100644
index 768c3fe..0000000
--- a/apps/plugins/codecmpa.c
+++ /dev/null
@@ -1,520 +0,0 @@
-/***************************************************************************
- *             __________               __   ___.
- *   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"
-
-#include <codecs/libmad/mad.h>
-
-#include "playback.h"
-#include "mp3data.h"
-#include "lib/codeclib.h"
-
-static struct plugin_api* rb;
-
-struct mad_stream  Stream IDATA_ATTR;
-struct mad_frame  Frame IDATA_ATTR;
-struct mad_synth  Synth IDATA_ATTR;
-mad_timer_t      Timer;
-struct dither d0, d1;
-
-/* The following function is used inside libmad - let's hope it's never
-   called. 
-*/
-
-void abort(void) {
-}
-
-/* The "dither" code to convert the 24-bit samples produced by libmad was
-   taken from the coolplayer project - coolplayer.sourceforge.net */
-
-struct dither {
-	mad_fixed_t error[3];
-	mad_fixed_t random;
-};
-
-# define SAMPLE_DEPTH	16
-# define scale(x, y)	dither((x), (y))
-
-/*
- * NAME:		prng()
- * DESCRIPTION:	32-bit pseudo-random number generator
- */
-static __inline
-unsigned long prng(unsigned long state)
-{
-  return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
-}
-
-/*
- * NAME:        dither()
- * DESCRIPTION:	dither and scale sample
- */
-static __inline
-signed int dither(mad_fixed_t sample, struct dither *dither)
-{
-  unsigned int scalebits;
-  mad_fixed_t output, mask, random;
-
-  enum {
-    MIN = -MAD_F_ONE,
-    MAX =  MAD_F_ONE - 1
-  };
-
-  /* noise shape */
-  sample += dither->error[0] - dither->error[1] + dither->error[2];
-
-  dither->error[2] = dither->error[1];
-  dither->error[1] = dither->error[0] / 2;
-
-  /* bias */
-  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
-
-  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
-  mask = (1L << scalebits) - 1;
-
-  /* dither */
-  random  = prng(dither->random);
-  output += (random & mask) - (dither->random & mask);
-
-  //dither->random = random;
-
-  /* clip */
-  if (output > MAX) {
-    output = MAX;
-
-    if (sample > MAX)
-      sample = MAX;
-  }
-  else if (output < MIN) {
-    output = MIN;
-
-    if (sample < MIN)
-      sample = MIN;
-  }
-
-  /* quantize */
-  output &= ~mask;
-
-  /* error feedback */
-  dither->error[0] = sample - output;
-
-  /* scale */
-  return output >> scalebits;
-}
-
-static __inline
-signed int detect_silence(mad_fixed_t sample)
-{
-  unsigned int scalebits;
-  mad_fixed_t output, mask;
-
-  enum {
-    MIN = -MAD_F_ONE,
-    MAX =  MAD_F_ONE - 1
-  };
-
-  /* bias */
-  output = sample + (1L << (MAD_F_FRACBITS + 1 - SAMPLE_DEPTH - 1));
-
-  scalebits = MAD_F_FRACBITS + 1 - SAMPLE_DEPTH;
-  mask = (1L << scalebits) - 1;
-
-  /* clip */
-  if (output > MAX) {
-    output = MAX;
-
-    if (sample > MAX)
-      sample = MAX;
-  }
-  else if (output < MIN) {
-    output = MIN;
-
-    if (sample < MIN)
-      sample = MIN;
-  }
-
-  /* quantize */
-  output &= ~mask;
-
-  /* scale */
-  output >>= scalebits + 4;
-  
-  if (output == 0x00 || output == 0xff)
-    return 1;
-    
-  return 0;
-}
-#define SHRT_MAX 32767
-
-#define INPUT_CHUNK_SIZE        8192
-#define OUTPUT_BUFFER_SIZE	65536 /* Must be an integer multiple of 4. */
-
-unsigned char OutputBuffer[OUTPUT_BUFFER_SIZE];
-unsigned char *OutputPtr;
-unsigned char *GuardPtr=NULL;
-const unsigned char *OutputBufferEnd=OutputBuffer+OUTPUT_BUFFER_SIZE;
-long resampled_data[2][5000]; /* enough to cope with 11khz upsampling */
-
-mad_fixed_t mad_frame_overlap[2][32][18] IDATA_ATTR;
-unsigned char mad_main_data[MAD_BUFFER_MDLEN] IDATA_ATTR;
-/* TODO: what latency does layer 1 have? */
-int mpeg_latency[3] = { 0, 481, 529 };
-#ifdef USE_IRAM
-extern char iramcopy[];
-extern char iramstart[];
-extern char iramend[];
-#endif
-
-#undef DEBUG_GAPLESS
-
-struct resampler {
-    long last_sample, phase, delta;
-};
-
-#if CONFIG_CPU==MCF5249 && !defined(SIMULATOR)
-
-#define INIT() asm volatile ("move.l #0xb0, %macsr") /* frac, round, clip */
-#define FRACMUL(x, y) \
-({ \
-    long t; \
-    asm volatile ("mac.l %[a], %[b], %%acc0\n\t" \
-                  "movclr.l %%acc0, %[t]\n\t" \
-                  : [t] "=r" (t) : [a] "r" (x), [b] "r" (y)); \
-    t; \
-})
-
-#else
-
-#define INIT()
-#define FRACMUL(x, y) (long)(((long long)(x)*(long long)(y)) << 1)
-#endif
-
-/* linear resampling, introduces one sample delay, because of our inability to
-   look into the future at the end of a frame */
-long downsample(long *in, long *out, int num, struct resampler *s)
-{
-    long i = 1, pos;
-    long last = s->last_sample;
-
-    INIT();
-    pos = s->phase >> 16;
-    /* check if we need last sample of previous frame for interpolation */
-    if (pos > 0)
-        last = in[pos - 1];
-    out[0] = last + FRACMUL((s->phase & 0xffff) << 15, in[pos] - last);
-    s->phase += s->delta;
-    while ((pos = s->phase >> 16) < num) {
-        out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
-        s->phase += s->delta;
-    }
-    /* wrap phase accumulator back to start of next frame */
-    s->phase -= num << 16;
-    s->last_sample = in[num - 1];
-    return i;
-}
-
-long upsample(long *in, long *out, int num, struct resampler *s)
-{
-    long i = 0, pos;
-
-    INIT();
-    while ((pos = s->phase >> 16) == 0) {
-        out[i++] = s->last_sample + FRACMUL((s->phase & 0xffff) << 15, in[pos] - s->last_sample);
-        s->phase += s->delta;
-    }
-    while ((pos = s->phase >> 16) < num) {
-        out[i++] = in[pos - 1] + FRACMUL((s->phase & 0xffff) << 15, in[pos] - in[pos - 1]);
-        s->phase += s->delta;
-    }
-    /* wrap phase accumulator back to start of next frame */
-    s->phase -= num << 16;
-    s->last_sample = in[num - 1];
-    return i;
-}
-
-long resample(long *in, long *out, int num, struct resampler *s)
-{
-    if (s->delta >= (1 << 16))
-        return downsample(in, out, num, s);
-    else
-        return upsample(in, out, num, s);
-}
-
-/* this is the plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  struct codec_api *ci = (struct codec_api *)parm;
-  struct mp3info *info;
-  int Status=0;
-  size_t size;
-  int file_end;
-  unsigned short Sample;
-  char *InputBuffer;
-  unsigned int samplecount;
-  unsigned int samplesdone;
-  bool first_frame;
-#ifdef DEBUG_GAPLESS
-  bool first = true;
-  int fd;
-#endif
-  int i;
-  int yieldcounter = 0;
-  int stop_skip, start_skip;
-  struct resampler lr = { 0, 0, 0 }, rr = { 0, 0, 0 };
-  long length;  
-  /* Generic plugin inititialisation */
-
-  TEST_PLUGIN_API(api);
-  rb = api;
-
-#ifdef USE_IRAM
-  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
-#endif
-
-  /* This function sets up the buffers and reads the file into RAM */
-  
-  if (codec_init(api, ci)) {
-    return PLUGIN_ERROR;
-  }
-
-  /* Create a decoder instance */
-
-  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
-  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*16));
-  
-  memset(&Stream, 0, sizeof(struct mad_stream));
-  memset(&Frame, 0, sizeof(struct mad_frame));
-  memset(&Synth, 0, sizeof(struct mad_synth));
-  memset(&Timer, 0, sizeof(mad_timer_t));
-  
-  mad_stream_init(&Stream);
-  mad_frame_init(&Frame);
-  mad_synth_init(&Synth);
-  mad_timer_reset(&Timer);
-
-  /* We do this so libmad doesn't try to call codec_calloc() */
-  memset(mad_frame_overlap, 0, sizeof(mad_frame_overlap));
-  Frame.overlap = &mad_frame_overlap;
-  Stream.main_data = &mad_main_data;
-  /* This label might need to be moved above all the init code, but I don't
-     think reiniting the codec is necessary for MPEG. It might even be unwanted
-     for gapless playback */
-  next_track:
-  
-#ifdef DEBUG_GAPLESS
-  if (first)
-    fd = rb->open("/first.pcm", O_WRONLY | O_CREAT);
-  else
-    fd = rb->open("/second.pcm", O_WRONLY | O_CREAT);
-  first = false;
-#endif
-  
-  info = ci->mp3data;
-  first_frame = false;
-  file_end = 0;
-  OutputPtr = OutputBuffer;
-    
-  while (!*ci->taginfo_ready)
-    rb->yield();
-  
-  ci->request_buffer(&size, ci->id3->first_frame_offset);
-  ci->advance_buffer(size);
- 
-  if (info->enc_delay >= 0 && info->enc_padding >= 0) {
-      stop_skip = info->enc_padding - mpeg_latency[info->layer];
-      if (stop_skip < 0) stop_skip = 0;
-      start_skip = info->enc_delay + mpeg_latency[info->layer];
-  } else {
-      stop_skip = 0;
-      /* We want to skip this amount anyway */
-      start_skip = mpeg_latency[info->layer];
-  }
-  
-  /* NOTE: currently this doesn't work, the below calculated samples_count
-     seems to be right, but sometimes libmad just can't supply us with
-     all the data we need... */
-  if (info->frame_count) {
-      /* TODO: 1152 is the frame size in samples for MPEG1 layer 2 and layer 3,
-         it's probably not correct at all for MPEG2 and layer 1 */
-      samplecount = info->frame_count*1152 - (start_skip + stop_skip);
-      samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
-  } else {
-      samplecount = ci->id3->length * (ci->id3->frequency / 100) / 10;
-      samplesdone = ci->id3->elapsed * (ci->id3->frequency / 100) / 10;
-  }
-  /* rb->snprintf(buf2, sizeof(buf2), "sc: %d", samplecount);
-  rb->splash(0, true, buf2);
-  rb->snprintf(buf2, sizeof(buf2), "length: %d", ci->id3->length);
-  rb->splash(HZ*5, true, buf2);
-  rb->snprintf(buf2, sizeof(buf2), "frequency: %d", ci->id3->frequency);
-  rb->splash(HZ*5, true, buf2); */
-  lr.delta = rr.delta = ci->id3->frequency*65536/44100; 
-  /* This is the decoding loop. */
-  while (1) {
-    rb->yield();
-    if (ci->stop_codec || ci->reload_codec) { 
-        break ;
-    }
-    
-    if (ci->seek_time) {
-        unsigned int sample_loc;
-        int newpos;
-        
-        sample_loc = ci->seek_time/1000 * ci->id3->frequency;
-        newpos = ci->mp3_get_filepos(ci->seek_time-1);
-        if (ci->seek_buffer(newpos)) {
-            if (sample_loc >= samplecount + samplesdone)
-                break ;
-            samplecount += samplesdone - sample_loc;
-            samplesdone = sample_loc;
-        }
-        ci->seek_time = 0;
-    }
-
-    /* Lock buffers */
-    if (Stream.error == 0) {
-        InputBuffer = ci->request_buffer(&size, INPUT_CHUNK_SIZE);
-        if (size == 0 || InputBuffer == NULL)
-            break ;
-        mad_stream_buffer(&Stream, InputBuffer, size);
-    }
-    
-    //if ((int)ci->curpos >= ci->id3->first_frame_offset)
-        //first_frame = true;
-        
-    if(mad_frame_decode(&Frame,&Stream))
-    {
-      if (Stream.error == MAD_FLAG_INCOMPLETE || Stream.error == MAD_ERROR_BUFLEN) {
-          // rb->splash(HZ*1, true, "Incomplete");
-        /* This makes the codec to support partially corrupted files too. */
-        if (file_end == 30)
-            break ;
-        
-        /* Fill the buffer */
-        Stream.error = 0;
-        file_end++;
-        continue ;
-      }
-      else if(MAD_RECOVERABLE(Stream.error))
-      {
-        if(Stream.error!=MAD_ERROR_LOSTSYNC || Stream.this_frame!=GuardPtr)
-        {
-          // rb->splash(HZ*1, true, "Recoverable...!");
-        }
-        continue;
-      }
-      else if(Stream.error==MAD_ERROR_BUFLEN) {
-          //rb->splash(HZ*1, true, "Buflen error");
-          break ;
-      } else {
-          //rb->splash(HZ*1, true, "Unrecoverable error");
-          Status=1;
-          break;
-      }
-    }
-    if (Stream.next_frame)
-        ci->advance_buffer_loc((void *)Stream.next_frame);
-    file_end = false;
-    /* ?? Do we need the timer module? */
-    // mad_timer_add(&Timer,Frame.header.duration);
-
-/* DAVE: This can be used to attenuate the audio */
-//    if(DoFilter)
-//      ApplyFilter(&Frame);   
-
-    mad_synth_frame(&Synth,&Frame);
-    
-    //if (!first_frame) {
-        //samplecount -= Synth.pcm.length;
-        //continue ;
-    //}
-    
-    /* Convert MAD's numbers to an array of 16-bit LE signed integers */
-    /* We skip start_skip number of samples here, this should only happen for
-       very first frame in the stream. */
-    /* TODO: possible for start_skip to exceed one frames worth of samples? */
-    length = resample((long *)&Synth.pcm.samples[0][start_skip], resampled_data[0], Synth.pcm.length, &lr);
-    if (MAD_NCHANNELS(&Frame.header) == 2)
-        resample((long *)&Synth.pcm.samples[1][start_skip], resampled_data[1], Synth.pcm.length, &rr);
-    for (i = 0;i<length;i++)
-    {
-      start_skip = 0; /* not very elegant, and might want to keep this value */
-      samplesdone++;
-      //if (ci->mp3data->padding > 0) {
-      //  ci->mp3data->padding--;
-      //  continue ;
-      //}
-      /*if (!first_frame) {
-        if (detect_silence(Synth.pcm.samples[0][i]))
-            continue ;
-        first_frame = true;
-      }*/
-      
-      /* Left channel */
-      Sample=scale(resampled_data[0][i],&d0);
-      *(OutputPtr++)=Sample>>8;
-      *(OutputPtr++)=Sample&0xff;
-
-      /* Right channel. If the decoded stream is monophonic then
-       * the right output channel is the same as the left one.
-       */
-      if(MAD_NCHANNELS(&Frame.header)==2)
-        Sample=scale(resampled_data[1][i],&d1);
-      *(OutputPtr++)=Sample>>8;
-      *(OutputPtr++)=Sample&0xff;
-      
-      samplecount--;
-      if (samplecount == 0) {
-#ifdef DEBUG_GAPLESS
-        rb->write(fd, OutputBuffer, (int)OutputPtr-(int)OutputBuffer);
-#endif
-        while (!ci->audiobuffer_insert(OutputBuffer, (int)OutputPtr-(int)OutputBuffer))
-            rb->yield();
-        goto song_end;
-      }
-  
-      if (yieldcounter++ == 200) {
-        rb->yield();
-        yieldcounter = 0;
-      }
-      
-      /* Flush the buffer if it is full. */
-      if(OutputPtr==OutputBufferEnd)
-      {
-#ifdef DEBUG_GAPLESS
-        rb->write(fd, OutputBuffer, OUTPUT_BUFFER_SIZE);
-#endif
-        while (!ci->audiobuffer_insert(OutputBuffer, OUTPUT_BUFFER_SIZE))
-            rb->yield();
-        OutputPtr=OutputBuffer;
-      }
-    }
-    ci->set_elapsed(samplesdone / (ci->id3->frequency/1000));
-  }
-  
-  song_end:
-#ifdef DEBUG_GAPLESS
-  rb->close(fd);
-#endif
-  Stream.error = 0;
-  
-  if (ci->request_next_track())
-    goto next_track;
-  return PLUGIN_OK;
-}