audiofile_plugin

git-svn-id: https://svn.musicpd.org/mpd/trunk@1248 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 180 insertions, 0 deletions

diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
new file mode 100644
index 000000000..8c1089e1b
--- /dev/null
+++ b/src/inputPlugins/audiofile_plugin.c
@@ -0,0 +1,180 @@
+/* the Music Player Daemon (MPD)
+ * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
+ * This project's homepage is: http://www.musicpd.org
+ * 
+ * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
+ * 
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "../inputPlugin.h"
+
+#ifdef HAVE_AUDIOFILE
+
+#include "../utils.h"
+#include "../audio.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+#include "../playerData.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+#include <audiofile.h>
+
+int getAudiofileTotalTime(char * file)
+{
+	int time;
+	AFfilehandle af_fp = afOpenFile(file, "r", NULL);
+	if(af_fp == AF_NULL_FILEHANDLE) {
+		return -1;
+	}
+	time = (int)
+		((double)afGetFrameCount(af_fp,AF_DEFAULT_TRACK)
+		 /afGetRate(af_fp,AF_DEFAULT_TRACK));
+	afCloseFile(af_fp);
+	return time;
+}
+
+int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) {
+	int fs, frame_count;
+	AFfilehandle af_fp;
+	int bits;
+	mpd_uint16 bitRate;
+	struct stat st;
+
+	if(stat(dc->file,&st) < 0) {
+		ERROR("failed to stat: %s\n",dc->file);
+		return -1;
+	}
+
+	af_fp = afOpenFile(dc->file,"r", NULL);
+	if(af_fp == AF_NULL_FILEHANDLE) {
+		ERROR("failed to open: %s\n",dc->file);
+		return -1;
+	}
+
+	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;
+	
+	if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
+		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
+			dc->file,dc->audioFormat.bits);
+		afCloseFile(af_fp);
+		return -1;
+	}
+	
+	fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK,1);
+
+	dc->state = DECODE_STATE_DECODE;
+	{
+		int ret, eof = 0, current = 0;
+		unsigned char chunk[CHUNK_SIZE];
+
+		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;
+			else {
+				current += ret;
+				sendDataToOutputBuffer(cb, 
+                                        NULL,
+                                        dc,
+                                        1,
+                                        chunk,
+                                        ret*fs,
+					(float)current /
+					(float)dc->audioFormat.sampleRate,
+					bitRate);
+				if(dc->stop) break;
+			}
+		}
+
+		flushOutputBuffer(cb);
+
+		/*if(dc->seek) {
+                        dc->seekError = 1;
+                        dc->seek = 0;
+                }*/
+
+		if(dc->stop) {
+			dc->state = DECODE_STATE_STOP;
+			dc->stop = 0;
+		}
+		else dc->state = DECODE_STATE_STOP;
+	}
+	afCloseFile(af_fp);
+
+	return 0;
+}
+
+MpdTag * audiofileTagDup(char * file) {
+	MpdTag * ret = NULL;
+	int time = getAudiofileTotalTime(file);
+	
+	if (time>=0) {
+		if(!ret) ret = newMpdTag();
+		ret->time = time;
+	}
+
+	return ret;
+}
+
+char * audiofileSuffixes[] = {"wav", NULL};
+
+InputPlugin audiofilePlugin = 
+{
+        "audiofile",
+        NULL,
+        audiofile_decode,
+        audiofileTagDup,
+        INPUT_PLUGIN_STREAM_FILE,
+        audiofileSuffixes,
+        NULL
+};
+
+#else
+
+InputPlugin audiofilePlugin =
+{
+        NULL,
+        NULL,
+        NULL,
+        NULL,
+        0,
+        NULL,
+        NULL
+};
+
+#endif /* HAVE_AUDIOFILE */