Cleaned up code. Now passes full precision samples to the playback engine, and DSP is enabled.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7620 a1c6a512-1295-4272-9138-f99709370657

1 files changed, 122 insertions, 171 deletions

diff --git a/apps/codecs/a52.c b/apps/codecs/a52.c
index 3dacc52..81d8e35 100644
--- a/apps/codecs/a52.c
+++ b/apps/codecs/a52.c
@@ -18,208 +18,159 @@
  ****************************************************************************/
 
 #include "codec.h"
-
+#include "lib/codeclib.h"
 #include <inttypes.h>  /* Needed by a52.h */
 #include <codecs/liba52/config-a52.h>
 #include <codecs/liba52/a52.h>
 
-#include "playback.h"
-#include "dsp.h"
-#include "lib/codeclib.h"
-
 #define BUFFER_SIZE 4096
 
-struct codec_api* rb;
-struct codec_api* ci;
+struct codec_api *ci;
 
-static float gain = 1;
-static a52_state_t * state;
+static a52_state_t *state;
 unsigned long samplesdone;
 unsigned long frequency;
 
-/* Two buffers used outside liba52 */
+/* 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)
+void output_audio(sample_t *samples, int flags)
 {
-    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;
+    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->pcmbuf_insert((unsigned char*)int16_samples,256*2*2))
-    rb->yield();
+    do {
+        ci->yield();
+    } while (!ci->pcmbuf_insert_split(&samples[0], &samples[256], 
+                                      256*sizeof(sample_t)));
 }
 
-
-void a52_decode_data (uint8_t * start, uint8_t * end)
+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;
-      }
+    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 = 1 << 26;
+                sample_t bias = 0;
+                int i;
+
+                /* This is the configuration for the downmixing: */
+                flags = A52_STEREO | A52_ADJUST_LEVEL | A52_LFE;
+
+                if (a52_frame(state, buf, &flags, &level, bias))
+                    goto error;
+
+                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
+#ifdef USE_IRAM
 extern char iramcopy[];
 extern char iramstart[];
 extern char iramend[];
 #endif
 
 /* this is the codec entry point */
-enum codec_status codec_start(struct codec_api* api)
+enum codec_status codec_start(struct codec_api *api)
 {
-  size_t n;
-  unsigned char* filebuf;
-
-  /* Generic codec initialisation */
-  TEST_CODEC_API(api);
-
-  rb = api;
-  ci = (struct codec_api*)api;
-
-#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));
-
-  ci->configure(DSP_DITHER, (bool *)false);
-  ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
-  ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
-  
-  next_track:
-
-  if (codec_init(api)) {
-      return CODEC_ERROR;
-  }
-
-  while (!rb->taginfo_ready)
-      rb->yield();
+    size_t n;
+    unsigned char *filebuf;
+
+    /* Generic codec initialisation */
+    TEST_CODEC_API(api);
+    ci = api;
+
+    #ifdef USE_IRAM 
+    ci->memcpy(iramstart, iramcopy, iramend - iramstart);
+    #endif
+
+    ci->configure(CODEC_DSP_ENABLE, (bool *)true);
+    ci->configure(DSP_DITHER, (bool *)false);
+    ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
+    ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)30);
+    ci->configure(DSP_SET_CLIP_MAX, (long *)((1 << 30) - 1));
+    ci->configure(DSP_SET_CLIP_MIN, (long *)-(1 << 30));
+    ci->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
+    ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*128));
+
+next_track:
+    if (codec_init(api))
+        return CODEC_ERROR;
+
+    while (!ci->taginfo_ready)
+        ci->yield();
     
-  if (rb->id3->frequency != NATIVE_FREQUENCY) {
-      rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
-      rb->configure(CODEC_DSP_ENABLE, (bool *)true);
-  } else {
-      rb->configure(CODEC_DSP_ENABLE, (bool *)false);
-  }
+    ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
     
-  /* Intialise the A52 decoder and check for success */
-  state = a52_init (0); // Parameter is "accel"
-
-  /* The main decoding loop */
+    /* Intialise the A52 decoder and check for success */
+    state = a52_init(0);
 
-  samplesdone=0;
-  while (1) {
-    if (ci->stop_codec || ci->reload_codec) {
-      break;
-    }
+    /* The main decoding loop */
+    samplesdone = 0;
+    while (1) {
+        if (ci->stop_codec || ci->reload_codec)
+            break;
 
-    filebuf=ci->request_buffer(&n,BUFFER_SIZE);
+        filebuf = ci->request_buffer(&n, BUFFER_SIZE);
 
-    if (n==0) {      /* End of Stream */
-      break;
-    }
+        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 CODEC_OK;
+        a52_decode_data(filebuf, filebuf + n);
+        ci->advance_buffer(n);
+    }
+    if (ci->request_next_track())
+        goto next_track;
+    a52_free(state);
+    return CODEC_OK;
 }