1 files changed, 14 insertions, 14 deletions

diff --git a/lib/rbcodec/codecs/aac.c b/lib/rbcodec/codecs/aac.c
index 11a84cfa24..15c75708e1 100644
--- a/lib/rbcodec/codecs/aac.c
+++ b/lib/rbcodec/codecs/aac.c
@@ -28,7 +28,7 @@
 CODEC_HEADER
 
 /* The maximum buffer size handled by faad. 12 bytes are required by libfaad
- * as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered 
+ * as headroom (see libfaad/bits.c). FAAD_BYTE_BUFFER_SIZE bytes are buffered
  * for each frame. */
 #define FAAD_BYTE_BUFFER_SIZE (2048-12)
 
@@ -62,7 +62,7 @@ enum codec_status codec_run(void)
     int framelength;
     int lead_trim = 0;
     unsigned int frame_samples;
-    unsigned int i;
+    uint32_t i;
     unsigned char* buffer;
     NeAACDecFrameInfo frame_info;
     NeAACDecHandle decoder;
@@ -129,7 +129,7 @@ enum codec_status codec_run(void)
 #endif
 
     i = 0;
-    
+
     if (param) {
         elapsed_time = param;
         action = CODEC_ACTION_SEEK_TIME;
@@ -138,7 +138,7 @@ enum codec_status codec_run(void)
          * upsampling files the resulting sound_samples_done must be expanded
          * by a factor of 2. This is done via using sbr_fac. */
         if (m4a_seek_raw(&demux_res, &input_stream, file_offset,
-                          &sound_samples_done, (int*) &i, &seek_idx)) {
+                         &sound_samples_done, &i, &seek_idx)) {
             sound_samples_done *= sbr_fac;
         } else {
             sound_samples_done = 0;
@@ -151,8 +151,8 @@ enum codec_status codec_run(void)
     }
 
     ci->set_elapsed(elapsed_time);
-    
-    if (i == 0) 
+
+    if (i == 0)
     {
         lead_trim = ci->id3->lead_trim;
     }
@@ -168,17 +168,17 @@ enum codec_status codec_run(void)
         /* Deal with any pending seek requests */
         if (action == CODEC_ACTION_SEEK_TIME) {
             /* Seek to the desired time position. Important: When seeking in SBR
-             * upsampling files the seek_time must be divided by 2 when calling 
-             * m4a_seek and the resulting sound_samples_done must be expanded 
+             * upsampling files the seek_time must be divided by 2 when calling
+             * m4a_seek and the resulting sound_samples_done must be expanded
              * by a factor 2. This is done via using sbr_fac. */
             if (m4a_seek(&demux_res, &input_stream,
                          (uint64_t) param * ci->id3->frequency / sbr_fac / 1000ULL,
-                          &sound_samples_done, (int*) &i, &seek_idx)) {
+                         &sound_samples_done, &i, &seek_idx)) {
                 sound_samples_done *= sbr_fac;
                 elapsed_time = sound_samples_done * 1000LL / ci->id3->frequency;
                 ci->set_elapsed(elapsed_time);
 
-                if (i == 0) 
+                if (i == 0)
                 {
                     lead_trim = ci->id3->lead_trim;
                 }
@@ -190,9 +190,9 @@ enum codec_status codec_run(void)
         action = CODEC_ACTION_NULL;
 
         /* There can be gaps between chunks, so skip ahead if needed. It
-         * doesn't seem to happen much, but it probably means that a 
+         * doesn't seem to happen much, but it probably means that a
          * "proper" file can have chunks out of order. Why one would want
-         * that an good question (but files with gaps do exist, so who 
+         * that an good question (but files with gaps do exist, so who
          * knows?), so we don't support that - for now, at least.
          */
         file_offset = m4a_check_sample_offset(&demux_res, i, &seek_idx);
@@ -219,7 +219,7 @@ enum codec_status codec_run(void)
 
         /* Output the audio */
         ci->yield();
-        
+
         frame_samples = frame_info.samples >> 1;
 
         if (empty_first_frame)
@@ -238,7 +238,7 @@ enum codec_status codec_run(void)
 
         /* Gather number of samples for the decoded frame. */
         framelength = frame_samples - lead_trim;
-        
+
         if (i == demux_res.num_sample_byte_sizes - 1)
         {
             // Size of the last frame