initial mp4/aac decoder, hasn't been tested at all yet, just compiles

git-svn-id: https://svn.musicpd.org/mpd/trunk@275 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 288 insertions, 0 deletions

diff --git a/src/mp4_decode.c b/src/mp4_decode.c
new file mode 100644
index 000000000..a96cc3536
--- /dev/null
+++ b/src/mp4_decode.c
@@ -0,0 +1,288 @@
+/* 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
+ * 
+ * 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
+ * (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 "mp4_decode.h"
+
+#ifdef HAVE_FAAD
+
+#include "command.h"
+#include "utils.h"
+#include "audio.h"
+#include "log.h"
+#include "pcm_utils.h"
+
+#include "mp4ff/mp4ff.h"
+
+#include <stdio.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <string.h>
+#include <faad.h>
+
+int mp4_getAACTrack(mp4ff_t *infile) {
+	/* find AAC track */
+	int i, rc;
+	int numTracks = mp4ff_total_tracks(infile);
+
+	for (i = 0; i < numTracks; i++) {
+		unsigned char *buff = NULL;
+		int buff_size = 0;
+		mp4AudioSpecificConfig mp4ASC;
+	
+		mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+		if (buff) {
+			rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+			free(buff);
+			if (rc < 0) continue;
+            		return i;
+		}
+	}
+
+	/* can't decode this */
+	return -1;
+}
+
+uint32_t mp4_readCallback(void *user_data, void *buffer, uint32_t length) {
+	return fread(buffer, 1, length, (FILE*)user_data);
+}
+            
+uint32_t mp4_seekCallback(void *user_data, uint64_t position) {
+	return fseek((FILE*)user_data, position, SEEK_SET);
+}       
+		    
+
+int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
+{
+	FILE * fh;
+	mp4ff_t * mp4fh;
+	mp4ff_callback_t * mp4cb; 
+	int32_t track;
+	int32_t time;
+	int32_t scale;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	mp4AudioSpecificConfig mp4ASC;
+	unsigned char * mp4Buffer;
+	int mp4BufferSize;
+	unsigned int frameSize;
+	unsigned int useAacLength;
+	unsigned long sampleRate;
+	unsigned char channels;
+	long sampleId;
+	long numSamples;
+
+	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;
+	}
+
+	track = mp4_getAACTrack(mp4fh);
+	if(track < 0) {
+		ERROR("No AAC track found in mp4 stream.\n");
+		mp4ff_close(mp4fh);
+		fclose(fh);
+		free(mp4cb);
+		return -1;
+	}
+
+	decoder = faacDecOpen();
+
+	config = faacDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	config->downMatrix = 1;
+	config->dontUpSampleImplicitSBR = 1;
+	faacDecSetConfiguration(decoder,config);
+
+	af->bits = 16;
+
+	mp4Buffer = NULL;
+	mp4BufferSize = 0;
+	mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
+
+	if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
+			< 0)
+	{
+		ERROR("Error initializing AAC decoder library.\n");
+		faacDecClose(decoder);
+		mp4ff_close(mp4fh);
+		free(mp4cb);
+		fclose(fh);
+		return -1;
+	}
+
+	af->sampleRate = sampleRate;
+	af->channels = channels;
+	time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
+	scale = mp4ff_time_scale(mp4fh,track);
+	frameSize = 1024;
+	useAacLength = 0;
+
+	if(mp4Buffer) {
+		if(AudioSpecificConfig(mp4Buffer,mp4BufferSize,&mp4ASC) >= 0) {
+			if(mp4ASC.frameLengthFlag==1) frameSize = 960;
+			if(mp4ASC.sbr_present_flag==1) frameSize*= 2;
+		}
+		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);
+
+	dc->state = DECODE_STATE_DECODE;
+	dc->start = 0;
+	{
+		int eof = 0;
+		int rc;
+		long dur;
+		unsigned int sampleCount;
+		unsigned int delay = 0;
+		char * sampleBuffer;
+		unsigned int initial = 1;
+		size_t sampleBufferLen;
+
+		for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
+			if(dc->seek) {
+				cb->end = 0;
+				cb->wrap = 0;
+//#warning implement seeking here!
+				dc->seek = 0;
+			}
+
+			dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
+			rc = mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
+					&mp4BufferSize);
+
+			if(rc==0) eof = 1;
+			else {
+				sampleBuffer = faacDecDecode(decoder,
+								&frameInfo,
+								mp4Buffer,
+								mp4BufferSize);
+				if(mp4Buffer) free(mp4Buffer);
+				if(sampleId==0) dur = 0;
+				if(useAacLength || scale!=sampleRate) {
+					sampleCount = frameInfo.samples;
+				}
+				else {
+					sampleCount = (unsigned long)(dur * 
+							frameInfo.channels);
+					if(!useAacLength && !initial && 
+						(sampleId < numSamples/2) &&
+						(sampleCount!=
+						frameInfo.samples))
+					{
+						useAacLength = 1;
+						sampleCount = frameInfo.samples;
+					}
+					
+					if(initial && (sampleCount < frameSize*
+						frameInfo.channels) &&
+						(frameInfo.samples > 
+						sampleCount))
+					{
+						delay = frameInfo.samples -
+							sampleCount;
+					}
+
+				}
+
+				if(sampleCount>0) initial =0;
+				sampleBufferLen = sampleCount*2;
+				sampleBuffer+=delay*2;
+				while(sampleBufferLen > 0) {
+					size_t size = sampleBufferLen>
+							CHUNK_SIZE?
+							CHUNK_SIZE:
+							sampleBufferLen;
+					while(cb->begin==cb->end && cb->wrap &&
+							!dc->stop && !dc->seek)
+					{
+						usleep(10000);
+					}
+					if(dc->stop) {
+						eof = 1;
+						break;
+					}
+					else if(dc->seek) break;
+				
+#ifdef WORDS_BIGENDIAN
+					pcm_changeBufferEndianness(sampleBuffer,
+							size,af->bits);
+#endif
+					memcpy(cb->chunks+cb->end*CHUNK_SIZE,
+						sampleBuffer,size);
+					cb->chunkSize[cb->end] = size;
+			
+//#warning implement time for AAC	
+					cb->times[cb->end] = 0;
+				
+					++cb->end;
+				
+					if(cb->end>=buffered_chunks) {
+						cb->end = 0;
+						cb->wrap = 1;
+					}
+				}
+			}
+		}
+
+		if(dc->seek) dc->seek = 0;
+
+		if(dc->stop) {
+			dc->state = DECODE_STATE_STOP;
+			dc->stop = 0;
+		}
+		else dc->state = DECODE_STATE_STOP;
+	}
+
+	faacDecClose(decoder);
+	mp4ff_close(mp4fh);
+	fclose(fh);
+	free(mp4cb);
+
+	return 0;
+}
+
+#endif /* HAVE_FAAD */