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switch to C99 types, part II

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Do full C99 integer type conversion in all modules which were not
touched by Eric's merged patch.
This commit is contained in:
Max Kellermann
2008-09-29 15:49:29 +02:00
parent 0352766dca
commit 226d52b36f
17 changed files with 82 additions and 58 deletions
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4
src/inputPlugins/_flac_common.c
View File

@@ -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:

8
src/inputPlugins/audiofile_plugin.c
View File

@@ -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",

10
src/inputPlugins/mpc_plugin.c
View File

@@ -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;