mp3 and ogg plugin stuff

git-svn-id: https://svn.musicpd.org/mpd/trunk@1245 09075e82-0dd4-0310-85a5-a0d7c8717e4f

2 files changed, 1023 insertions, 0 deletions

diff --git a/src/inputPlugins/mp3_plugin.c b/src/inputPlugins/mp3_plugin.c
new file mode 100644
index 000000000..5d34a1068
--- /dev/null
+++ b/src/inputPlugins/mp3_plugin.c
@@ -0,0 +1,659 @@
+/* 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
+ *
+ * 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_MAD
+
+#include "../pcm_utils.h"
+#ifdef USE_MPD_MAD
+#include "../libmad/mad.h"
+#else
+#include <mad.h>
+#endif
+#ifdef HAVE_ID3TAG
+#ifdef USE_MPD_ID3TAG
+#include "../libid3tag/id3tag.h"
+#else
+#include <id3tag.h>
+#endif
+#endif
+#include "../log.h"
+#include "../utils.h"
+#include "../tag.h"
+#include "../path.h"
+
+#include <stdio.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <unistd.h>
+#include <errno.h>
+
+#define FRAMES_CUSHION		2000
+
+#define READ_BUFFER_SIZE	40960
+
+#define DECODE_SKIP		-3
+#define DECODE_BREAK		-2
+#define DECODE_CONT		-1
+#define DECODE_OK		0
+
+#define MUTEFRAME_SKIP          1
+#define MUTEFRAME_SEEK          2
+
+/* this is stolen from mpg321! */
+struct audio_dither {
+	mad_fixed_t error[3];
+	mad_fixed_t random;
+};
+
+unsigned long prng(unsigned long state) {
+	return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
+}
+
+signed long audio_linear_dither(unsigned int bits, mad_fixed_t sample, struct audio_dither *dither) {
+	unsigned int scalebits;
+	mad_fixed_t output, mask, random;
+
+	enum {
+		MIN = -MAD_F_ONE,
+		MAX =  MAD_F_ONE - 1
+	};
+
+	sample += dither->error[0] - dither->error[1] + dither->error[2];
+
+	dither->error[2] = dither->error[1];
+	dither->error[1] = dither->error[0] / 2;
+
+	output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
+
+	scalebits = MAD_F_FRACBITS + 1 - bits;
+	mask = (1L << scalebits) - 1;
+
+	random  = prng(dither->random);
+	output += (random & mask) - (dither->random & mask);
+
+	dither->random = random;
+
+	if (output > MAX) {
+		output = MAX;
+
+		if (sample > MAX)
+			sample = MAX;
+	}
+	else if (output < MIN) {
+	        output = MIN;
+
+		if (sample < MIN)
+			sample = MIN;
+	}
+
+	output &= ~mask;
+
+	dither->error[0] = sample - output;
+
+	return output >> scalebits;
+}
+/* end of stolen stuff from mpg321 */
+
+/* decoder stuff is based on madlld */
+
+#define MP3_DATA_OUTPUT_BUFFER_SIZE 4096
+
+typedef struct _mp3DecodeData {
+	struct mad_stream stream;
+	struct mad_frame frame;
+	struct mad_synth synth;
+	mad_timer_t timer;
+	unsigned char readBuffer[READ_BUFFER_SIZE];
+	char outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
+	char * outputPtr;
+	char * outputBufferEnd;
+	float totalTime;
+	float elapsedTime;
+	int muteFrame;
+	long * frameOffset;
+	mad_timer_t * times;
+	long highestFrame;
+	long maxFrames;
+	long currentFrame;
+	int flush;
+	unsigned long bitRate;
+	InputStream * inStream;
+} mp3DecodeData;
+
+void initMp3DecodeData(mp3DecodeData * data, InputStream * inStream) {
+	data->outputPtr = data->outputBuffer;
+	data->outputBufferEnd = data->outputBuffer+MP3_DATA_OUTPUT_BUFFER_SIZE;
+	data->muteFrame = 0;
+	data->highestFrame = 0;
+	data->maxFrames = 0;
+	data->frameOffset = NULL;
+	data->times = NULL;
+	data->currentFrame = 0;
+	data->flush = 1;
+        data->inStream = inStream;
+
+	mad_stream_init(&data->stream);
+	mad_frame_init(&data->frame);
+	mad_synth_init(&data->synth);
+	mad_timer_reset(&data->timer);
+}
+
+int seekMp3InputBuffer(mp3DecodeData * data, long offset) {
+	if(seekInputStream(data->inStream,offset,SEEK_SET) < 0) {
+                return -1;
+        }
+
+	mad_stream_buffer(&data->stream,data->readBuffer,0);
+	(data->stream).error = 0;
+
+	return 0;
+}
+
+int fillMp3InputBuffer(mp3DecodeData * data) {
+	size_t readSize;
+	size_t remaining;
+        size_t readed;
+	unsigned char * readStart;
+
+	if((data->stream).next_frame!=NULL) {
+		remaining = (data->stream).bufend-(data->stream).next_frame;
+		memmove(data->readBuffer,(data->stream).next_frame,remaining);
+		readStart = (data->readBuffer)+remaining;
+		readSize = READ_BUFFER_SIZE-remaining;
+	}
+	else {
+		readSize = READ_BUFFER_SIZE;
+		readStart = data->readBuffer,
+		remaining = 0;
+	}
+
+	readed = readFromInputStream(data->inStream, readStart, (size_t)1, 
+			readSize);
+	if(readed <= 0 && inputStreamAtEOF(data->inStream)) return -1;
+	/* sleep for a fraction of a second! */
+	else if(readed <= 0) my_usleep(10000);
+
+	mad_stream_buffer(&data->stream,data->readBuffer,readed+remaining);
+	(data->stream).error = 0;
+
+	return 0;
+}
+
+int decodeNextFrameHeader(mp3DecodeData * data) {
+	if((data->stream).buffer==NULL || (data->stream).error==MAD_ERROR_BUFLEN) {
+		if(fillMp3InputBuffer(data) < 0) {
+			return DECODE_BREAK;
+		}
+	}
+	if(mad_header_decode(&data->frame.header,&data->stream)) {
+#ifdef HAVE_ID3TAG
+		if((data->stream).error==MAD_ERROR_LOSTSYNC && 
+				(data->stream).this_frame) 
+		{
+			signed long tagsize = id3_tag_query(
+					(data->stream).this_frame,
+					(data->stream).bufend-
+					(data->stream).this_frame);
+			if(tagsize>0) {
+				mad_stream_skip(&(data->stream),tagsize);
+				return DECODE_CONT;
+			}
+		}
+#endif
+		if(MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		}
+		else {
+			if((data->stream).error==MAD_ERROR_BUFLEN) return DECODE_CONT;
+			else
+			{
+				ERROR("unrecoverable frame level error "
+					"(%s).\n",
+					mad_stream_errorstr(&data->stream));
+				data->flush = 0;
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	return DECODE_OK;
+}
+
+int decodeNextFrame(mp3DecodeData * data) {
+	if((data->stream).buffer==NULL || (data->stream).error==MAD_ERROR_BUFLEN) {
+		if(fillMp3InputBuffer(data) < 0) {
+			return DECODE_BREAK;
+		}
+	}
+	if(mad_frame_decode(&data->frame,&data->stream)) {
+#ifdef HAVE_ID3TAG
+		if((data->stream).error==MAD_ERROR_LOSTSYNC) {
+			signed long tagsize = id3_tag_query(
+					(data->stream).this_frame,
+					(data->stream).bufend-
+					(data->stream).this_frame);
+			if(tagsize>0) {
+				mad_stream_skip(&(data->stream),tagsize);
+				return DECODE_CONT;
+			}
+		}
+#endif
+		if(MAD_RECOVERABLE((data->stream).error)) {
+			return DECODE_SKIP;
+		}
+		else {
+			if((data->stream).error==MAD_ERROR_BUFLEN) return DECODE_CONT;
+			else
+			{
+				ERROR("unrecoverable frame level error "
+					"(%s).\n",
+					mad_stream_errorstr(&data->stream));
+				data->flush = 0;
+				return DECODE_BREAK;
+			}
+		}
+	}
+
+	return DECODE_OK;
+}
+
+/* xing stuff stolen from alsaplayer */
+# define XING_MAGIC	(('X' << 24) | ('i' << 16) | ('n' << 8) | 'g')
+
+struct xing {
+  	long flags;			/* valid fields (see below) */
+  	unsigned long frames;		/* total number of frames */
+  	unsigned long bytes;		/* total number of bytes */
+  	unsigned char toc[100];	/* 100-point seek table */
+  	long scale;			/* ?? */
+};
+
+enum {
+  	XING_FRAMES = 0x00000001L,
+  	XING_BYTES  = 0x00000002L,
+  	XING_TOC    = 0x00000004L,
+  	XING_SCALE  = 0x00000008L
+};
+
+int parse_xing(struct xing *xing, struct mad_bitptr ptr, unsigned int bitlen)
+{
+  	if (bitlen < 64 || mad_bit_read(&ptr, 32) != XING_MAGIC) goto fail;
+
+  	xing->flags = mad_bit_read(&ptr, 32);
+  	bitlen -= 64;
+
+  	if (xing->flags & XING_FRAMES) {
+    		if (bitlen < 32) goto fail;
+    		xing->frames = mad_bit_read(&ptr, 32);
+    		bitlen -= 32;
+  	}
+
+  	if (xing->flags & XING_BYTES) {
+    		if (bitlen < 32) goto fail;
+    		xing->bytes = mad_bit_read(&ptr, 32);
+    		bitlen -= 32;
+  	}
+
+  	if (xing->flags & XING_TOC) {
+    		int i;
+    		if (bitlen < 800) goto fail;
+      		for (i = 0; i < 100; ++i) xing->toc[i] = mad_bit_read(&ptr, 8);
+    		bitlen -= 800;
+  	}
+
+  	if (xing->flags & XING_SCALE) {
+    		if (bitlen < 32) goto fail;
+    		xing->scale = mad_bit_read(&ptr, 32);
+    		bitlen -= 32;
+  	}
+
+ 	 return 1;
+
+fail:
+  	xing->flags = 0;
+  	return 0;
+}
+
+int decodeFirstFrame(mp3DecodeData * data, DecoderControl * dc) {
+	struct xing xing;
+	int ret;
+	int skip;
+
+	memset(&xing,0,sizeof(struct xing));
+	xing.flags = 0;
+
+	while(1) {
+		skip = 0;
+		while((ret = decodeNextFrameHeader(data))==DECODE_CONT && 
+				(!dc || !dc->stop));
+		if(ret==DECODE_SKIP) skip = 1;
+		else if(ret==DECODE_BREAK || (dc && dc->stop)) return -1;
+		while((ret = decodeNextFrame(data))==DECODE_CONT && 
+				(!dc || !dc->stop));
+		if(ret==DECODE_BREAK || (dc && dc->stop)) return -1;
+		if(!skip && ret==DECODE_OK) break;
+	}
+
+	if(parse_xing(&xing,data->stream.anc_ptr,data->stream.anc_bitlen)) {
+		if(xing.flags & XING_FRAMES) {
+			mad_timer_t duration = data->frame.header.duration;
+			mad_timer_multiply(&duration,xing.frames);
+			data->muteFrame = MUTEFRAME_SKIP;
+			data->totalTime = ((float)mad_timer_count(duration,
+						MAD_UNITS_MILLISECONDS))/1000;
+			data->maxFrames = xing.frames;
+		}
+	}
+	else {
+		size_t offset = data->inStream->offset;
+		mad_timer_t duration = data->frame.header.duration;
+		float frameTime = ((float)mad_timer_count(duration,
+					MAD_UNITS_MILLISECONDS))/1000;
+		if(data->stream.this_frame!=NULL) {
+			offset-= data->stream.bufend-data->stream.this_frame;
+		}
+		else {
+			offset-= data->stream.bufend-data->stream.buffer;
+		}
+		if(data->inStream->size >= offset) {
+			data->totalTime = ((data->inStream->size-offset)*8.0)/
+					(data->frame).header.bitrate;
+			data->maxFrames = 
+				data->totalTime/frameTime+FRAMES_CUSHION;
+		}
+		else {
+			data->maxFrames = FRAMES_CUSHION;
+			data->totalTime = 0;
+		}
+	}
+
+	data->frameOffset = malloc(sizeof(long)*data->maxFrames);
+	data->times = malloc(sizeof(mad_timer_t)*data->maxFrames);
+
+	return 0;
+}
+
+void mp3DecodeDataFinalize(mp3DecodeData * data) {
+	mad_synth_finish(&data->synth);
+	mad_frame_finish(&data->frame);
+	mad_stream_finish(&data->stream);
+
+	closeInputStream(data->inStream);
+	if(data->frameOffset) free(data->frameOffset);
+	if(data->times) free(data->times);
+}
+
+/* this is primarily used for getting total time for tags */
+int getMp3TotalTime(char * file) {
+        InputStream inStream;
+	mp3DecodeData data;
+	int ret;
+
+        if(openInputStream(&inStream, file) < 0) return -1;
+	initMp3DecodeData(&data,&inStream);
+        data.stream.options |= MAD_OPTION_IGNORECRC;
+	if(decodeFirstFrame(&data, NULL)<0) ret = -1;
+	else ret = data.totalTime+0.5;
+	mp3DecodeDataFinalize(&data);
+
+	return ret;
+}
+
+int openMp3FromInputStream(InputStream * inStream, mp3DecodeData * data,
+		DecoderControl * dc) 
+{
+	initMp3DecodeData(data, inStream);
+        data->stream.options |= MAD_OPTION_IGNORECRC;
+	if(decodeFirstFrame(data, dc)<0) {
+		mp3DecodeDataFinalize(data);
+		return -1;
+	}
+
+	return 0;
+}
+
+int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc) {
+	int i;
+	int ret;
+	struct audio_dither dither;
+	int skip;
+
+	if(data->currentFrame>=data->highestFrame) { 
+		mad_timer_add(&data->timer,(data->frame).header.duration);
+		data->bitRate = (data->frame).header.bitrate;
+		if(data->currentFrame>=data->maxFrames) {
+			data->currentFrame = data->maxFrames - 1;
+		}
+		else data->highestFrame++;
+		data->frameOffset[data->currentFrame] = data->inStream->offset;
+		if(data->stream.this_frame!=NULL) {
+			data->frameOffset[data->currentFrame]-= 
+					data->stream.bufend-
+					data->stream.this_frame;
+		}
+		else {
+			data->frameOffset[data->currentFrame]-= 
+					data->stream.bufend-data->stream.buffer;
+		}
+		data->times[data->currentFrame] = data->timer;
+	}
+	else data->timer = data->times[data->currentFrame];
+	data->currentFrame++;
+	data->elapsedTime = ((float)mad_timer_count(data->timer,MAD_UNITS_MILLISECONDS))/1000;
+
+	switch(data->muteFrame) {
+        case MUTEFRAME_SKIP:
+		data->muteFrame = 0;
+                break;
+        case MUTEFRAME_SEEK:
+		if(dc->seekWhere<=data->elapsedTime) {
+                        data->outputPtr = data->outputBuffer;
+                        clearOutputBuffer(cb);
+			data->muteFrame = 0;
+			dc->seek = 0;
+		}
+                break;
+        default:
+		mad_synth_frame(&data->synth,&data->frame);
+
+		for(i=0;i<(data->synth).pcm.length;i++) {
+			mpd_sint16 * sample;
+
+			sample = (mpd_sint16 *)data->outputPtr;	
+			*sample = (mpd_sint16) audio_linear_dither(16,
+					(data->synth).pcm.samples[0][i],
+					&dither);
+			data->outputPtr+=2;
+
+			if(MAD_NCHANNELS(&(data->frame).header)==2) {
+				sample = (mpd_sint16 *)data->outputPtr;	
+				*sample = (mpd_sint16) audio_linear_dither(16,
+						(data->synth).pcm.samples[1][i],
+						&dither);
+				data->outputPtr+=2;
+			}
+
+			if(data->outputPtr==data->outputBufferEnd) {
+                                long ret;
+                                ret = sendDataToOutputBuffer(cb,
+                                                data->inStream,
+                                                dc,
+                                                data->inStream->seekable,
+                                                data->outputBuffer,
+                                                MP3_DATA_OUTPUT_BUFFER_SIZE,
+                                                data->elapsedTime,
+                                                data->bitRate/1000);
+                                if(ret == OUTPUT_BUFFER_DC_STOP) {
+                                        return DECODE_BREAK;
+                                }
+
+                                data->outputPtr = data->outputBuffer;
+
+                                if(ret == OUTPUT_BUFFER_DC_SEEK) break;
+			}
+		}
+
+		if(dc->seek && data->inStream->seekable) {
+			long i = 0;
+			data->muteFrame = MUTEFRAME_SEEK;
+			while(i<data->highestFrame && dc->seekWhere >
+					((float)mad_timer_count(data->times[i],
+					MAD_UNITS_MILLISECONDS))/1000) 
+			{
+				i++;
+			}
+			if(i<data->highestFrame) {
+				if(seekMp3InputBuffer(data,
+						data->frameOffset[i]) == 0)
+                                {
+                                        data->outputPtr = data->outputBuffer;
+                                        clearOutputBuffer(cb);
+				        data->currentFrame = i;
+                                }
+                                else dc->seekError = 1;
+				data->muteFrame = 0;
+				dc->seek = 0;
+			}
+		}
+                else if(dc->seek && !data->inStream->seekable) {
+                        dc->seek = 0;
+                        dc->seekError = 1;
+                }
+	}
+
+	while(1) {
+		skip = 0;
+		while((ret = decodeNextFrameHeader(data))==DECODE_CONT &&
+				!dc->stop && !dc->seek);
+		if(ret==DECODE_BREAK || dc->stop || dc->seek) break;
+		else if(ret==DECODE_SKIP) skip = 1;
+		if(!data->muteFrame) {
+			while((ret = decodeNextFrame(data))==DECODE_CONT &&
+					!dc->stop && !dc->seek);
+			if(ret==DECODE_BREAK || dc->stop || dc->seek) break;
+		}
+		if(!skip && ret==DECODE_OK) break;
+	}
+
+	if(dc->stop) return DECODE_BREAK;
+
+	return ret;
+}
+
+void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, AudioFormat * af) {
+	af->bits = 16;
+	af->sampleRate = (data->frame).header.samplerate;
+	af->channels = MAD_NCHANNELS(&(data->frame).header);
+}
+
+int mp3_decode(OutputBuffer * cb, DecoderControl * dc, InputStream * inStream) {
+	mp3DecodeData data;
+
+	if(openMp3FromInputStream(inStream, &data, dc) < 0) {
+                closeInputStream(inStream);
+		if(!dc->stop) {
+                        ERROR("Input does not appear to be a mp3 bit stream.\n");
+		        return -1;
+                }
+                else {
+                        dc->state = DECODE_STATE_STOP;
+                        dc->stop = 0;
+                }
+                return 0;
+	}
+
+	initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
+        getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+        
+	dc->totalTime = data.totalTime;
+	dc->state = DECODE_STATE_DECODE;
+
+	while(mp3Read(&data,cb,dc)!=DECODE_BREAK);
+	/* send last little bit if not dc->stop */
+	if(data.outputPtr!=data.outputBuffer && data.flush)  {
+        	sendDataToOutputBuffer(cb, NULL, dc, 
+                                data.inStream->seekable,
+                                data.outputBuffer,
+                                data.outputPtr-data.outputBuffer,
+                                data.elapsedTime,data.bitRate/1000);
+	}
+
+	flushOutputBuffer(cb);
+	mp3DecodeDataFinalize(&data);
+
+	/*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;
+		
+	return 0;
+}
+
+MpdTag * mp3_tagDup(char * utf8file) {
+	MpdTag * ret = NULL;
+	int time;
+
+	ret = id3Dup(utf8file);
+
+	time = getMp3TotalTime(rmp2amp(utf8ToFsCharset(utf8file)));
+
+	if(time>=0) {
+		if(!ret) ret = newMpdTag();
+		ret->time = time;
+	}
+
+	if(ret) validateUtf8Tag(ret);
+
+	return ret;
+}
+
+char * mp3_suffixes[] = {"mp3", NULL};
+char * mp3_mimeTypes[] = {"audio/mpeg", NULL};
+
+InputPlugin mp3Plugin = 
+{
+	"mp3",
+	mp3_decode,
+	NULL,
+	mp3_tagDup,
+	INPUT_PLUGIN_STREAM_FILE | INPUT_PLUGIN_STREAM_URL,
+	mp3_suffixes,
+	mp3_mimeTypes
+};
+#else
+
+InputPlugin mp3Plugin = 
+{
+	NULL,
+	NULL,
+	NULL,
+	0,
+	NULL,
+	NULL
+};
+
+#endif
diff --git a/src/inputPlugins/ogg_plugin.c b/src/inputPlugins/ogg_plugin.c
new file mode 100644
index 000000000..af617378e
--- /dev/null
+++ b/src/inputPlugins/ogg_plugin.c
@@ -0,0 +1,364 @@
+/* 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
+ *
+ * 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_OGG
+
+#include "../command.h"
+#include "../utils.h"
+#include "../audio.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+#include "../inputStream.h"
+#include "../outputBuffer.h"
+#include "../replayGain.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <vorbis/vorbisfile.h>
+#include <errno.h>
+
+#ifdef WORDS_BIGENDIAN
+#define OGG_DECODE_USE_BIGENDIAN	1
+#else
+#define OGG_DECODE_USE_BIGENDIAN	0
+#endif
+
+typedef struct _OggCallbackData {
+        InputStream * inStream;
+        DecoderControl * dc;
+} OggCallbackData;
+
+/* this is just for tag parsing for db import! */
+int getOggTotalTime(char * file) {
+	OggVorbis_File vf;
+	FILE * oggfp;
+	int totalTime;
+	
+	if(!(oggfp = fopen(file,"r"))) return -1;
+		
+	if(ov_open(oggfp, &vf, NULL, 0) < 0) {
+		fclose(oggfp);
+		return -1;
+	}
+	
+	totalTime = ov_time_total(&vf,-1)+0.5;
+
+	ov_clear(&vf);
+
+	return totalTime;
+}
+
+size_t ogg_read_cb(void * ptr, size_t size, size_t nmemb, void * vdata)
+{
+	size_t ret = 0;
+        OggCallbackData * data = (OggCallbackData *)vdata;
+
+        while(1) {
+	        ret = readFromInputStream(data->inStream,ptr,size,nmemb);
+                if(ret == 0 && !inputStreamAtEOF(data->inStream) && 
+                                !data->dc->stop) 
+                {
+                        my_usleep(10000);
+                }
+                else break;
+        }
+        errno = 0;
+	/*if(ret<0) errno = ((InputStream *)inStream)->error;*/
+
+	return ret;
+}
+
+int ogg_seek_cb(void * vdata, ogg_int64_t offset, int whence) {
+        OggCallbackData * data = (OggCallbackData *)vdata;
+
+	return seekInputStream(data->inStream,offset,whence);
+}
+
+int ogg_close_cb(void * vdata) {
+        OggCallbackData * data = (OggCallbackData *)vdata;
+
+	return closeInputStream(data->inStream);
+}
+
+long ogg_tell_cb(void * vdata) {
+        OggCallbackData * data = (OggCallbackData *)vdata;
+
+	return (long)(data->inStream->offset);
+}
+
+char * ogg_parseComment(char * comment, char * needle) {
+        int len = strlen(needle);
+
+        if(strncasecmp(comment,needle,len)) return comment+len;
+
+        return NULL;
+}
+
+float ogg_getReplayGainScale(char ** comments) {
+        int trackGainFound = 0;
+        int albumGainFound = 0;
+        float trackGain = 1.0;
+        float albumGain = 1.0;
+        float trackPeak = 0.0;
+        float albumPeak = 0.0;
+        char * temp;
+        int replayGainState = getReplayGainState();
+
+        if(replayGainState == REPLAYGAIN_OFF) return 1.0;
+
+        while(*comments) {
+                if((temp = ogg_parseComment(*comments,"replaygain_track_gain"))) 
+                {
+                        trackGain = atof(temp);
+                        trackGainFound = 1;
+                }
+                else if((temp = ogg_parseComment(*comments,
+                                        "replaygain_album_gain"))) 
+                {
+                        albumGain = atof(temp);
+                        albumGainFound = 1;
+                }
+                else if((temp = ogg_parseComment(*comments,
+                                        "replaygain_track_peak"))) 
+                {
+                        trackPeak = atof(temp);
+                }
+                else if((temp = ogg_parseComment(*comments,
+                                        "replaygain_album_peak"))) 
+                {
+                        albumPeak = atof(temp);
+                }
+
+                comments++;
+        }
+
+        switch(replayGainState) {
+        case REPLAYGAIN_ALBUM:
+                if(albumGainFound) {
+                        return computeReplayGainScale(albumGain,albumPeak);
+                }
+        default:
+                return computeReplayGainScale(trackGain,trackPeak);
+        }
+
+        return 1.0;
+}
+
+int ogg_decode(OutputBuffer * cb, DecoderControl * dc, InputStream * inStream)
+{
+	OggVorbis_File vf;
+	ov_callbacks callbacks;
+        OggCallbackData data;
+
+        data.inStream = inStream;
+        data.dc = dc;
+
+	callbacks.read_func = ogg_read_cb;
+	callbacks.seek_func = ogg_seek_cb;
+	callbacks.close_func = ogg_close_cb;
+	callbacks.tell_func = ogg_tell_cb;
+	
+	if(ov_open_callbacks(&data, &vf, NULL, 0, callbacks) < 0) {
+		closeInputStream(inStream);
+		if(!dc->stop) {
+		        ERROR("Input does not appear to be an Ogg Vorbis stream.\n");
+                        return -1;
+                }
+                else {
+                        dc->state = DECODE_STATE_STOP;
+                        dc->stop = 0;
+                }
+                return 0;
+	}
+	
+	{
+		vorbis_info *vi=ov_info(&vf,-1);
+		dc->audioFormat.bits = 16;
+		dc->audioFormat.channels = vi->channels;
+		dc->audioFormat.sampleRate = vi->rate;
+                getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
+	}
+
+	dc->totalTime = ov_time_total(&vf,-1);
+        if(dc->totalTime < 0) dc->totalTime = 0;
+	dc->state = DECODE_STATE_DECODE;
+
+	{
+		int current_section;
+		int eof = 0;
+		long ret;
+#define OGG_CHUNK_SIZE 4096
+		char chunk[OGG_CHUNK_SIZE];
+		int chunkpos = 0;
+		long bitRate = 0;
+		long test;
+                float replayGainScale = ogg_getReplayGainScale(
+                                ov_comment(&vf,-1)->user_comments);
+
+		while(!eof) {
+			if(dc->seek) {
+				if(0 == ov_time_seek_page(&vf,dc->seekWhere)) {
+                                        clearOutputBuffer(cb);
+				        chunkpos = 0;
+                                }
+                                else dc->seekError = 1;
+				dc->seek = 0;
+			}
+			ret = ov_read(&vf, chunk+chunkpos, 
+					OGG_CHUNK_SIZE-chunkpos,
+					OGG_DECODE_USE_BIGENDIAN,
+					2, 1, &current_section);
+
+			if(ret <= 0 && ret != OV_HOLE) {
+				eof = 1;
+				break;
+			}
+                        if(ret == OV_HOLE) ret = 0;
+
+			chunkpos+=ret;
+
+			if(chunkpos >= OGG_CHUNK_SIZE) {
+				if((test = ov_bitrate_instant(&vf))>0) {
+					bitRate = test/1000;
+				}
+                                doReplayGain(chunk,ret,&(dc->audioFormat),
+                                                replayGainScale);
+				sendDataToOutputBuffer(cb, inStream, dc, 
+                                        inStream->seekable, chunk, chunkpos, 
+					ov_time_tell(&vf), bitRate);
+				if(dc->stop) break;
+				chunkpos = 0;
+			}
+		}
+
+		if(!dc->stop && chunkpos > 0) {
+			sendDataToOutputBuffer(cb, NULL, dc, inStream->seekable,                                        chunk, chunkpos,
+					ov_time_tell(&vf), bitRate);
+		}
+
+		ov_clear(&vf);
+
+		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;
+	}
+
+	return 0;
+}
+
+MpdTag * oggTagDup(char * utf8file) {
+	MpdTag * ret = NULL;
+	FILE * fp;
+	OggVorbis_File vf;
+	char ** comments;
+	char * temp;
+	char * s1;
+	char * s2;
+
+	fp = fopen(rmp2amp(utf8ToFsCharset(utf8file)),"r"); 
+	if(!fp) return NULL;
+	if(ov_open(fp,&vf,NULL,0)<0) {
+		fclose(fp);
+		return NULL;
+	}
+
+	ret = newMpdTag();
+	ret->time = (int)(ov_time_total(&vf,-1)+0.5);
+
+	comments = ov_comment(&vf,-1)->user_comments;
+
+	while(*comments) {
+		temp = strdup(*comments);
+		++comments;
+		if(!(s1 = strtok(temp,"="))) continue;
+		s2 = strtok(NULL,"");
+		if(!s1 || !s2);
+		else if(0==strcasecmp(s1,"artist")) {
+			if(!ret->artist) {
+				stripReturnChar(s2);
+				ret->artist = strdup(s2);
+			}
+		}
+		else if(0==strcasecmp(s1,"title")) {
+			if(!ret->title) {
+				stripReturnChar(s2);
+				ret->title = strdup(s2);
+			}
+		}
+		else if(0==strcasecmp(s1,"album")) {
+			if(!ret->album) {
+				stripReturnChar(s2);
+				ret->album = strdup(s2);
+			}
+		}
+		else if(0==strcasecmp(s1,"tracknumber")) {
+			if(!ret->track) {
+				stripReturnChar(s2);
+				ret->track = strdup(s2);
+			}
+		}
+		free(temp);
+	}
+
+	ov_clear(&vf);
+
+	if(ret) validateUtf8Tag(ret);
+
+	return ret;	
+}
+
+char * oggSuffixes[] = {"ogg", NULL};
+char * oggMimeTypes[] = {"application/ogg", NULL};
+
+InputPlugin oggPlugin =
+{
+        "ogg",
+        ogg_decode,
+        NULL,
+        oggTagDup,
+        INPUT_PLUGIN_STREAM_URL | INPUT_PLUGIN_STREAM_FILE,
+        oggSuffixes,
+        oggMimeTypes
+};
+
+#else
+
+InputPlugin oggPlugin = 
+{
+        NULL,
+        NULL,
+        NULL,
+        0,
+        NULL,
+        NULL
+};
+
+#endif