Switch to C99 types (retaining compat with old compilers)

Seeing the "mpd_" prefix _everywhere_ is mind-numbing as the
mind needs to retrain itself to skip over the first 4 tokens of
a type to get to its meaning.  So avoid having extra characters
on my terminal to make it easier to follow code at 2:30 am in
the morning.

Please report any new issues you may come across on Free
toolchains.  I realize how difficult it can be to build/maintain
cross-compiling toolchains and I have no intention of forcing
people to upgrade their toolchains to build mpd.

Tested with gcc 2.95.4 and and gcc 4.3.1 on x86-32.

6 files changed, 19 insertions, 19 deletions

diff --git a/src/inputPlugins/_flac_common.c b/src/inputPlugins/_flac_common.c
index beff5d431..546f5c192 100644
--- a/src/inputPlugins/_flac_common.c
+++ b/src/inputPlugins/_flac_common.c
@@ -159,9 +159,9 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 
 	switch (block->type) {
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		dc.audio_format.bits = (mpd_sint8)si->bits_per_sample;
+		dc.audio_format.bits = (int8_t)si->bits_per_sample;
 		dc.audio_format.sampleRate = si->sample_rate;
-		dc.audio_format.channels = (mpd_sint8)si->channels;
+		dc.audio_format.channels = (int8_t)si->channels;
 		dc.total_time = ((float)si->total_samples) / (si->sample_rate);
 		break;
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
index dc97c1e08..2e5df8f48 100644
--- a/src/inputPlugins/aac_plugin.c
+++ b/src/inputPlugins/aac_plugin.c
@@ -313,7 +313,7 @@ static int aac_stream_decode(InputStream *inStream)
 	unsigned int sampleCount;
 	char *sampleBuffer;
 	size_t sampleBufferLen;
-	mpd_uint16 bitRate = 0;
+	uint16_t bitRate = 0;
 	AacBuffer b;
 
 	initAacBuffer(inStream, &b);
@@ -442,7 +442,7 @@ static int aac_decode(char *path)
 	/*float * seekTable;
 	   long seekTableEnd = -1;
 	   int seekPositionFound = 0; */
-	mpd_uint16 bitRate = 0;
+	uint16_t bitRate = 0;
 	AacBuffer b;
 	InputStream inStream;
 
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 6fcc98239..858b71229 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -45,7 +45,7 @@ static int audiofile_decode(char *path)
 	int fs, frame_count;
 	AFfilehandle af_fp;
 	int bits;
-	mpd_uint16 bitRate;
+	uint16_t bitRate;
 	struct stat st;
 	int ret, current = 0;
 	char chunk[CHUNK_SIZE];
@@ -64,18 +64,18 @@ static int audiofile_decode(char *path)
 	afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
 	                         AF_SAMPFMT_TWOSCOMP, 16);
 	afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	dc.audio_format.bits = (mpd_uint8)bits;
+	dc.audio_format.bits = (uint8_t)bits;
 	dc.audio_format.sampleRate =
 	                      (unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
 	dc.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);
 
 	dc.total_time = ((float)frame_count /
 	                 (float)dc.audio_format.sampleRate);
 
-	bitRate = (mpd_uint16)(st.st_size * 8.0 / dc.total_time / 1000.0 + 0.5);
+	bitRate = (uint16_t)(st.st_size * 8.0 / dc.total_time / 1000.0 + 0.5);
 
 	if (dc.audio_format.bits != 8 && dc.audio_format.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
index 1d6333fd6..0d664b020 100644
--- a/src/inputPlugins/mp3_plugin.c
+++ b/src/inputPlugins/mp3_plugin.c
@@ -65,7 +65,7 @@ static unsigned long prng(unsigned long state)
 	return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
 }
 
-static mpd_sint16 audio_linear_dither(unsigned int bits, mad_fixed_t sample,
+static int16_t audio_linear_dither(unsigned int bits, mad_fixed_t sample,
 				      struct audio_dither *dither)
 {
 	unsigned int scalebits;
@@ -107,15 +107,15 @@ static mpd_sint16 audio_linear_dither(unsigned int bits, mad_fixed_t sample,
 
 	dither->error[0] = sample - output;
 
-	return (mpd_sint16)(output >> scalebits);
+	return (int16_t)(output >> scalebits);
 }
 
-static unsigned dither_buffer(mpd_sint16 *dest0, const struct mad_synth *synth,
+static unsigned dither_buffer(int16_t *dest0, const struct mad_synth *synth,
 			      struct audio_dither *dither,
 			      unsigned int start, unsigned int end,
 			      unsigned int num_channels)
 {
-	mpd_sint16 *dest = dest0;
+	int16_t *dest = dest0;
 	unsigned int i;
 
 	for (i = start; i < end; ++i) {
@@ -153,7 +153,7 @@ typedef struct _mp3DecodeData {
 	struct mad_synth synth;
 	mad_timer_t timer;
 	unsigned char readBuffer[READ_BUFFER_SIZE];
-	mpd_sint16 outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
+	int16_t outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
 	float totalTime;
 	float elapsedTime;
 	enum muteframe muteFrame;
diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
index 1e65f6667..0763c6c12 100644
--- a/src/inputPlugins/mp4_plugin.c
+++ b/src/inputPlugins/mp4_plugin.c
@@ -103,7 +103,7 @@ static int mp4_decode(InputStream * inStream)
 	long seekTableEnd = -1;
 	int seekPositionFound = 0;
 	long offset;
-	mpd_uint16 bitRate = 0;
+	uint16_t bitRate = 0;
 	int seeking = 0;
 
 	mp4cb = xmalloc(sizeof(mp4ff_callback_t));
diff --git a/src/inputPlugins/mpc_plugin.c b/src/inputPlugins/mpc_plugin.c
index e5502a2f0..735916cdc 100644
--- a/src/inputPlugins/mpc_plugin.c
+++ b/src/inputPlugins/mpc_plugin.c
@@ -74,10 +74,10 @@ 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;
@@ -121,7 +121,7 @@ static int mpc_decode(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;
@@ -175,7 +175,7 @@ static int mpc_decode(InputStream * inStream)
 			dc_action_begin();
 			samplePos = dc.seek_where * dc.audio_format.sampleRate;
 			if (mpc_decoder_seek_sample(&decoder, samplePos)) {
-				s16 = (mpd_sint16 *) chunk;
+				s16 = (int16_t *) chunk;
 				chunkpos = 0;
 			} else
 				dc.seek_where = DC_SEEK_ERROR;
@@ -214,7 +214,7 @@ static int mpc_decode(InputStream * inStream)
 				        bitRate, replayGainInfo);
 
 				chunkpos = 0;
-				s16 = (mpd_sint16 *) chunk;
+				s16 = (int16_t *) chunk;
 				if (dc_intr()) {
 					eof = 1;
 					break;