Merge branch 'mk/cleanups'

* mk/cleanups: (60 commits)
  pass constant pointers
  const pointers
  unsigned integers and size_t
  oggflac: fix GCC warnings
  include cleanup
  protect locate.h from double inclusion
  playlist: eliminate unused fd parameters
  jack: made "sample_size" static const
  moved jack configuration to the JackData struct
  jack: removed unused macros
  jack: don't set audioOutput->data=NULL
  jack: initialize JackData in jack_initDriver()
  jack: added freeJackClient()
  jack: initialize jd->client after !jd check
  jack: eliminate superfluous freeJackData() calls
  mp3: converted the MUTEFRAME_ macros to an enum
  mp3: converted the DECODE_ constants to an enum
  wavpack: don't use "isp" before initialization
  wavpack: moved code to wavpack_open_wvc()
  simplified code in the ogg decoder plugin
  ...

1 files changed, 25 insertions, 32 deletions

diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 114a87786..fcebf562b 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -22,11 +22,7 @@
 
 #ifdef HAVE_AUDIOFILE
 
-#include "../utils.h"
-#include "../audio.h"
 #include "../log.h"
-#include "../pcm_utils.h"
-#include "../os_compat.h"
 
 #include <audiofile.h>
 
@@ -51,6 +47,8 @@ static int audiofile_decode(char *path)
 	int bits;
 	mpd_uint16 bitRate;
 	struct stat st;
+	int ret, current = 0;
+	char chunk[CHUNK_SIZE];
 
 	if (stat(path, &st) < 0) {
 		ERROR("failed to stat: %s\n", path);
@@ -88,34 +86,29 @@ static int audiofile_decode(char *path)
 
 	fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
 
-	{
-		int ret, eof = 0, current = 0;
-		char chunk[CHUNK_SIZE];
-
-		while (!eof) {
-			if (dc_seek()) {
-				dc_action_begin();
-				current = dc.seek_where *
-				    dc.audio_format.sampleRate;
-				afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
-				dc_action_end();
-			}
-
-			ret =
-			    afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
-					 CHUNK_SIZE / fs);
-			if (ret <= 0)
-				eof = 1;
-			else {
-				current += ret;
-				ob_send(chunk, ret * fs,
-				        (float)current /
-				        (float)dc.audio_format.sampleRate,
-				        bitRate,
-				        NULL);
-				if (dc_intr())
-					break;
-			}
+	while (1) {
+		if (dc_seek()) {
+			dc_action_begin();
+			current = dc.seek_where *
+			    dc.audio_format.sampleRate;
+			afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
+			dc_action_end();
+		}
+
+		ret =
+		    afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
+				 CHUNK_SIZE / fs);
+		if (ret <= 0)
+			break;
+		else {
+			current += ret;
+			ob_send(chunk, ret * fs,
+				(float)current /
+				(float)dc.audio_format.sampleRate,
+				bitRate,
+				NULL);
+			if (dc_intr())
+				break;
 		}
 	}
 	afCloseFile(af_fp);