new OutputBuffer abstraction stuff, implemented for mp3, now need to

src/Makefile.am

@@ -5,13 +5,15 @@ mpd_headers = buffer2array.h interface.h command.h playlist.h ls.h \
 	tag.h player.h listen.h conf.h ogg_decode.h volume.h flac_decode.h \
 	audio.h playerData.h stats.h myfprintf.h sig_handlers.h decode.h log.h \
 	audiofile_decode.h charConv.h permission.h mpd_types.h pcm_utils.h \
-	mp4_decode.h aac_decode.h signal_check.h utf8.h inputStream.h
+	mp4_decode.h aac_decode.h signal_check.h utf8.h inputStream.h \
+	outputBuffer.h
 mpd_SOURCES = main.c buffer2array.c interface.c command.c playlist.c ls.c \
 	song.c list.c directory.c tables.c utils.c path.c mp3_decode.c \
 	tag.c player.c listen.c conf.c ogg_decode.c volume.c flac_decode.c \
 	audio.c playerData.c stats.c myfprintf.c sig_handlers.c decode.c log.c \
 	audiofile_decode.c charConv.c permission.c pcm_utils.c mp4_decode.c \
-	aac_decode.c signal_check.c utf8.c inputStream.c $(mpd_headers)
+	aac_decode.c signal_check.c utf8.c inputStream.c outputBuffer.c \
+	$(mpd_headers)
 
 mpd_CFLAGS = $(MPD_CFLAGS)
 mpd_LDADD = $(MPD_LIBS) $(ID3_LIB) $(MAD_LIB) $(MP4FF_LIB)

src/aac_decode.c

@@ -250,7 +250,7 @@ int getAacTotalTime(char * file) {
 }
 
 
-int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
+int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 	float time;
 	float totalTime;
 	faacDecHandle decoder;

src/aac_decode.h

@@ -27,7 +27,7 @@
 
 int getAacTotalTime(char * file);
 
-int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);
+int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 #endif /* HAVE_FAAD */
 

src/audiofile_decode.c

@@ -51,7 +51,7 @@ int getAudiofileTotalTime(char * file)
 	return time;
 }
 
-int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
+int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 {
 	int fs, frame_count;
 	AFfilehandle af_fp;

src/audiofile_decode.h

@@ -27,7 +27,7 @@
 
 #include "playerData.h"
 
-int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);;
+int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 int getAudiofileTotalTime(char * file);
 

src/decode.c

@@ -112,7 +112,7 @@ int calculateCrossFadeChunks(PlayerControl * pc, AudioFormat * af) {
 }
 
 int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
-		Buffer * cb) 
+		OutputBuffer * cb) 
 {
 	while(decode_pid && *decode_pid>0 && dc->start) my_usleep(1000);
 
@@ -143,7 +143,7 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 }
 
 void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc, 
-		Buffer * cb) 
+		OutputBuffer * cb) 
 {
 	if(decode_pid && *decode_pid>0) {
 		cb->next = -1;
@@ -217,7 +217,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 		return; \
 	}
 
-int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af, 
+int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af, 
 			DecoderControl * dc) {
 	int pid;
 	int ret; 
@@ -311,7 +311,7 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af,
  * parent process does playing audio
  */
 void decode() {
-	Buffer * cb;
+	OutputBuffer * cb;
 	PlayerControl * pc;
 	AudioFormat * af;
 	DecoderControl * dc;

src/flac_decode.c

@@ -37,7 +37,7 @@ typedef struct {
 	float time;
 	int bitRate;
 	FLAC__uint64 position;
-	Buffer * cb;
+	OutputBuffer * cb;
 	AudioFormat * af;
 	DecoderControl * dc;
         char * file;
@@ -65,7 +65,7 @@ FLAC__SeekableStreamDecoderLengthStatus flacLength(
                 const FLAC__SeekableStreamDecoder *, FLAC__uint64 *, void *);
 FLAC__bool flacEOF(const FLAC__SeekableStreamDecoder *, void *);
 
-void flacPlayFile(char *file, Buffer * cb, AudioFormat * af, 
+void flacPlayFile(char *file, OutputBuffer * cb, AudioFormat * af, 
 	DecoderControl *dc)
 {
 	FLAC__SeekableStreamDecoder * flacDec;
@@ -392,7 +392,7 @@ int getFlacTotalTime(char * file) {
 	return (int)(totalTime+0.5);
 }
 
-int flac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
+int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 	if(flac_getAudioFormatAndTime(dc->file,af,&(cb->totalTime))<0) {
 		ERROR("\"%s\" doesn't seem to be a flac\n",dc->file);
 		return -1;

src/flac_decode.h

@@ -25,7 +25,7 @@
 
 #include <stdio.h>
 
-int flac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);
+int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 int getFlacTotalTime(char * file);
 

src/mp3_decode.c

@@ -37,6 +37,7 @@
 #include "log.h"
 #include "utils.h"
 #include "inputStream.h"
+#include "outputBuffer.h"
 
 #include <stdio.h>
 #include <string.h>
@@ -111,13 +112,15 @@ signed long audio_linear_dither(unsigned int bits, mad_fixed_t sample, struct au
 
 /* 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[CHUNK_SIZE];
+	char outputBuffer[MP3_DATA_OUTPUT_BUFFER_SIZE];
 	char * outputPtr;
 	char * outputBufferEnd;
 	float totalTime;
@@ -141,7 +144,7 @@ int initMp3DecodeData(mp3DecodeData * data, char * file) {
 	if(ret<0) return -1;
 
 	data->outputPtr = data->outputBuffer;
-	data->outputBufferEnd = data->outputBuffer+CHUNK_SIZE;
+	data->outputBufferEnd = data->outputBuffer+MP3_DATA_OUTPUT_BUFFER_SIZE;
 	data->muteFrame = 0;
 	data->highestFrame = 0;
 	data->maxFrames = 0;
@@ -406,29 +409,7 @@ int openMp3(char * file, mp3DecodeData * data) {
 	return 0;
 }
 
-int mp3ChildSendData(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
-	while(cb->begin==cb->end && cb->wrap && !dc->stop && !dc->seek) 
-		my_usleep(10000);
-	if(dc->stop) return -1;
-	/* just for now, so it doesn't hang */
-	if(dc->seek) return 0;
-	/* be sure to remove this! */
-
-	memcpy(cb->chunks+cb->end*CHUNK_SIZE,data->outputBuffer,CHUNK_SIZE);
-	cb->chunkSize[cb->end] = data->outputPtr-data->outputBuffer;
-	cb->bitRate[cb->end] = data->bitRate/1000;
-	cb->times[cb->end] = data->elapsedTime;
-
-	cb->end++;
-	if(cb->end>=buffered_chunks) {
-		cb->end = 0;
-		cb->wrap = 1;
-	}
-
-	return 0;
-}
-
-int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
+int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc) {
 	static int i;
 	static int ret;
 	static struct audio_dither dither;
@@ -464,6 +445,8 @@ int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
 		}
 	}
 	else {
+                long ret;
+            
 		mad_synth_frame(&data->synth,&data->frame);
 
 		for(i=0;i<(data->synth).pcm.length;i++) {
@@ -484,12 +467,24 @@ int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
 			}
 
 			if(data->outputPtr==data->outputBufferEnd) {
-				if(mp3ChildSendData(data,cb,dc)<0) {
-					data->flush = 0;
-					return DECODE_BREAK;
-				}
-				data->outputPtr = data->outputBuffer;
-				if(dc->seek) break;
+                                ret = sendDataToOutputBuffer(cb,dc,
+                                                0,data->outputBuffer,
+                                                MP3_DATA_OUTPUT_BUFFER_SIZE,
+                                                data->elapsedTime,
+                                                data->bitRate/1000);
+                                if(ret == OUTPUT_BUFFER_DC_STOP) {
+                                        return DECODE_BREAK;
+                                }
+                                if(ret >= 0) {
+                                        memmove(data->outputBuffer,
+                                                data->outputBuffer+ret,
+                                                MP3_DATA_OUTPUT_BUFFER_SIZE-
+                                                ret);
+                                        data->outputPtr-=ret;
+                                }
+                                else data->outputPtr = data->outputBuffer;
+
+                                if(ret == OUTPUT_BUFFER_DC_SEEK) break;
 			}
 		}
 
@@ -534,7 +529,7 @@ void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, AudioFormat * af) {
 	af->channels = MAD_NCHANNELS(&(data->frame).header);
 }
 
-int mp3_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
+int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 	mp3DecodeData data;
 
 	if(openMp3(dc->file,&data) < 0) {
@@ -550,7 +545,9 @@ int mp3_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 	while(mp3Read(&data,cb,dc)!=DECODE_BREAK);
 	/* send last little bit if not dc->stop */
 	if(data.outputPtr!=data.outputBuffer && data.flush)  {
-		mp3ChildSendData(&data,cb,dc);
+        	sendDataToOutputBuffer(cb,dc,1,data.outputBuffer,
+                                data.outputPtr-data.outputBuffer,
+                                data.elapsedTime,data.bitRate/1000);
 	}
 
 	mp3DecodeDataFinalize(&data);
@@ -567,4 +564,4 @@ int mp3_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 }
 
 #endif
-/* vim:set shiftwidth=4 tabstop=8 expandtab: */
+/* vim:set shiftwidth=8 tabstop=8 expandtab: */

src/mp3_decode.h

@@ -28,7 +28,7 @@
 /* this is primarily used in tag.c */
 int getMp3TotalTime(char * file);
 
-int mp3_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);
+int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 #endif
 

src/mp4_decode.c

@@ -84,7 +84,7 @@ uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position) {
 }       
 		    
 
-int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
+int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 	mp4ff_t * mp4fh;
 	mp4ff_callback_t * mp4cb; 
 	int32_t track;

src/mp4_decode.h

@@ -29,7 +29,7 @@
 
 int mp4_getAACTrack(mp4ff_t *infile);
 
-int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);
+int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer, 
 	uint32_t length);

src/ogg_decode.c

@@ -82,7 +82,7 @@ long ogg_tell_cb(void * inStream) {
 	return ((InputStream *)inStream)->offset;
 }
 
-int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
+int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 {
 	OggVorbis_File vf;
 	ov_callbacks callbacks;

src/ogg_decode.h

@@ -25,7 +25,7 @@
 
 #include <stdio.h>
 
-int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc);
+int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
 
 int getOggTotalTime(char * file);
 

src/outputBuffer.c

@@ -20,17 +20,13 @@
 
 #include "pcm_utils.h"
 #include "playerData.h"
-#include "log.h"
 #include "utils.h"
 
 #include <string.h>
-#include <errno.h>
 
-#define OUTPUT_BUFFER_DC_STOP   -1
-#define OUTPUT_BUFFER_DC_SEEK   -2
-
-long sendDataToOutputBuffer(Buffer * cb, DecoderControl * dc, int flushAllData,
-                char * data, long datalen, float time, mpd_uint16 bitRate)
+long sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc, 
+                int flushAllData, char * data, long datalen, float time, 
+		mpd_uint16 bitRate)
 {
         long dataSent = 0;
         long dataToSend;

src/outputBuffer.h

+/* 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
+ */
+
+#ifndef OUTPUT_BUFFER_H
+#define OUTPUT_BUFFER_H
+
+#include "mpd_types.h"
+#include "decode.h"
+
+#define OUTPUT_BUFFER_DC_STOP   -1
+#define OUTPUT_BUFFER_DC_SEEK   -2
+
+typedef struct _OutputBuffer {
+	char * volatile chunks;
+	mpd_uint16 * volatile chunkSize;
+	mpd_uint16 * volatile bitRate;
+	float * volatile times;
+	mpd_sint16 volatile begin;
+	mpd_sint16 volatile end;
+	mpd_sint16 volatile next;
+	mpd_sint8 volatile wrap;
+	float totalTime;
+} OutputBuffer;
+
+long sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc, 
+               int flushAllData, char * data, long datalen, float time, 
+               mpd_uint16 bitRate);
+
+#endif
+/* vim:set shiftwidth=4 tabstop=8 expandtab: */

src/playerData.c

@@ -39,7 +39,7 @@ void initPlayerData() {
 	int crossfade = 0;
 	size_t bufferSize;
 	size_t allocationSize;
-	Buffer * buffer;
+	OutputBuffer * buffer;
 
 	bufferSize = strtol(getConf()[CONF_BUFFER_SIZE],&test,10);
 	if(*test!='\0' || bufferSize<=0) {

src/playerData.h

@@ -25,6 +25,7 @@
 #include "player.h"
 #include "decode.h"
 #include "mpd_types.h"
+#include "outputBuffer.h"
 
 /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
 #define CHUNK_SIZE		1020
@@ -32,20 +33,8 @@
 extern int buffered_before_play;
 extern int buffered_chunks;
 
-typedef struct _Buffer {
-	char * volatile chunks;
-	mpd_uint16 * volatile chunkSize;
-	mpd_uint16 * volatile bitRate;
-	float * volatile times;
-	mpd_sint16 volatile begin;
-	mpd_sint16 volatile end;
-	mpd_sint16 volatile next;
-	mpd_sint8 volatile wrap;
-	float totalTime;
-} Buffer;
-
 typedef struct _PlayerData {
-	Buffer buffer;
+	OutputBuffer buffer;
 	AudioFormat audioFormat;
 	PlayerControl playerControl;
 	DecoderControl decoderControl;
@@ -54,7 +43,6 @@ typedef struct _PlayerData {
 void initPlayerData();
 
 PlayerData * getPlayerData();
-Buffer * getBuffer();
 
 void freePlayerData();