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authorWarren Dukes <warren.dukes@gmail.com>2004-05-07 15:58:04 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-05-07 15:58:04 +0000
commit3794126e5609112d68a2e0c9cbae5a923da301b6 (patch)
treee44ad37c6b27f6f4a52b63c6c1057a860f93ff80
parent9196023f14acf37ee47ac7798029270476d7d3cb (diff)
downloadmpd-3794126e5609112d68a2e0c9cbae5a923da301b6.tar.gz
mpd-3794126e5609112d68a2e0c9cbae5a923da301b6.tar.xz
mpd-3794126e5609112d68a2e0c9cbae5a923da301b6.zip
new OutputBuffer abstraction stuff, implemented for mp3, now need to
implement in other decoders git-svn-id: https://svn.musicpd.org/mpd/trunk@940 09075e82-0dd4-0310-85a5-a0d7c8717e4f
-rw-r--r--src/Makefile.am6
-rw-r--r--src/aac_decode.c2
-rw-r--r--src/aac_decode.h2
-rw-r--r--src/audiofile_decode.c2
-rw-r--r--src/audiofile_decode.h2
-rw-r--r--src/decode.c8
-rw-r--r--src/flac_decode.c6
-rw-r--r--src/flac_decode.h2
-rw-r--r--src/mp3_decode.c65
-rw-r--r--src/mp3_decode.h2
-rw-r--r--src/mp4_decode.c2
-rw-r--r--src/mp4_decode.h2
-rw-r--r--src/ogg_decode.c2
-rw-r--r--src/ogg_decode.h2
-rw-r--r--src/outputBuffer.c10
-rw-r--r--src/outputBuffer.h45
-rw-r--r--src/playerData.c2
-rw-r--r--src/playerData.h16
18 files changed, 103 insertions, 75 deletions
diff --git a/src/Makefile.am b/src/Makefile.am
index 2d19a13ff..33919516f 100644
--- a/src/Makefile.am
+++ b/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)
diff --git a/src/aac_decode.c b/src/aac_decode.c
index 9d56e3392..42da876be 100644
--- a/src/aac_decode.c
+++ b/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;
diff --git a/src/aac_decode.h b/src/aac_decode.h
index bd8149778..62e77eb1e 100644
--- a/src/aac_decode.h
+++ b/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 */
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c
index 79f25e3d4..9a5103716 100644
--- a/src/audiofile_decode.c
+++ b/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;
diff --git a/src/audiofile_decode.h b/src/audiofile_decode.h
index 5562b628e..774d58c1b 100644
--- a/src/audiofile_decode.h
+++ b/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);
diff --git a/src/decode.c b/src/decode.c
index fc0bd84af..221ca4a9b 100644
--- a/src/decode.c
+++ b/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;
diff --git a/src/flac_decode.c b/src/flac_decode.c
index 9760bcb11..ee9255efe 100644
--- a/src/flac_decode.c
+++ b/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;
diff --git a/src/flac_decode.h b/src/flac_decode.h
index 02b3aa092..db7662334 100644
--- a/src/flac_decode.h
+++ b/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);
diff --git a/src/mp3_decode.c b/src/mp3_decode.c
index 9103742e4..973f96541 100644
--- a/src/mp3_decode.c
+++ b/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: */
diff --git a/src/mp3_decode.h b/src/mp3_decode.h
index e75db0763..06525e8c3 100644
--- a/src/mp3_decode.h
+++ b/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
diff --git a/src/mp4_decode.c b/src/mp4_decode.c
index b6cea1d62..e1faff837 100644
--- a/src/mp4_decode.c
+++ b/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;
diff --git a/src/mp4_decode.h b/src/mp4_decode.h
index 0904dcc28..8116758db 100644
--- a/src/mp4_decode.h
+++ b/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);
diff --git a/src/ogg_decode.c b/src/ogg_decode.c
index e37dae2a2..5aad02f78 100644
--- a/src/ogg_decode.c
+++ b/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;
diff --git a/src/ogg_decode.h b/src/ogg_decode.h
index 036b6fc3a..6710c5b8e 100644
--- a/src/ogg_decode.h
+++ b/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);
diff --git a/src/outputBuffer.c b/src/outputBuffer.c
index 507c6fc01..191ea7556 100644
--- a/src/outputBuffer.c
+++ b/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;
diff --git a/src/outputBuffer.h b/src/outputBuffer.h
new file mode 100644
index 000000000..1a5335df2
--- /dev/null
+++ b/src/outputBuffer.h
@@ -0,0 +1,45 @@
+/* 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: */
diff --git a/src/playerData.c b/src/playerData.c
index 21fc30200..17e79543b 100644
--- a/src/playerData.c
+++ b/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) {
diff --git a/src/playerData.h b/src/playerData.h
index eb69ad0fc..387eb07cf 100644
--- a/src/playerData.h
+++ b/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();