switch to C99 types, part II

Do full C99 integer type conversion in all modules which were not
touched by Eric's merged patch.

3 files changed, 11 insertions, 11 deletions

diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index 550df77df..22d8774a3 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -161,9 +161,9 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		data->audio_format.bits = (mpd_sint8)si->bits_per_sample;
+		data->audio_format.bits = (int8_t)si->bits_per_sample;
 		data->audio_format.sampleRate = si->sample_rate;
-		data->audio_format.channels = (mpd_sint8)si->channels;
+		data->audio_format.channels = (int8_t)si->channels;
 		data->total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index a2fc39881..1dd488fb1 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -50,7 +50,7 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	int bits;
 	struct audio_format audio_format;
 	float total_time;
-	mpd_uint16 bitRate;
+	uint16_t bitRate;
 	struct stat st;
 	int ret, current = 0;
 	char chunk[CHUNK_SIZE];
@@ -69,17 +69,17 @@ static int audiofile_decode(struct decoder * decoder, char *path)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, 16);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	audio_format.bits = (mpd_uint8)bits;
+	audio_format.bits = (uint8_t)bits;
 	audio_format.sampleRate =
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
 	audio_format.channels =
-	              (mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
+	              (uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
 
 	frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
 
 	total_time = ((float)frame_count / (float)audio_format.sampleRate);
 
-	bitRate = (mpd_uint16)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
+	bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
 
 	if (audio_format.bits != 8 && audio_format.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index ec76b035f..f439b68e6 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -66,10 +66,10 @@ static mpc_int32_t mpc_getsize_cb(void *vdata)
 }
 
 /* this _looks_ performance-critical, don't de-inline -- eric */
-static inline mpd_sint16 convertSample(MPC_SAMPLE_FORMAT sample)
+static inline int16_t convertSample(MPC_SAMPLE_FORMAT sample)
 {
 	/* only doing 16-bit audio for now */
-	mpd_sint32 val;
+	int32_t val;
 
 	const int clip_min = -1 << (16 - 1);
 	const int clip_max = (1 << (16 - 1)) - 1;
@@ -114,7 +114,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 	char chunk[MPC_CHUNK_SIZE];
 	int chunkpos = 0;
 	long bitRate = 0;
-	mpd_sint16 *s16 = (mpd_sint16 *) chunk;
+	int16_t *s16 = (int16_t *) chunk;
 	unsigned long samplePos = 0;
 	mpc_uint32_t vbrUpdateAcc;
 	mpc_uint32_t vbrUpdateBits;
@@ -171,7 +171,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 				audio_format.sampleRate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
 				decoder_clear(mpd_decoder);
-				s16 = (mpd_sint16 *) chunk;
+				s16 = (int16_t *) chunk;
 				chunkpos = 0;
 				decoder_command_finished(mpd_decoder);
 			} else
@@ -213,7 +213,7 @@ static int mpc_decode(struct decoder * mpd_decoder, InputStream * inStream)
 					     bitRate, replayGainInfo);
 
 				chunkpos = 0;
-				s16 = (mpd_sint16 *) chunk;
+				s16 = (int16_t *) chunk;
 				if (decoder_get_command(mpd_decoder) == DECODE_COMMAND_STOP) {
 					eof = 1;
 					break;