do input buffering in while sleeping loop of sending stuff to output buffer

git-svn-id: https://svn.musicpd.org/mpd/trunk@1125 09075e82-0dd4-0310-85a5-a0d7c8717e4f

TODO

@@ -1,21 +1,16 @@
 1) play streams
-	a) make seekings non-blocking:
+	a) deal with pausing better
-		1) player: first check that seekWhere isn't already buffered
-	b) bufferInput in outputBuffer waiting!
-		1) implement some sort of callback mechanism for this
-			for abstraction sake
-	c) deal with pausing better
 		1) seekable, on resuming pause, check if we need to reconnect,
 			jumping to offset
 		2) if seekable, at some point after init, mark this!
 		3) if not seekable, reset buffer, and elapsedTime when
 			unpaused
-	d) put some sort of error reporting for streaming/inputStream!
+	b) put some sort of error reporting for streaming/inputStream!
-	e) fetch metadata and store in DecoderControl and pass to 
+	c) fetch metadata and store in DecoderControl and pass to 
 		PlayerControl
 		1) eventually deal with icy-metadata
 		2) parse metadata on the fly in decoders
-	f) command for dealing with the changing metadata, currentsonginfo
+	d) command for dealing with the changing metadata, currentsonginfo
 		or something
 
 2) how to deal with streams and the db

src/aac_decode.c

@@ -362,8 +362,8 @@ int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
 			
 		sampleBufferLen = sampleCount*2;
 
-		sendDataToOutputBuffer(cb,dc,sampleBuffer,sampleBufferLen,
+		sendDataToOutputBuffer(cb, NULL, dc, sampleBuffer,
-				time,bitRate);
+                                sampleBufferLen, time, bitRate);
 		if(dc->seek) dc->seek = 0;
 		else if(dc->stop) {
 			eof = 1;

src/audiofile_decode.c

@@ -109,7 +109,11 @@ int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) {
 			if(ret<=0) eof = 1;
 			else {
 				current += ret;
-				sendDataToOutputBuffer(cb,dc,chunk,ret*fs,
+				sendDataToOutputBuffer(cb, 
+                                        NULL,
+                                        dc,
+                                        chunk,
+                                        ret*fs,
 					(float)current /
 					(float)dc->audioFormat.sampleRate,
 					bitRate);

src/flac_decode.c

@@ -376,8 +376,8 @@ int flacSendChunk(FlacData * data) {
         doReplayGain(data->chunk,data->chunk_length,&(data->dc->audioFormat),
                         data->replayGainScale);
 
-	switch(sendDataToOutputBuffer(data->cb,data->dc,data->chunk,
+	switch(sendDataToOutputBuffer(data->cb, NULL, data->dc, data->chunk,
-			data->chunk_length,data->time,data->bitRate)) 
+			data->chunk_length, data->time, data->bitRate)) 
 	{
 	case OUTPUT_BUFFER_DC_STOP:
 		return -1;

src/inputStream.h

@@ -53,6 +53,9 @@ int openInputStream(InputStream * inStream, char * url);
 int seekInputStream(InputStream * inStream, long offset, int whence);
 int closeInputStream(InputStream * inStream);
 int inputStreamAtEOF(InputStream * inStream);
+
+/* return value: -1 is error, 1 inidicates stuff was buffered, 0 means nothing
+   was buffered */
 int bufferInputStream(InputStream * inStream);
 
 size_t readFromInputStream(InputStream * inStream, void * ptr, size_t size,

src/inputStream_http.c

@@ -542,20 +542,20 @@ int inputStream_httpBuffer(InputStream * inStream) {
                 readed = read(data->sock, data->buffer+data->buflen, 
                                 (size_t)(HTTP_BUFFER_SIZE-1-data->buflen));
 
-                if(readed < 0 && (errno == EAGAIN || errno == EINTR));
+                if(readed < 0 && (errno == EAGAIN || errno == EINTR)) {
+                        readed = 0;
+                }
                 else if(readed <= 0) {
                         close(data->sock);
                         data->connState = HTTP_CONN_STATE_CLOSED;
+                        readed = 0;
                 }
-                else {
+		/*fwrite(data->buffer+data->buflen,1,readed,stdout);*/
-			/*fwrite(data->buffer+data->buflen,1,readed,stdout);*/
+		data->buflen += readed;
-			data->buflen += readed;
-		}
-
         }
 
 	if(data->buflen > HTTP_PREBUFFER_SIZE) data->prebuffer = 0;
 
-        return 0;
+        return (readed ? 1 : 0);
 }
 /* vim:set shiftwidth=8 tabstop=8 expandtab: */

src/mp3_decode.c

@@ -494,7 +494,9 @@ int mp3Read(mp3DecodeData * data, OutputBuffer * cb, DecoderControl * dc) {
 
 			if(data->outputPtr==data->outputBufferEnd) {
                                 long ret;
-                                ret = sendDataToOutputBuffer(cb,dc,
+                                ret = sendDataToOutputBuffer(cb,
+                                                data->inStream,
+                                                dc,
                                                 data->outputBuffer,
                                                 MP3_DATA_OUTPUT_BUFFER_SIZE,
                                                 data->elapsedTime,
@@ -584,7 +586,7 @@ int mp3_decode(OutputBuffer * cb, DecoderControl * dc, InputStream * inStream) {
 	while(mp3Read(&data,cb,dc)!=DECODE_BREAK);
 	/* send last little bit if not dc->stop */
 	if(data.outputPtr!=data.outputBuffer && data.flush)  {
-        	if(sendDataToOutputBuffer(cb,dc,data.outputBuffer,
+        	if(sendDataToOutputBuffer(cb,NULL,dc,data.outputBuffer,
                                 data.outputPtr-data.outputBuffer,
                                 data.elapsedTime,data.bitRate/1000) == 0)
 		{

src/mp4_decode.c

@@ -279,8 +279,8 @@ int mp4_decode(OutputBuffer * cb, DecoderControl * dc) {
 
 		sampleBuffer+=offset*channels*2;
 
-		sendDataToOutputBuffer(cb,dc,sampleBuffer,
+		sendDataToOutputBuffer(cb, NULL, dc, sampleBuffer,
-				sampleBufferLen,time,bitRate);
+				sampleBufferLen, time, bitRate);
 		if(dc->stop) {
 			eof = 1;
 			break;

src/ogg_decode.c

@@ -241,16 +241,16 @@ int ogg_decode(OutputBuffer * cb, DecoderControl * dc, InputStream * inStream)
 				}
                                 doReplayGain(chunk,ret,&(dc->audioFormat),
                                                 replayGainScale);
-				sendDataToOutputBuffer(cb,dc,chunk,chunkpos,
+				sendDataToOutputBuffer(cb, inStream, dc, chunk,
-					ov_time_tell(&vf),bitRate);
+					chunkpos, ov_time_tell(&vf), bitRate);
 				if(dc->stop) break;
 				chunkpos = 0;
 			}
 		}
 
 		if(!dc->stop && chunkpos > 0) {
-			sendDataToOutputBuffer(cb,dc,chunk,chunkpos,
+			sendDataToOutputBuffer(cb, NULL, dc, chunk, chunkpos,
-					ov_time_tell(&vf),bitRate);
+					ov_time_tell(&vf), bitRate);
 		}
 
 		ov_clear(&vf);

src/outputBuffer.c

@@ -43,8 +43,9 @@ void flushOutputBuffer(OutputBuffer * cb) {
 	}
 }
 
-int sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc, 
+int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,  
-                char * dataIn, long dataInLen, float time, mpd_uint16 bitRate)
+                DecoderControl * dc, char * dataIn, long dataInLen, float time,
+                mpd_uint16 bitRate)
 {
         mpd_uint16 dataToSend;
 	mpd_uint16 chunkLeft;
@@ -75,7 +76,11 @@ int sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc,
 		if(currentChunk != cb->end) {
 	        	while(cb->begin==cb->end && cb->wrap && !dc->stop)
 			{
-		        	my_usleep(10000);
+                                if(!inStream || 
+                                        bufferInputStream(inStream) <= 0)
+                                {
+		        	        my_usleep(10000);
+                                }
 			}
 	        	if(dc->stop) return OUTPUT_BUFFER_DC_STOP;
 

src/outputBuffer.h

@@ -22,6 +22,7 @@
 #include "mpd_types.h"
 #include "decode.h"
 #include "audio.h"
+#include "inputStream.h"
 
 #define OUTPUT_BUFFER_DC_STOP   -1
 #define OUTPUT_BUFFER_DC_SEEK   -2
@@ -42,8 +43,11 @@ void clearOutputBuffer(OutputBuffer * cb);
 
 void flushOutputBuffer(OutputBuffer * cb);
 
-int sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc, 
+/* we send inStream where for buffering the inputStream while waiting to
-               char * data, long datalen, float time, mpd_uint16 bitRate);
+   send the next chunk */
+int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
+                DecoderControl * dc, char * data, long datalen, float time,
+                mpd_uint16 bitRate);
 
 #endif
 /* vim:set shiftwidth=4 tabstop=8 expandtab: */