First codeca52 (A52 aka AC3 playback) - it is already faster than realtime, with zero optimisations

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@6668 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 213 insertions, 0 deletions

diff --git a/apps/plugins/codeca52.c b/apps/plugins/codeca52.c
new file mode 100644
index 0000000000..5cb27defc8
--- /dev/null
+++ b/apps/plugins/codeca52.c
@@ -0,0 +1,213 @@
+/***************************************************************************
+ *             __________               __   ___.
+ *   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;
+
+/* A post-processing buffer used outside liba52 */
+static uint8_t buf[3840] 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;
+  static int16_t int16_samples[256*2];
+
+  flags &= A52_CHANNEL_MASK | A52_LFE;
+
+  if (flags==A52_STEREO) {
+    for (i = 0; i < 256; i++) {
+        int16_samples[2*i] = convert (samples[i]);
+        int16_samples[2*i+1] = convert (samples[i+256]);
+    }
+  } else {
+    DEBUGF("ERROR: unsupported format: %d\n",flags);
+  }
+
+  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*16));
+
+  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;
+}