3 files changed, 34 insertions, 34 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
diff --git a/lib/rbcodec/codecs/libm4a/m4a.c b/lib/rbcodec/codecs/libm4a/m4a.c
index 6adc58dab0..295c39c5ff 100644
--- a/lib/rbcodec/codecs/libm4a/m4a.c
+++ b/lib/rbcodec/codecs/libm4a/m4a.c
@@ -114,10 +114,10 @@ void stream_create(stream_t *stream,struct codec_api* ci)
 
 /* Check if there is a dedicated byte position contained for the given frame.
  * Return this byte position in case of success or return -1. This allows to
- * skip empty samples. 
- * During standard playback the search result (index i) will always increase. 
+ * skip empty samples.
+ * During standard playback the search result (index i) will always increase.
  * Therefor we save this index and let the caller set this value again as start
- * index when calling m4a_check_sample_offset() for the next frame. This 
+ * index when calling m4a_check_sample_offset() for the next frame. This
  * reduces the overall loop count significantly. */
 int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *start)
 {
@@ -139,9 +139,9 @@ int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *st
 
 /* Seek to desired sound sample location. Return 1 on success (and modify
  * sound_samples_done and current_sample), 0 if failed. */
-unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream, 
-    uint64_t sound_sample_loc, uint64_t* sound_samples_done, 
-    int* current_sample, int* lookup_table_idx)
+unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
+    uint64_t sound_sample_loc, uint64_t* sound_samples_done,
+    uint32_t* current_sample, uint32_t* lookup_table_idx)
 {
     uint32_t i, sample_i;
     uint32_t time, time_cnt, time_dur;
@@ -257,7 +257,7 @@ unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
  * 1) the lookup_table array contains the file offset for the first sample
  *    of each chunk.
  *
- * 2) the time_to_sample array contains the duration (in sound samples) 
+ * 2) the time_to_sample array contains the duration (in sound samples)
  *    of each sample of data.
  *
  * Locate the chunk containing location (using lookup_table), find the first
@@ -265,8 +265,8 @@ unsigned int m4a_seek(demux_res_t* demux_res, stream_t* stream,
  * calculate the sound_samples_done value.
  */
 unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
-    uint32_t file_loc, uint64_t* sound_samples_done, 
-    int* current_sample, int* lookup_table_idx)
+    uint32_t file_loc, uint64_t* sound_samples_done,
+    uint32_t* current_sample, uint32_t* lookup_table_idx)
 {
     uint32_t i;
     uint32_t chunk_sample     = 0;
@@ -276,7 +276,7 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
     uint32_t tmp_cnt;
     uint32_t new_pos;
 
-    /* We know the desired byte offset, search for the chunk right before. 
+    /* We know the desired byte offset, search for the chunk right before.
      * Return the associated sample to this chunk as chunk_sample. */
     for (i=0; i < demux_res->num_lookup_table; ++i)
     {
@@ -287,7 +287,7 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
     *lookup_table_idx = i;
     chunk_sample = demux_res->lookup_table[i].sample;
     new_pos      = demux_res->lookup_table[i].offset;
-    
+
     /* Get sound sample offset. */
     i = 0;
     time_to_sample_t *tab2 = demux_res->time_to_sample;
@@ -306,12 +306,12 @@ unsigned int m4a_seek_raw(demux_res_t* demux_res, stream_t* stream,
     }
 
     /* Go to the new file position. */
-    if (stream->ci->seek_buffer(new_pos)) 
+    if (stream->ci->seek_buffer(new_pos))
     {
         *sound_samples_done = new_sound_sample;
         *current_sample = chunk_sample;
         return 1;
-    } 
+    }
 
     return 0;
 }
diff --git a/lib/rbcodec/codecs/libm4a/m4a.h b/lib/rbcodec/codecs/libm4a/m4a.h
index bcc32f53bb..7120f8b4c6 100644
--- a/lib/rbcodec/codecs/libm4a/m4a.h
+++ b/lib/rbcodec/codecs/libm4a/m4a.h
@@ -45,13 +45,13 @@ typedef struct {
 
 typedef uint32_t fourcc_t;
 
-typedef struct 
+typedef struct
 {
     uint32_t first_chunk;
     uint32_t num_samples;
 } sample_to_chunk_t;
 
-typedef struct 
+typedef struct
 {
     uint32_t sample_count;
     uint32_t sample_duration;
@@ -73,10 +73,10 @@ typedef struct
 
     int32_t sample_to_chunk_offset;
     uint32_t num_sample_to_chunks;
-    
+
     sample_offset_t *lookup_table;
     uint32_t num_lookup_table;
-    
+
     time_to_sample_t *time_to_sample;
     uint32_t num_time_to_samples;
 
@@ -130,10 +130,10 @@ int stream_eof(stream_t *stream);
 void stream_create(stream_t *stream,struct codec_api* ci);
 unsigned int get_sample_offset(demux_res_t *demux_res, uint32_t sample);
 unsigned int m4a_seek (demux_res_t* demux_res, stream_t* stream,
-    uint64_t sound_sample_loc, uint64_t* sound_samples_done, 
-    int* current_sample, int* lookup_table_idx);
+    uint64_t sound_sample_loc, uint64_t* sound_samples_done,
+    uint32_t* current_sample, uint32_t* lookup_table_idx);
 unsigned int m4a_seek_raw (demux_res_t* demux_res, stream_t* stream,
-    uint32_t file_loc, uint64_t* sound_samples_done, int* current_sample, int* lookup_table_idx);
+    uint32_t file_loc, uint64_t* sound_samples_done, uint32_t* current_sample, uint32_t* lookup_table_idx);
 int m4a_check_sample_offset(demux_res_t *demux_res, uint32_t frame, uint32_t *start);
 
 #endif /* STREAM_H */