Fix a few more warnings from -Wshadow

git-svn-id: https://svn.musicpd.org/mpd/trunk@7300 09075e82-0dd4-0310-85a5-a0d7c8717e4f

3 files changed, 29 insertions, 27 deletions

diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index 5d8eb74f3..2962b57c6 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -273,18 +273,18 @@ static float getAacFloatTotalTime(char *file)
 
 static int getAacTotalTime(char *file)
 {
-	int time = -1;
+	int file_time = -1;
 	float length;
 
 	if ((length = getAacFloatTotalTime(file)) >= 0)
-		time = length + 0.5;
+		file_time = length + 0.5;
 
-	return time;
+	return file_time;
 }
 
 static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 {
-	float time;
+	float file_time;
 	float totalTime;
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
@@ -343,7 +343,7 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 
 	dc->totalTime = totalTime;
 
-	time = 0.0;
+	file_time = 0.0;
 
 	advanceAacBuffer(&b, bread);
 
@@ -388,7 +388,7 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 			bitRate = frameInfo.bytesconsumed * 8.0 *
 			    frameInfo.channels * sampleRate /
 			    frameInfo.samples / 1000 + 0.5;
-			time +=
+			file_time +=
 			    (float)(frameInfo.samples) / frameInfo.channels /
 			    sampleRate;
 		}
@@ -396,7 +396,8 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 		sampleBufferLen = sampleCount * 2;
 
 		sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer,
-				       sampleBufferLen, time, bitRate, NULL);
+				       sampleBufferLen, file_time,
+				       bitRate, NULL);
 		if (dc->seek) {
 			dc->seekError = 1;
 			dc->seek = 0;
@@ -428,14 +429,12 @@ static int aac_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 static MpdTag *aacTagDup(char *file)
 {
 	MpdTag *ret = NULL;
-	int time;
+	int file_time = getAacTotalTime(file);
 
-	time = getAacTotalTime(file);
-
-	if (time >= 0) {
+	if (file_time >= 0) {
 		if ((ret = id3Dup(file)) == NULL)
 			ret = newMpdTag();
-		ret->time = time;
+		ret->time = file_time;
 	} else {
 		DEBUG("aacTagDup: Failed to get total song time from: %s\n",
 		      file);
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 7f7ff9433..1e755b95c 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -90,7 +90,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 	mp4ff_t *mp4fh;
 	mp4ff_callback_t *mp4cb;
 	int32_t track;
-	float time;
+	float file_time;
 	int32_t scale;
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
@@ -163,7 +163,7 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 
 	dc->audioFormat.sampleRate = sampleRate;
 	dc->audioFormat.channels = channels;
-	time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
 	scale = mp4ff_time_scale(mp4fh, track);
 
 	if (mp4Buffer)
@@ -176,11 +176,11 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 		free(mp4cb);
 		return -1;
 	}
-	dc->totalTime = ((float)time) / scale;
+	dc->totalTime = ((float)file_time) / scale;
 
 	numSamples = mp4ff_num_samples(mp4fh, track);
 
-	time = 0.0;
+	file_time = 0.0;
 
 	seekTable = xmalloc(sizeof(float) * numSamples);
 
@@ -194,14 +194,14 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 			while (seekTable[i] < dc->seekWhere)
 				i++;
 			sampleId = i - 1;
-			time = seekTable[sampleId];
+			file_time = seekTable[sampleId];
 		}
 
 		dur = mp4ff_get_sample_duration(mp4fh, track, sampleId);
 		offset = mp4ff_get_sample_offset(mp4fh, track, sampleId);
 
 		if (sampleId > seekTableEnd) {
-			seekTable[sampleId] = time;
+			seekTable[sampleId] = file_time;
 			seekTableEnd = sampleId;
 		}
 
@@ -211,9 +211,9 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 			dur = 0;
 		else
 			dur -= offset;
-		time += ((float)dur) / scale;
+		file_time += ((float)dur) / scale;
 
-		if (seeking && time > dc->seekWhere)
+		if (seeking && file_time > dc->seekWhere)
 			seekPositionFound = 1;
 
 		if (seeking && seekPositionFound) {
@@ -279,7 +279,8 @@ static int mp4_decode(OutputBuffer * cb, DecoderControl * dc,
 		sampleBuffer += offset * channels * 2;
 
 		sendDataToOutputBuffer(cb, inStream, dc, 1, sampleBuffer,
-				       sampleBufferLen, time, bitRate, NULL);
+				       sampleBufferLen, file_time,
+				       bitRate, NULL);
 		if (dc->stop) {
 			eof = 1;
 			break;
@@ -311,7 +312,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound)
 	mp4ff_t *mp4fh;
 	mp4ff_callback_t *cb;
 	int32_t track;
-	int32_t time;
+	int32_t file_time;
 	int32_t scale;
 	int i;
 
@@ -343,7 +344,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound)
 	}
 
 	ret = newMpdTag();
-	time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
+	file_time = mp4ff_get_track_duration_use_offsets(mp4fh, track);
 	scale = mp4ff_time_scale(mp4fh, track);
 	if (scale < 0) {
 		mp4ff_close(mp4fh);
@@ -352,7 +353,7 @@ static MpdTag *mp4DataDup(char *file, int *mp4MetadataFound)
 		freeMpdTag(ret);
 		return NULL;
 	}
-	ret->time = ((float)time) / scale + 0.5;
+	ret->time = ((float)file_time) / scale + 0.5;
 
 	for (i = 0; i < mp4ff_meta_get_num_items(mp4fh); i++) {
 		char *item;
diff --git a/src/inputPlugins/wavpack_plugin.c b/src/inputPlugins/wavpack_plugin.c
index 527e60a56..41753c62f 100644
--- a/src/inputPlugins/wavpack_plugin.c
+++ b/src/inputPlugins/wavpack_plugin.c
@@ -133,7 +133,7 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
 {
 	void (*format_samples)(int Bps, void *buffer, uint32_t samcnt);
 	char chunk[CHUNK_SIZE];
-	float time;
+	float file_time;
 	int samplesreq, samplesgot;
 	int allsamples;
 	int position, outsamplesize;
@@ -204,14 +204,16 @@ static void wavpack_decode(OutputBuffer *cb, DecoderControl *dc,
 			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
 			              1000 + 0.5);
 			position += samplesgot;
-			time = (float)position / dc->audioFormat.sampleRate;
+			file_time = (float)position /
+			            dc->audioFormat.sampleRate;
 
 			format_samples(Bps, chunk,
 			               samplesgot * dc->audioFormat.channels);
 
 			sendDataToOutputBuffer(cb, NULL, dc, 0, chunk,
 			                       samplesgot * outsamplesize,
-			                       time, bitrate, replayGainInfo);
+			                       file_time, bitrate,
+					       replayGainInfo);
 		}
 	} while (samplesgot == samplesreq);