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authorAvuton Olrich <avuton@gmail.com>2006-07-20 16:02:40 +0000
committerAvuton Olrich <avuton@gmail.com>2006-07-20 16:02:40 +0000
commit29a25b9933b32800f58dd73d5d1fc21993071c92 (patch)
tree4f456a6f8e44d42acc289c35534ea3e59c0aa96f /src/inputPlugins/audiofile_plugin.c
parent099f0e103f7591eef81183292d704b3a77a99018 (diff)
downloadmpd-29a25b9933b32800f58dd73d5d1fc21993071c92.tar.gz
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Add mpd-indent.sh
Indent the entire tree, hopefully we can keep it indented. git-svn-id: https://svn.musicpd.org/mpd/trunk@4410 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/inputPlugins/audiofile_plugin.c151
1 files changed, 79 insertions, 72 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
index 2284b4f01..19038bbab 100644
--- a/src/inputPlugins/audiofile_plugin.c
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -37,34 +37,35 @@
#include <unistd.h>
#include <audiofile.h>
-static int getAudiofileTotalTime(char * file)
+static int getAudiofileTotalTime(char *file)
{
int time;
AFfilehandle af_fp = afOpenFile(file, "r", NULL);
- if(af_fp == AF_NULL_FILEHANDLE) {
+ if (af_fp == AF_NULL_FILEHANDLE) {
return -1;
}
time = (int)
- ((double)afGetFrameCount(af_fp,AF_DEFAULT_TRACK)
- /afGetRate(af_fp,AF_DEFAULT_TRACK));
+ ((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
+ / afGetRate(af_fp, AF_DEFAULT_TRACK));
afCloseFile(af_fp);
return time;
}
-static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char * path) {
+static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
+{
int fs, frame_count;
AFfilehandle af_fp;
int bits;
mpd_uint16 bitRate;
struct stat st;
- if(stat(path, &st) < 0) {
+ if (stat(path, &st) < 0) {
ERROR("failed to stat: %s\n", path);
return -1;
}
af_fp = afOpenFile(path, "r", NULL);
- if(af_fp == AF_NULL_FILEHANDLE) {
+ if (af_fp == AF_NULL_FILEHANDLE) {
ERROR("failed to open: %s\n", path);
return -1;
}
@@ -72,119 +73,125 @@ static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char * path)
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
dc->audioFormat.bits = bits;
dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
- dc->audioFormat.channels = afGetChannels(af_fp,AF_DEFAULT_TRACK);
- getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
-
- frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK);
-
- dc->totalTime = ((float)frame_count/(float)dc->audioFormat.sampleRate);
-
- bitRate = st.st_size*8.0/dc->totalTime/1000.0+0.5;
-
+ dc->audioFormat.channels = afGetChannels(af_fp, AF_DEFAULT_TRACK);
+ getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+
+ frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
+
+ dc->totalTime =
+ ((float)frame_count / (float)dc->audioFormat.sampleRate);
+
+ bitRate = 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",
- path, dc->audioFormat.bits);
+ path, dc->audioFormat.bits);
afCloseFile(af_fp);
return -1;
}
-
- fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK,1);
+
+ fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
dc->state = DECODE_STATE_DECODE;
{
int ret, eof = 0, current = 0;
char chunk[CHUNK_SIZE];
- while(!eof) {
- if(dc->seek) {
- clearOutputBuffer(cb);
- current = dc->seekWhere *
- dc->audioFormat.sampleRate;
- afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
+ while (!eof) {
+ if (dc->seek) {
+ clearOutputBuffer(cb);
+ current = dc->seekWhere *
+ dc->audioFormat.sampleRate;
+ afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
dc->seek = 0;
}
- ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, CHUNK_SIZE/fs);
- if(ret<=0) eof = 1;
+ ret =
+ afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
+ CHUNK_SIZE / fs);
+ if (ret <= 0)
+ eof = 1;
else {
current += ret;
- sendDataToOutputBuffer(cb,
- NULL,
- dc,
- 1,
- chunk,
- ret*fs,
- (float)current /
- (float)dc->audioFormat.sampleRate,
- bitRate,
- NULL);
- if(dc->stop) break;
+ sendDataToOutputBuffer(cb,
+ NULL,
+ dc,
+ 1,
+ chunk,
+ ret * fs,
+ (float)current /
+ (float)dc->audioFormat.
+ sampleRate, bitRate,
+ NULL);
+ if (dc->stop)
+ break;
}
}
flushOutputBuffer(cb);
/*if(dc->seek) {
- dc->seekError = 1;
- dc->seek = 0;
- }*/
+ dc->seekError = 1;
+ dc->seek = 0;
+ } */
- if(dc->stop) {
+ if (dc->stop) {
dc->state = DECODE_STATE_STOP;
dc->stop = 0;
- }
- else dc->state = DECODE_STATE_STOP;
+ } else
+ dc->state = DECODE_STATE_STOP;
}
afCloseFile(af_fp);
return 0;
}
-static MpdTag * audiofileTagDup(char * file) {
- MpdTag * ret = NULL;
+static MpdTag *audiofileTagDup(char *file)
+{
+ MpdTag *ret = NULL;
int time = getAudiofileTotalTime(file);
-
- if (time>=0) {
- if(!ret) ret = newMpdTag();
+
+ if (time >= 0) {
+ if (!ret)
+ ret = newMpdTag();
ret->time = time;
- }
- else {
- DEBUG("audiofileTagDup: Failed to get total song time from: %s\n", file);
+ } else {
+ DEBUG
+ ("audiofileTagDup: Failed to get total song time from: %s\n",
+ file);
}
return ret;
}
-static char * audiofileSuffixes[] = {"wav", "au", "aiff", "aif", NULL};
+static char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL };
-InputPlugin audiofilePlugin =
-{
- "audiofile",
- NULL,
+InputPlugin audiofilePlugin = {
+ "audiofile",
+ NULL,
NULL,
NULL,
NULL,
- audiofile_decode,
- audiofileTagDup,
- INPUT_PLUGIN_STREAM_FILE,
- audiofileSuffixes,
- NULL
+ audiofile_decode,
+ audiofileTagDup,
+ INPUT_PLUGIN_STREAM_FILE,
+ audiofileSuffixes,
+ NULL
};
#else
-InputPlugin audiofilePlugin =
-{
- NULL,
- NULL,
+InputPlugin audiofilePlugin = {
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL,
NULL,
- NULL,
- NULL,
NULL,
+ 0,
NULL,
- 0,
- NULL,
- NULL
+ NULL
};
-#endif /* HAVE_AUDIOFILE */
+#endif /* HAVE_AUDIOFILE */