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

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git-svn-id: https://svn.musicpd.org/mpd/trunk@1125 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes 2004-05-21 22:31:07 +00:00
parent aea1ae9b9e
commit 547e358796
11 changed files with 47 additions and 34 deletions
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13
TODO
View File

@ -1,21 +1,16 @@
1) play streams
a) make seekings non-blocking:
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
a) 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!
e) fetch metadata and store in DecoderControl and pass to
b) put some sort of error reporting for streaming/inputStream!
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

4
src/aac_decode.c
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@ -362,8 +362,8 @@ int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
sampleBufferLen = sampleCount*2;
sendDataToOutputBuffer(cb,dc,sampleBuffer,sampleBufferLen,
time,bitRate);
sendDataToOutputBuffer(cb, NULL, dc, sampleBuffer,
sampleBufferLen, time, bitRate);
if(dc->seek) dc->seek = 0;
else if(dc->stop) {
eof = 1;

6
src/audiofile_decode.c
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@ -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);

4
src/flac_decode.c
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@ -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,
data->chunk_length,data->time,data->bitRate))
switch(sendDataToOutputBuffer(data->cb, NULL, data->dc, data->chunk,
data->chunk_length, data->time, data->bitRate))
{
case OUTPUT_BUFFER_DC_STOP:
return -1;

3
src/inputStream.h
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@ -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,

14
src/inputStream_http.c
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@ -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);*/
data->buflen += readed;
}
/*fwrite(data->buffer+data->buflen,1,readed,stdout);*/
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: */

6
src/mp3_decode.c
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@ -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)
{

4
src/mp4_decode.c
View File

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

8
src/ogg_decode.c
View File

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

11
src/outputBuffer.c
View File

@ -43,8 +43,9 @@ void flushOutputBuffer(OutputBuffer * cb) {
}
}
int sendDataToOutputBuffer(OutputBuffer * cb, DecoderControl * dc,
char * dataIn, long dataInLen, float time, mpd_uint16 bitRate)
int sendDataToOutputBuffer(OutputBuffer * cb, InputStream * inStream,
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;

8
src/outputBuffer.h
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@ -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,
char * data, long datalen, float time, mpd_uint16 bitRate);
/* we send inStream where for buffering the inputStream while waiting to
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: */