mp4_plugin

git-svn-id: https://svn.musicpd.org/mpd/trunk@1249 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 426 insertions, 0 deletions

diff --git a/src/inputPlugins/mp4_plugin.c b/src/inputPlugins/mp4_plugin.c
new file mode 100644
index 000000000..b34b3c421
--- /dev/null
+++ b/src/inputPlugins/mp4_plugin.c
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+/* 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
+
+#include "../utils.h"
+#include "../audio.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+#include "../inputStream.h"
+#include "../outputBuffer.h"
+#include "../decode.h"
+
+#include "../mp4ff/mp4ff.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 */
+
+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;
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+		mp4AudioSpecificConfig mp4ASC;
+#else
+		unsigned long dummy1_32;
+            	unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+                		dummy7_8, dummy8_8;
+#endif
+	
+		mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
+
+		if (buff) {
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
+			rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+#else
+			rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
+					&dummy3_8, &dummy4_8, &dummy5_8, 
+					&dummy6_8, &dummy7_8, &dummy8_8);
+#endif
+			free(buff);
+			if (rc < 0) continue;
+            		return i;
+		}
+	}
+
+	/* can't decode this */
+	return -1;
+}
+
+uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer, 
+		uint32_t length) 
+{
+	return readFromInputStream((InputStream*) inStream, buffer, 1, length);
+}
+            
+uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position) {
+	return seekInputStream((InputStream *) inStream, position, SEEK_SET);
+}       
+		    
+
+int mp4_decode(OutputBuffer * cb, DecoderControl * dc) {
+	mp4ff_t * mp4fh;
+	mp4ff_callback_t * mp4cb; 
+	int32_t track;
+	float time;
+	int32_t scale;
+	faacDecHandle decoder;
+	faacDecFrameInfo frameInfo;
+	faacDecConfigurationPtr config;
+	unsigned char * mp4Buffer;
+	int mp4BufferSize;
+	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;
+	float * seekTable;
+	long seekTableEnd = -1;
+	int seekPositionFound = 0;
+	long offset;
+	mpd_uint16 bitRate = 0;
+	InputStream inStream;
+
+	if(openInputStream(&inStream,dc->file) < 0) {
+		ERROR("failed to open %s\n",dc->file);
+		return -1;
+	}
+
+	mp4cb = malloc(sizeof(mp4ff_callback_t));
+	mp4cb->read = mp4_inputStreamReadCallback;
+	mp4cb->seek = mp4_inputStreamSeekCallback;
+	mp4cb->user_data = &inStream;
+
+	mp4fh = mp4ff_open_read(mp4cb);
+	if(!mp4fh) {
+		ERROR("Input does not appear to be a mp4 stream.\n");
+		free(mp4cb);
+		closeInputStream(&inStream);
+		return -1;
+	}
+
+	track = mp4_getAACTrack(mp4fh);
+	if(track < 0) {
+		ERROR("No AAC track found in mp4 stream.\n");
+		mp4ff_close(mp4fh);
+		closeInputStream(&inStream);
+		free(mp4cb);
+		return -1;
+	}
+
+	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);
+
+	dc->audioFormat.bits = 16;
+
+	mp4Buffer = NULL;
+	mp4BufferSize = 0;
+	mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
+
+	if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
+			< 0)
+	{
+		ERROR("Error not a AAC stream.\n");
+		faacDecClose(decoder);
+		mp4ff_close(mp4fh);
+		free(mp4cb);
+		closeInputStream(&inStream);
+		return -1;
+	}
+
+	dc->audioFormat.sampleRate = sampleRate;
+	dc->audioFormat.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);
+		closeInputStream(&inStream);
+		free(mp4cb);
+		return -1;
+	}
+	dc->totalTime = ((float)time)/scale;
+
+	numSamples = mp4ff_num_samples(mp4fh,track);
+
+	time = 0.0;
+
+	seekTable = malloc(sizeof(float)*numSamples);
+
+	for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
+		if(dc->seek && seekTableEnd>1 && 
+				seekTable[seekTableEnd]>=dc->seekWhere)
+		{
+			int i = 2;
+			while(seekTable[i]<dc->seekWhere) i++;
+			sampleId = i-1;
+			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;
+		}
+
+		if(sampleId==0) dur = 0;
+		if(offset>dur) dur = 0;
+		else dur-=offset;
+		time+=((float)dur)/scale;
+
+		if(dc->seek && time>dc->seekWhere) seekPositionFound = 1;
+
+		if(dc->seek && seekPositionFound) {
+			seekPositionFound = 0;
+                        clearOutputBuffer(cb);
+			dc->seek = 0;
+		}
+
+		if(dc->seek) continue;
+		
+		if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
+				&mp4BufferSize) == 0)
+		{
+			eof = 1;
+			continue;
+		}
+
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
+		sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
+						mp4BufferSize);
+#else
+		sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer);
+#endif
+
+		if(mp4Buffer) free(mp4Buffer);
+		if(frameInfo.error > 0) {
+			ERROR("error decoding MP4 file: %s\n",dc->file);
+			ERROR("faad2 error: %s\n",
+				faacDecGetErrorMessage(frameInfo.error));
+			eof = 1;
+			break;
+		}
+
+		if(dc->state != DECODE_STATE_DECODE) {
+			channels = frameInfo.channels;
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
+			scale = frameInfo.samplerate;
+#endif
+			dc->audioFormat.sampleRate = scale;
+			dc->audioFormat.channels = frameInfo.channels;
+                        getOutputAudioFormat(&(dc->audioFormat),
+                                        &(cb->audioFormat));
+			dc->state = DECODE_STATE_DECODE;
+		}
+
+		if(channels*(dur+offset) > frameInfo.samples) {
+			dur = frameInfo.samples/channels;
+			offset = 0;
+		}
+
+		sampleCount = (unsigned long)(dur*channels);
+
+		if(sampleCount>0) {
+			initial =0;
+			bitRate = frameInfo.bytesconsumed*8.0*
+				frameInfo.channels*scale/
+				frameInfo.samples/1000+0.5;
+		}
+			
+
+		sampleBufferLen = sampleCount*2;
+
+		sampleBuffer+=offset*channels*2;
+
+		sendDataToOutputBuffer(cb, NULL, dc, 1, sampleBuffer,
+				sampleBufferLen, time, bitRate);
+		if(dc->stop) {
+			eof = 1;
+			break;
+		}
+	}
+
+	flushOutputBuffer(cb);
+
+	free(seekTable);
+	faacDecClose(decoder);
+	mp4ff_close(mp4fh);
+	closeInputStream(&inStream);
+	free(mp4cb);
+
+	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 * mp4DataDup(char * file, int * mp4MetadataFound) {
+	MpdTag * ret = NULL;
+	InputStream inStream;
+	mp4ff_t * mp4fh;
+	mp4ff_callback_t * cb; 
+	int32_t track;
+	int32_t time;
+	int32_t scale;
+
+	*mp4MetadataFound = 0;
+	
+	if(openInputStream(file) < 0) return NULL;
+
+	cb = malloc(sizeof(mp4ff_callback_t));
+	cb->read = mp4_inputStreamReadCallback;
+	cb->seek = mp4_inputStreamSeekCallback;
+	cb->user_data = &inStream;
+
+	mp4fh = mp4ff_open_read(cb);
+	if(!mp4fh) {
+		free(cb);
+		closeInputStream(&inStream);
+		return NULL;
+	}
+
+	track = mp4_getAACTrack(mp4fh);
+	if(track < 0) {
+		mp4ff_close(mp4fh);
+		closeInputStream(&inStream);
+		free(cb);
+		return NULL;
+	}
+
+	ret = newMpdTag();
+	time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
+	scale = mp4ff_time_scale(mp4fh,track);
+	if(scale < 0) {
+		mp4ff_close(mp4fh);
+		closeInputStream(&inStream);
+		free(cb);
+		freeMpdTag(ret);
+		return NULL;
+	}
+	ret->time = ((float)time)/scale+0.5;
+
+	if(!mp4ff_meta_get_artist(mp4fh,&ret->artist)) {
+		*mp4MetadataFound = 1;
+	}
+
+	if(!mp4ff_meta_get_album(mp4fh,&ret->album)) {
+		*mp4MetadataFound = 1;
+	}
+
+	if(!mp4ff_meta_get_title(mp4fh,&ret->title)) {
+		*mp4MetadataFound = 1;
+	}
+
+	if(!mp4ff_meta_get_track(mp4fh,&ret->track)) {
+		*mp4MetadataFound = 1;
+	}
+
+	mp4ff_close(mp4fh);
+	closeInputStream(&inStream);
+	free(cb);
+
+	return ret;
+}
+
+MpdTag * mp4TagDup(char * file) {
+	MpdTag * ret = NULL;
+	int mp4MetadataFound = 0;
+
+	ret = mp4DataDup(file, &mp4MetadataFound);
+        if(!ret) return NULL;
+	if(!mp4MetadataFound) {
+		MpdTag * temp = id3Dup(file);
+		if(temp) {
+			temp->time = ret->time;
+			freeMpdTag(ret);
+			ret = temp;
+		}
+	}
+
+	return ret;
+}
+
+char * mp4Suffixes[] = {"m4a", "mp4", NULL};
+
+InputPlugin mp4Plugin =
+{
+        "mp4",
+        NULL,
+        mp4_decode,
+        mp4TagDup,
+        INPUT_PLUGIN_STREAM_FILE,
+        mp4Suffixes,
+        NULL
+};
+
+#else
+
+InputPlugin mp4Plugin =
+{
+        NULL,
+        NULL,
+        NULL,
+        NULL,
+        0,
+        NULL,
+        NULL
+};
+
+#endif /* HAVE_FAAD */