aac_plugin

git-svn-id: https://svn.musicpd.org/mpd/trunk@1250 09075e82-0dd4-0310-85a5-a0d7c8717e4f

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 */