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-rw-r--r--src/aac_decode.c149
-rw-r--r--src/aac_decode.h2
-rw-r--r--src/audiofile_decode.c2
-rw-r--r--src/decode.c8
-rw-r--r--src/decode.h13
-rw-r--r--src/flac_decode.c2
-rw-r--r--src/mp3_decode.c2
-rw-r--r--src/mp4_decode.c8
-rw-r--r--src/mp4_decode.h2
-rw-r--r--src/ogg_decode.c2
10 files changed, 71 insertions, 119 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c
index 26e430d06..40e1217e4 100644
--- a/src/aac_decode.c
+++ b/src/aac_decode.c
@@ -2,8 +2,6 @@
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- *
* 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
@@ -196,7 +194,11 @@ int initAacBuffer(char * file, AacBuffer * b, float * length) {
if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
}
- if(*length<0) return -1;
+ if(*length<0) {
+ fclose(b->infile);
+ if(b->buffer) free(b->buffer);
+ return -1;
+ }
return 0;
}
@@ -215,59 +217,29 @@ int getAacTotalTime(char * file) {
int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
- /*FILE * fh;
- mp4ff_t * mp4fh;
- mp4ff_callback_t * mp4cb;
- int32_t track;
float time;
- int32_t scale;
+ float totalTime;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
- unsigned char * mp4Buffer;
- int mp4BufferSize;
+ size_t bread;
unsigned long sampleRate;
unsigned char channels;
- long sampleId;
- long numSamples;
int eof = 0;
- long dur;
unsigned int sampleCount;
char * sampleBuffer;
size_t sampleBufferLen;
- unsigned int initial = 1;
int chunkLen = 0;
- float * seekTable;
+ /*float * seekTable;
long seekTableEnd = -1;
- int seekPositionFound = 0;
- long offset;
+ int seekPositionFound = 0;*/
mpd_uint16 bitRate = 0;
+ AacBuffer b;
- fh = fopen(dc->file,"r");
- if(!fh) {
- ERROR("failed to open %s\n",dc->file);
- return -1;
- }
-
- mp4cb = malloc(sizeof(mp4ff_callback_t));
- mp4cb->read = mp4_readCallback;
- mp4cb->seek = mp4_seekCallback;
- mp4cb->user_data = fh;
-
- mp4fh = mp4ff_open_read(mp4cb);
- if(!mp4fh) {
- ERROR("Input does not appear to be a mp4 stream.\n");
- free(mp4cb);
- fclose(fh);
- return -1;
- }
+ printf("aac_decode!\n");
- track = mp4_getAACTrack(mp4fh);
- if(track < 0) {
- ERROR("No AAC track found in mp4 stream.\n");
- mp4ff_close(mp4fh);
- fclose(fh);
- free(mp4cb);
+ if(initAacBuffer(dc->file,&b,&totalTime) < 0) {
+ ERROR("Not AAC file no ADTS or ADIF headers found.\n");
return -1;
}
@@ -285,48 +257,33 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
af->bits = 16;
- mp4Buffer = NULL;
- mp4BufferSize = 0;
- mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
-
- if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
- < 0)
+ fillAacBuffer(&b);
+ if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+ &sampleRate,&channels)) < 0)
{
- ERROR("Error initializing AAC decoder library.\n");
+ ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
- mp4ff_close(mp4fh);
- free(mp4cb);
- fclose(fh);
+ fclose(b.infile);
+ if(b.buffer) free(b.buffer);
return -1;
}
af->sampleRate = sampleRate;
af->channels = channels;
- time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
- scale = mp4ff_time_scale(mp4fh,track);
- if(mp4Buffer) free(mp4Buffer);
-
- if(scale < 0) {
- ERROR("Error getting audio format of mp4 AAC track.\n");
- faacDecClose(decoder);
- mp4ff_close(mp4fh);
- fclose(fh);
- free(mp4cb);
- return -1;
- }
- cb->totalTime = ((float)time)/scale;
-
- numSamples = mp4ff_num_samples(mp4fh,track);
+ cb->totalTime = totalTime+0.5;
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
time = 0.0;
- seekTable = malloc(sizeof(float)*numSamples);
+ advanceAacBuffer(&b,bread);
+ fillAacBuffer(&b);
+
+ /*seekTable = malloc(sizeof(float)*numSamples);*/
- for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
- if(dc->seek && seekTableEnd>1 &&
+ do {
+ /*if(dc->seek && seekTableEnd>1 &&
seekTable[seekTableEnd]>=dc->seekWhere)
{
int i = 2;
@@ -335,9 +292,6 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
time = seekTable[sampleId];
}
- dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
- offset = mp4ff_get_sample_offset(mp4fh,track,sampleId);
-
if(sampleId>seekTableEnd) {
seekTable[sampleId] = time;
seekTableEnd = sampleId;
@@ -358,42 +312,36 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
dc->seek = 0;
}
- if(dc->seek) continue;
-
- if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
- &mp4BufferSize) == 0)
- {
- eof = 1;
- continue;
+ if(dc->seek) continue;*/
+
+ if(dc->seek) {
+ /*chunkLen = 0;
+ cb->wrap = 0;
+ cb->end = 0;*/
+ dc->seekError = 1;
+ dc->seek = 0;
}
+
+ sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer,
+ b.bytesIntoBuffer);
+ advanceAacBuffer(&b,frameInfo.bytesconsumed);
- sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
- mp4BufferSize);
- if(mp4Buffer) free(mp4Buffer);
if(frameInfo.error > 0) {
eof = 1;
break;
}
- if(channels*(dur+offset) > frameInfo.samples) {
- dur = frameInfo.samples;
- offset = 0;
- }
-
- sampleCount = (unsigned long)(dur*channels);
+ sampleCount = (unsigned long)(frameInfo.samples);
if(sampleCount>0) {
- initial =0;
bitRate = frameInfo.bytesconsumed*8.0*
- frameInfo.channels*scale/
+ frameInfo.channels*sampleRate/
frameInfo.samples/1024+0.5;
+ time+= (float)(frameInfo.samples)/channels/sampleRate;
}
-
sampleBufferLen = sampleCount*2;
- sampleBuffer+=offset*channels*2;
-
while(sampleBufferLen>0 && !dc->seek) {
size_t size = sampleBufferLen>CHUNK_SIZE-chunkLen ?
CHUNK_SIZE-chunkLen:
@@ -427,7 +375,11 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
}
}
}
- }
+
+ fillAacBuffer(&b);
+
+ if(b.bytesIntoBuffer==0) eof = 1;
+ } while (!eof);
if(!dc->stop && !dc->seek && chunkLen>0) {
cb->chunkSize[cb->end] = chunkLen;
@@ -440,11 +392,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
chunkLen = 0;
}
- free(seekTable);
+ /*free(seekTable);*/
faacDecClose(decoder);
- mp4ff_close(mp4fh);
- fclose(fh);
- free(mp4cb);
+ fclose(b.infile);
+ if(b.buffer) free(b.buffer);
if(dc->seek) dc->seek = 0;
@@ -452,7 +403,7 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
dc->state = DECODE_STATE_STOP;
dc->stop = 0;
}
- else dc->state = DECODE_STATE_STOP;*/
+ else dc->state = DECODE_STATE_STOP;
return 0;
}
diff --git a/src/aac_decode.h b/src/aac_decode.h
index ad0d75d75..7ce9781f4 100644
--- a/src/aac_decode.h
+++ b/src/aac_decode.h
@@ -2,8 +2,6 @@
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- *
* 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
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c
index 0c2824dc6..9889fbf2f 100644
--- a/src/audiofile_decode.c
+++ b/src/audiofile_decode.c
@@ -86,7 +86,7 @@ int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
while(!eof) {
if(dc->seek) {
- cb->end = 0;
+ cb->end = cb->begin;
cb->wrap = 0;
current = dc->seekWhere * af->sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
diff --git a/src/decode.c b/src/decode.c
index b80166af4..334812d8b 100644
--- a/src/decode.c
+++ b/src/decode.c
@@ -40,6 +40,7 @@
#endif
#ifdef HAVE_FAAD
#include "mp4_decode.h"
+#include "aac_decode.h"
#endif
#include <signal.h>
@@ -159,11 +160,11 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
pc->totalTime-0.1 :
pc->seekWhere;
dc->seekWhere = 0 > dc->seekWhere ? 0 : dc->seekWhere;
- cb->begin = 0;
+ dc->seekError = 0;
dc->seek = 1;
- pc->elapsedTime = dc->seekWhere;
pc->bitRate = 0;
while(*decode_pid>0 && dc->seek) usleep(1000);
+ if(dc->seekError) pc->elapsedTime = dc->seekWhere;
}
}
pc->seek = 0;
@@ -229,6 +230,9 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af,
break;
#endif
#ifdef HAVE_FAAD
+ case DECODE_TYPE_AAC:
+ dc->error = aac_decode(cb,af,dc);
+ break;
case DECODE_TYPE_MP4:
dc->error = mp4_decode(cb,af,dc);
break;
diff --git a/src/decode.h b/src/decode.h
index 9f6e2f0e5..faf18f344 100644
--- a/src/decode.h
+++ b/src/decode.h
@@ -21,6 +21,8 @@
#include "../config.h"
+#include "mpd_types.h"
+
#include <stdio.h>
#include <sys/param.h>
@@ -38,11 +40,12 @@
#define DECODE_ERROR_UNKTYPE 1
typedef struct _DecoderControl {
- int state;
- int stop;
- int start;
- int error;
- int seek;
+ mpd_sint8 state;
+ mpd_sint8 stop;
+ mpd_sint8 start;
+ mpd_uint16 error;
+ mpd_sint8 seek;
+ mpd_sint8 seekError;
double seekWhere;
char file[MAXPATHLEN+1];
} DecoderControl;
diff --git a/src/flac_decode.c b/src/flac_decode.c
index a684e5091..29f7dbdaa 100644
--- a/src/flac_decode.c
+++ b/src/flac_decode.c
@@ -104,7 +104,7 @@ void flacPlayFile(char *file, Buffer * cb, AudioFormat * af,
if(dc->seek) {
FLAC__uint64 sampleToSeek = dc->seekWhere*
af->sampleRate+0.5;
- cb->end = 0;
+ cb->end = cb->begin;
cb->wrap = 0;
if(FLAC__file_decoder_seek_absolute(flacDec,
sampleToSeek))
diff --git a/src/mp3_decode.c b/src/mp3_decode.c
index a07485a32..6fa931fe7 100644
--- a/src/mp3_decode.c
+++ b/src/mp3_decode.c
@@ -487,7 +487,7 @@ int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
if(dc->seek) {
long i = 0;
cb->wrap = 0;
- cb->end = 0;
+ cb->end = cb->begin;
data->muteFrame = 1;
while(i<data->highestFrame && dc->seekWhere >
((float)mad_timer_count(data->times[i],
diff --git a/src/mp4_decode.c b/src/mp4_decode.c
index a2cfc2ee0..d3c16b73e 100644
--- a/src/mp4_decode.c
+++ b/src/mp4_decode.c
@@ -2,8 +2,6 @@
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- *
* 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
@@ -149,7 +147,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
< 0)
{
- ERROR("Error initializing AAC decoder library.\n");
+ ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
@@ -210,7 +208,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
if(dc->seek && seekPositionFound) {
seekPositionFound = 0;
chunkLen = 0;
- cb->end = 0;
+ cb->end = cb->begin;
cb->wrap = 0;
dc->seek = 0;
}
@@ -233,7 +231,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
}
if(channels*(dur+offset) > frameInfo.samples) {
- dur = frameInfo.samples;
+ dur = frameInfo.samples/channels;
offset = 0;
}
diff --git a/src/mp4_decode.h b/src/mp4_decode.h
index 58d40aeff..a981e1e0a 100644
--- a/src/mp4_decode.h
+++ b/src/mp4_decode.h
@@ -2,8 +2,6 @@
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- *
* 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
diff --git a/src/ogg_decode.c b/src/ogg_decode.c
index 7fe3dfeb6..ff8d2024c 100644
--- a/src/ogg_decode.c
+++ b/src/ogg_decode.c
@@ -95,7 +95,7 @@ int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
while(!eof) {
if(dc->seek) {
- cb->end = 0;
+ cb->end = cb->begin;
cb->wrap = 0;
chunkpos = 0;
ov_time_seek_page(&vf,dc->seekWhere);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-16 08:52:54 +0000