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, 210 deletions

diff --git a/apps/plugins/codeca52.c b/apps/plugins/codeca52.c
deleted file mode 100644
index ced5ced..0000000
--- a/apps/plugins/codeca52.c
+++ /dev/null
@@ -1,210 +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 <inttypes.h>  /* Needed by a52.h */
-#include <codecs/liba52/config-a52.h>
-#include <codecs/liba52/a52.h>
-
-#include "playback.h"
-#include "lib/codeclib.h"
-
-#define BUFFER_SIZE 4096
-
-struct plugin_api* rb;
-struct codec_api* ci;
-
-static float gain = 1;
-static a52_state_t * state;
-unsigned long samplesdone;
-unsigned long frequency;
-
-/* Two buffers used outside liba52 */
-static uint8_t buf[3840] IDATA_ATTR;
-static int16_t int16_samples[256*2] IDATA_ATTR;
-
-static inline int16_t convert (int32_t i)
-{
-    i >>= 15;
-    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
-}
-
-void output_audio(sample_t* samples,int flags) {
-  int i;
-
-  flags &= A52_CHANNEL_MASK | A52_LFE;
-
-  /* We may need to check the output format in flags - I'm not sure... */
-  for (i = 0; i < 256; i++) {
-    int16_samples[2*i] = convert (samples[i]);
-    int16_samples[2*i+1] = convert (samples[i+256]);
-  }
-
-  rb->yield();
-  while(!ci->audiobuffer_insert((unsigned char*)int16_samples,256*2*2))
-    rb->yield();
-}
-
-
-void a52_decode_data (uint8_t * start, uint8_t * end)
-{
-  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) {
-          DEBUGF("skip\n");
-          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;
-          frequency=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;
-          }
-
-          output_audio(a52_samples (state),flags);
-          samplesdone+=256;
-        }
-        ci->set_elapsed(samplesdone/(frequency/1000));
-        bufptr = buf;
-        bufpos = buf + 7;
-        continue;
-
-        error:
-
-        //logf("Error decoding A52 stream\n");
-        bufptr = buf;
-        bufpos = buf + 7;
-      }
-    }
-  }
-}
-
-#ifndef SIMULATOR
-extern char iramcopy[];
-extern char iramstart[];
-extern char iramend[];
-#endif
-
-/* this is the plugin entry point */
-enum plugin_status plugin_start(struct plugin_api* api, void* parm)
-{
-  size_t n;
-  unsigned char* filebuf;
-
-  /* Generic plugin initialisation */
-  TEST_PLUGIN_API(api);
-
-  rb = api;
-  ci = (struct codec_api*)parm;
-
-#ifndef SIMULATOR
-  rb->memcpy(iramstart, iramcopy, iramend-iramstart);
-#endif
-
-  ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
-  ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
-
-  next_track:
-
-  if (codec_init(api, ci)) {
-  return PLUGIN_ERROR;
-  }
-
-  /* Intialise the A52 decoder and check for success */
-  state = a52_init (0); // Parameter is "accel"
-
-  /* The main decoding loop */
-
-  samplesdone=0;
-  while (1) {
-    if (ci->stop_codec || ci->reload_codec) {
-      break;
-    }
-
-    filebuf=ci->request_buffer(&n,BUFFER_SIZE);
-
-    if (n==0) {      /* End of Stream */
-      break;
-    }
-  
-    a52_decode_data(filebuf,filebuf+n);
-
-    ci->advance_buffer(n);
-  }
-
-  if (ci->request_next_track())
-   goto next_track;
-
-//NOT NEEDED??:  a52_free (state);
-
-  return PLUGIN_OK;
-}