explicitly downcast

Tools like "sparse" check for missing downcasts, since implicit cast
may be dangerous.  Although that does not change the compiler result,
it may make the code more readable (IMHO), because you always see when
there may be data cut off.

git-svn-id: https://svn.musicpd.org/mpd/trunk@7196 09075e82-0dd4-0310-85a5-a0d7c8717e4f

5 files changed, 14 insertions, 14 deletions

diff --git a/src/audio.c b/src/audio.c
index 145e6be3..b0c73163 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -197,7 +197,7 @@ int parseAudioConfig(AudioFormat * audioFormat, char *conf)
 		return -1;
 	}
 
-	audioFormat->bits = strtol(test + 1, &test, 10);
+	audioFormat->bits = (mpd_sint8)strtol(test + 1, &test, 10);
 
 	if (*test != ':') {
 		ERROR("error parsing audio output format: %s\n", conf);
@@ -213,7 +213,7 @@ int parseAudioConfig(AudioFormat * audioFormat, char *conf)
 		return -1;
 	}
 
-	audioFormat->channels = strtol(test + 1, &test, 10);
+	audioFormat->channels = (mpd_sint8)strtol(test + 1, &test, 10);
 
 	if (*test != '\0') {
 		ERROR("error parsing audio output format: %s\n", conf);
@@ -428,7 +428,7 @@ void sendMetadataToAudioDevice(MpdTag * tag)
 
 int enableAudioDevice(int fd, int device)
 {
-	if (device < 0 || device >= audioOutputArraySize) {
+	if (device >= audioOutputArraySize) {
 		commandError(fd, ACK_ERROR_ARG, "audio output device id %i "
 			     "doesn't exist\n", device);
 		return -1;
@@ -442,7 +442,7 @@ int enableAudioDevice(int fd, int device)
 
 int disableAudioDevice(int fd, int device)
 {
-	if (device < 0 || device >= audioOutputArraySize) {
+	if (device >= audioOutputArraySize) {
 		commandError(fd, ACK_ERROR_ARG, "audio output device id %i "
 			     "doesn't exist\n", device);
 		return -1;
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 1f90c6ee..3da3e470 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -215,7 +215,7 @@ configure_hw:
 		      snd_strerror(-err));
 		goto fail;
 	}
-	audioFormat->channels = channels;
+	audioFormat->channels = (mpd_sint8)channels;
 
 	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
 					      &sampleRate, NULL);
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
index b2d913d6..1051869f 100644
--- a/src/audioOutputs/audioOutput_oss.c
+++ b/src/audioOutputs/audioOutput_oss.c
@@ -485,9 +485,9 @@ static int oss_openDevice(AudioOutput * audioOutput)
 	OssData *od = audioOutput->data;
 	AudioFormat *audioFormat = &audioOutput->outAudioFormat;
 
-	od->channels = audioFormat->channels;
+	od->channels = (mpd_sint8)audioFormat->channels;
 	od->sampleRate = audioFormat->sampleRate;
-	od->bits = audioFormat->bits;
+	od->bits = (mpd_sint8)audioFormat->bits;
 
 	if ((ret = oss_open(audioOutput)) < 0)
 		return ret;
diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 0a2adf6b..3b351d3a 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -66,7 +66,7 @@ static int flacFindVorbisCommentFloat(const FLAC__StreamMetadata * block,
 					     comments[offset].entry[pos]);
 			tmp = p[len];
 			p[len] = '\0';
-			*fl = atof((char *)p);
+			*fl = (float)atof((char *)p);
 			p[len] = tmp;
 
 			return 1;
@@ -170,9 +170,9 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		dc->audioFormat.bits = si->bits_per_sample;
+		dc->audioFormat.bits = (mpd_sint8)si->bits_per_sample;
 		dc->audioFormat.sampleRate = si->sample_rate;
-		dc->audioFormat.channels = si->channels;
+		dc->audioFormat.channels = (mpd_sint8)si->channels;
 		dc->totalTime = ((float)si->total_samples) / (si->sample_rate);
 		getOutputAudioFormat(&(dc->audioFormat),
 				     &(data->cb->audioFormat));
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 1213d31e..3ca9a14c 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -67,9 +67,9 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, 16);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	dc->audioFormat.bits = bits;
-	dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
-	dc->audioFormat.channels = afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+	dc->audioFormat.bits = (mpd_uint8)bits;
+	dc->audioFormat.sampleRate = (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
+	dc->audioFormat.channels = (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
 	getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
@@ -77,7 +77,7 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
 	dc->totalTime =
 	    ((float)frame_count / (float)dc->audioFormat.sampleRate);
 
-	bitRate = st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5;
+	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc->totalTime / 1000.0 + 0.5);
 
 	if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",