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-rw-r--r--src/inputPlugins/aac_plugin.c438
1 files changed, 438 insertions, 0 deletions
diff --git a/src/inputPlugins/aac_plugin.c b/src/inputPlugins/aac_plugin.c
new file mode 100644
index 000000000..0dd23f955
--- /dev/null
+++ b/src/inputPlugins/aac_plugin.c
@@ -0,0 +1,438 @@
+/* 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
+ */
+
+#include "../inputPlugin.h"
+
+#ifdef HAVE_FAAD
+
+#define AAC_MAX_CHANNELS 6
+
+#include "../utils.h"
+#include "../audio.h"
+#include "../log.h"
+#include "../inputStream.h"
+#include "../outputBuffer.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <faad.h>
+
+/* all code here is either based on or copied from FAAD2's frontend code */
+typedef struct {
+ InputStream * inStream;
+ long bytesIntoBuffer;
+ long bytesConsumed;
+ long fileOffset;
+ unsigned char *buffer;
+ int atEof;
+} AacBuffer;
+
+void fillAacBuffer(AacBuffer *b) {
+ if(b->bytesConsumed > 0) {
+ int bread;
+
+ if(b->bytesIntoBuffer) {
+ memmove((void *)b->buffer,(void*)(b->buffer+
+ b->bytesConsumed),b->bytesIntoBuffer);
+ }
+
+ if(!b->atEof) {
+ bread = readFromInputStream(b->inStream,
+ (void *)(b->buffer+b->bytesIntoBuffer),
+ 1,b->bytesConsumed);
+ if(bread!=b->bytesConsumed) b->atEof = 1;
+ b->bytesIntoBuffer+=bread;
+ }
+
+ b->bytesConsumed = 0;
+
+ if(b->bytesIntoBuffer > 3) {
+ if(memcmp(b->buffer,"TAG",3)==0) b->bytesIntoBuffer = 0;
+ }
+ if(b->bytesIntoBuffer > 11) {
+ if(memcmp(b->buffer,"LYRICSBEGIN",11)==0) {
+ b->bytesIntoBuffer = 0;
+ }
+ }
+ if(b->bytesIntoBuffer > 8) {
+ if(memcmp(b->buffer,"APETAGEX",8)==0) {
+ b->bytesIntoBuffer = 0;
+ }
+ }
+ }
+}
+
+void advanceAacBuffer(AacBuffer * b, int bytes) {
+ b->fileOffset+=bytes;
+ b->bytesConsumed = bytes;
+ b->bytesIntoBuffer-=bytes;
+}
+
+static int adtsSampleRates[] = {96000,88200,64000,48000,44100,32000,24000,22050,
+ 16000,12000,11025,8000,7350,0,0,0};
+
+int adtsParse(AacBuffer * b, float * length) {
+ int frames, frameLength;
+ int tFrameLength = 0;
+ int sampleRate = 0;
+ float framesPerSec, bytesPerFrame;
+
+ /* Read all frames to ensure correct time and bitrate */
+ for(frames = 0; ;frames++) {
+ fillAacBuffer(b);
+
+ if(b->bytesIntoBuffer > 7) {
+ /* check syncword */
+ if (!((b->buffer[0] == 0xFF) &&
+ ((b->buffer[1] & 0xF6) == 0xF0)))
+ {
+ break;
+ }
+
+ if(frames==0) {
+ sampleRate = adtsSampleRates[
+ (b->buffer[2]&0x3c)>>2];
+ }
+
+ frameLength = ((((unsigned int)b->buffer[3] & 0x3))
+ << 11) | (((unsigned int)b->buffer[4])
+ << 3) | (b->buffer[5] >> 5);
+
+ tFrameLength+=frameLength;
+
+ if(frameLength > b->bytesIntoBuffer) break;
+
+ advanceAacBuffer(b,frameLength);
+ }
+ else break;
+ }
+
+ framesPerSec = (float)sampleRate/1024.0;
+ if(frames!=0) {
+ bytesPerFrame = (float)tFrameLength/(float)(frames*1000);
+ }
+ else bytesPerFrame = 0;
+ if(framesPerSec!=0) *length = (float)frames/framesPerSec;
+
+ return 1;
+}
+
+void initAacBuffer(InputStream * inStream, AacBuffer * b, float * length,
+ size_t * retFileread, size_t * retTagsize)
+{
+ size_t fileread;
+ size_t bread;
+ size_t tagsize;
+
+ if(length) *length = -1;
+
+ memset(b,0,sizeof(AacBuffer));
+
+ b->inStream = inStream;
+
+ fileread = inStream->size;
+
+ b->buffer = malloc(FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+ memset(b->buffer,0,FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+
+ bread = readFromInputStream(inStream,b->buffer,1,
+ FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+ b->bytesIntoBuffer = bread;
+ b->bytesConsumed = 0;
+ b->fileOffset = 0;
+
+ if(bread!=FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
+
+ tagsize = 0;
+ if(!memcmp(b->buffer,"ID3",3)) {
+ tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
+ (b->buffer[8] << 7) | (b->buffer[9] << 0);
+
+ tagsize+=10;
+ advanceAacBuffer(b,tagsize);
+ fillAacBuffer(b);
+ }
+
+ if(retFileread) *retFileread = fileread;
+ if(retTagsize) *retTagsize = tagsize;
+
+ if(length==NULL) return;
+
+ if((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
+ adtsParse(b, length);
+ seekInputStream(b->inStream, tagsize, SEEK_SET);
+
+ bread = readFromInputStream(b->inStream, b->buffer, 1,
+ FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
+ if(bread != FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
+ else b->atEof = 0;
+ b->bytesIntoBuffer = bread;
+ b->bytesConsumed = 0;
+ b->fileOffset = tagsize;
+ }
+ else if(memcmp(b->buffer,"ADIF",4) == 0) {
+ int bitRate;
+ int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
+ bitRate = ((unsigned int)(b->buffer[4 + skipSize] & 0x0F)<<19) |
+ ((unsigned int)b->buffer[5 + skipSize]<<11) |
+ ((unsigned int)b->buffer[6 + skipSize]<<3) |
+ ((unsigned int)b->buffer[7 + skipSize] & 0xE0);
+
+ *length = fileread;
+ if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
+ }
+}
+
+float getAacFloatTotalTime(char * file) {
+ AacBuffer b;
+ float length;
+ size_t fileread, tagsize;
+ faacDecHandle decoder;
+ faacDecConfigurationPtr config;
+ unsigned long sampleRate;
+ unsigned char channels;
+ InputStream inStream;
+ size_t bread;
+
+ if(openInputStream(&inStream,file) < 0) return -1;
+
+ initAacBuffer(&inStream,&b,&length,&fileread,&tagsize);
+
+ if(length < 0) {
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+ faacDecSetConfiguration(decoder,config);
+
+ fillAacBuffer(&b);
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+ &sampleRate,&channels);
+#else
+ bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels);
+#endif
+ if(bread >= 0 && sampleRate > 0 && channels > 0) length = 0;
+
+ faacDecClose(decoder);
+ }
+
+ if(b.buffer) free(b.buffer);
+ closeInputStream(&inStream);
+
+ return length;
+}
+
+int getAacTotalTime(char * file) {
+ int time = -1;
+ float length;
+
+ if((length = getAacFloatTotalTime(file))>=0) time = length+0.5;
+
+ return time;
+}
+
+
+int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
+ float time;
+ float totalTime;
+ faacDecHandle decoder;
+ faacDecFrameInfo frameInfo;
+ faacDecConfigurationPtr config;
+ size_t bread;
+ unsigned long sampleRate;
+ unsigned char channels;
+ int eof = 0;
+ unsigned int sampleCount;
+ char * sampleBuffer;
+ size_t sampleBufferLen;
+ /*float * seekTable;
+ long seekTableEnd = -1;
+ int seekPositionFound = 0;*/
+ mpd_uint16 bitRate = 0;
+ AacBuffer b;
+ InputStream inStream;
+
+ if((totalTime = getAacFloatTotalTime(dc->file)) < 0) return -1;
+
+ if(openInputStream(&inStream,dc->file) < 0) return -1;
+
+ initAacBuffer(&inStream,&b,NULL,NULL,NULL);
+
+ decoder = faacDecOpen();
+
+ config = faacDecGetCurrentConfiguration(decoder);
+ config->outputFormat = FAAD_FMT_16BIT;
+#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
+ config->downMatrix = 1;
+#endif
+#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
+ config->dontUpSampleImplicitSBR = 0;
+#endif
+ faacDecSetConfiguration(decoder,config);
+
+ fillAacBuffer(&b);
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+ &sampleRate,&channels);
+#else
+ bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels);
+#endif
+ if(bread < 0) {
+ ERROR("Error not a AAC stream.\n");
+ faacDecClose(decoder);
+ closeInputStream(b.inStream);
+ if(b.buffer) free(b.buffer);
+ return -1;
+ }
+
+ dc->audioFormat.bits = 16;
+
+ dc->totalTime = totalTime;
+
+ time = 0.0;
+
+ advanceAacBuffer(&b,bread);
+
+ while(!eof) {
+ fillAacBuffer(&b);
+
+ if(b.bytesIntoBuffer==0) {
+ eof = 1;
+ break;
+ }
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+ sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer,
+ b.bytesIntoBuffer);
+#else
+ sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer);
+#endif
+
+ if(frameInfo.error > 0) {
+ ERROR("error decoding AAC file: %s\n",dc->file);
+ ERROR("faad2 error: %s\n",
+ faacDecGetErrorMessage(frameInfo.error));
+ eof = 1;
+ break;
+ }
+
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+ sampleRate = frameInfo.samplerate;
+#endif
+
+ if(dc->state != DECODE_STATE_DECODE) {
+ dc->audioFormat.channels = frameInfo.channels;
+ dc->audioFormat.sampleRate = sampleRate;
+ getOutputAudioFormat(&(dc->audioFormat),
+ &(cb->audioFormat));
+ dc->state = DECODE_STATE_DECODE;
+ }
+
+ advanceAacBuffer(&b,frameInfo.bytesconsumed);
+
+ sampleCount = (unsigned long)(frameInfo.samples);
+
+ if(sampleCount>0) {
+ bitRate = frameInfo.bytesconsumed*8.0*
+ frameInfo.channels*sampleRate/
+ frameInfo.samples/1000+0.5;
+ time+= (float)(frameInfo.samples)/frameInfo.channels/
+ sampleRate;
+ }
+
+ sampleBufferLen = sampleCount*2;
+
+ sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer,
+ sampleBufferLen, time, bitRate);
+ if(dc->seek) {
+ dc->seekError = 1;
+ dc->seek = 0;
+ }
+ else if(dc->stop) {
+ eof = 1;
+ break;
+ }
+ } while (!eof);
+
+ flushOutputBuffer(cb);
+
+ faacDecClose(decoder);
+ closeInputStream(b.inStream);
+ if(b.buffer) free(b.buffer);
+
+ if(dc->state != DECODE_STATE_DECODE) return -1;
+
+ if(dc->seek) {
+ dc->seekError = 1;
+ dc->seek = 0;
+ }
+
+ if(dc->stop) {
+ dc->state = DECODE_STATE_STOP;
+ dc->stop = 0;
+ }
+ else dc->state = DECODE_STATE_STOP;
+
+ return 0;
+}
+
+MpdTag * aacTagDup(char * file) {
+ MpdTag * ret = NULL;
+ int time;
+
+ time = getAacTotalTime(file);
+
+ if(time>=0) {
+ if((ret = id3Dup(file))==NULL) ret = newMpdTag();
+ ret->time = time;
+ }
+
+ return ret;
+}
+
+char * aacSuffixes[] = {"aac", NULL};
+
+InputPlugin aacPlugin =
+{
+ "aac",
+ NULL,
+ aac_decode,
+ aacTagDup,
+ INPUT_PLUGIN_STREAM_FILE,
+ aacSuffixes,
+ NULL
+};
+
+#else
+
+InputPlugin aacPlugin =
+{
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ 0,
+ NULL,
+ NULL,
+};
+
+#endif /* HAVE_FAAD */
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