Initial commit.

Basically the old code from 2006/07, slightly
modified to compile on modern GCC/Linux.
Changed the tool names.

5 files changed, 429 insertions, 0 deletions

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000..3c6e429
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,2 @@
+ac32spdif
+dts2spdif
diff --git a/Makefile b/Makefile
new file mode 100644
index 0000000..bf64657
--- /dev/null
+++ b/Makefile
@@ -0,0 +1,11 @@
+all: ac32spdif dts2spdif
+
+ac32spdif: ac32spdif.c
+	cc -O2 -o ac32spdif ac32spdif.c
+
+dts2spdif: dts2spdif.c
+	cc -O2 -o dts2spdif dts2spdif.c
+
+.PHONY: clean
+clean:
+	rm -f ac32spdif dts2spdif
diff --git a/README.md b/README.md
new file mode 100644
index 0000000..ed9a663
--- /dev/null
+++ b/README.md
@@ -0,0 +1,57 @@
+S/PDIF tools
+===========
+
+These simple programs convert raw AC3 (Dolby Digital, A/52) and DTS streams
+into WAV files containing the S/PDIF fake PCM encoding thereof. This means
+that if such a .wav file is replayed correctly bit-by-bit via the S/PDIF
+output of a device and supplied to a standards compliant decoder, it
+will be recognized as a standards compliant AC3/DTS stream,
+indistinguishable from a stream generated e.g. by a DVD player.
+
+This is useful to e.g. author CDs containing compressed multichannel
+data (such as DTS encoded CDs that commercially available).
+Take care: Playing back the files produced via speakers at high volume
+may damage these or your hearing.
+
+
+A note about coding style
+-------------------------
+
+These sources are very old (from around 2006/2007) and show my coding
+abilities back then when I started with C. I hope to clean this up
+and modernize it at some point in the future.
+
+
+ac32spdif
+---------
+
+This tool takes a raw .ac3 file and converts it into a .wav file of the
+same sample rate containing one AC3 frame plus padding zeros for the
+remaining time of that frame's time slot.
+
+
+dts2spdif
+---------
+
+In a similar fashion, this tool translates a .dts file to a .wav file.
+Currently it only supports 16 bit encapsulation, i.e. the data stream
+is included in the .wav file 1:1.
+The DTS specification also defines a 14 bit format for use with CDs
+where there is a possibility that the data will be played back via
+speakers in order to reduce the possibility of speaker/hearing damage
+when the medium is played back without a DTS decoder. In this case,
+only the lower 14 bits of a word would be used, with the high 2 bits
+set to zero. This encoding may be added to dts2spdif later on.
+
+
+License
+-------
+
+As usual, this code comes with no warranty whatsoever. I am especially
+not responsible if you blow up your speakers or ears using these tools.
+
+Do whatever you like with this code, but if you use it for your own
+application I would appreciate an attribution note in the documentation.
+Thanks!
+
+ -- norly
diff --git a/ac32spdif.c b/ac32spdif.c
new file mode 100755
index 0000000..a1ab79f
--- /dev/null
+++ b/ac32spdif.c
@@ -0,0 +1,182 @@
+

+#include <stdio.h>

+//#include <stdlib.h>

+//#include <string.h>

+#include <memory.h>

+

+//#include "fs_codes.h"

+

+

+short bitrates[] = { 32,  40,  48,  56,  64,  80,  96, 112, 128, 160,

+192, 224, 256, 320, 384, 448, 512, 576, 640 };

+

+short fsize44[] = {  70,   88,  105,  122,  140,  175,  209,  244,  279,  349,

+418,  488,  558,  697,  836,  976, 1115, 1254, 1394 };

+

+unsigned long DecFS(char sratecode, unsigned long fsizecode)

+{

+	switch (sratecode)

+	{

+		case 0:

+			return (bitrates[fsizecode / 2] * 2);

+		case 1:

+			return ((fsizecode % 2) ? fsize44[fsizecode / 2] : fsize44[fsizecode / 2] - 1);

+		case 2:

+			return (bitrates[fsizecode / 2] * 3);

+		default:

+			return 0;

+	}

+

+	return 0;

+}

+

+

+

+

+struct WAVEHEADER {

+    unsigned long  ChunkID;

+    unsigned long  ChunkSize;

+    unsigned long  Format;

+    unsigned long  SubChunk1ID;

+    unsigned long  SubChunk1Size;

+    unsigned short AudioFormat;

+    unsigned short NumChannels;

+    unsigned long  SampleRate;

+    unsigned long  ByteRate;

+    unsigned short BlockAlign;

+    unsigned short BitsPerSample;

+    unsigned long  SubChunk2ID;

+    unsigned long  SubChunk2Size;

+};

+

+

+struct AC3PREAMBLE {

+	unsigned short sync1;

+	unsigned short sync2;

+	unsigned char burst_infoLSB;

+	unsigned char burst_infoMSB;

+	unsigned short lengthcode;

+};

+

+char burst[6144];

+

+FILE *infile, *outfile;

+

+unsigned long bytesread;

+unsigned long payloadbytes;

+char sampleratecode;

+unsigned long framesizecode;

+

+struct AC3PREAMBLE ac3p = { 0xF872, 0x4E1F, 1, 0, 0x3800 };

+struct WAVEHEADER wavhdr = { 0x46464952,

+							0,

+							0x45564157,

+							0x20746D66,

+							16,

+							1,

+							2,

+							0,

+							0,

+							4,

+							16,

+							0x61746164,

+							0 };

+

+unsigned long i;

+char temp;

+

+int main( int argc, char *argv[ ], char *envp[ ] )

+{

+	if (argc < 3)

+	{

+		printf("Wrong syntax. AC3PACK <In.ac3> <Out.wav>.\n");

+		return 0;

+	}

+

+	infile = fopen(argv[1], "rb");

+	outfile = fopen(argv[2], "wb");

+	//fseek(outfile, SEEK_SET, 44);

+	fwrite (&wavhdr, sizeof(struct WAVEHEADER), 1, outfile);

+

+

+

+	for(;;)

+	{

+		memset (burst, 0, 6144);

+		bytesread = fread(&burst[8], 1, 6, infile);

+		if (bytesread < 6)

+		{

+			printf ("EOF reached (Frame Header reading)!\nCurrent position in INFILE: %i\n", ftell(infile));

+			break;

+		}

+		if ((burst[8] != 0x0B) || (burst[9] != 0x77))

+		{

+			printf("ERROR: INVALID SYNCWORD !\nCurrent position in INFILE: %i\n", ftell(infile));

+			break;

+		}

+		framesizecode = (burst[12] & 63);

+		sampleratecode = ((burst[12] & 192) / 64);

+

+		if (wavhdr.SampleRate == 0)

+		{

+				printf ("First Sampleratecode: %i\n", sampleratecode);

+

+				switch (sampleratecode)

+				{

+				case 0:

+					wavhdr.SampleRate = 48000;

+					wavhdr.ByteRate = 192000;

+					break;

+				case 1:

+					wavhdr.SampleRate = 44100;

+					wavhdr.ByteRate = 176000;

+					break;

+				case 2:

+					wavhdr.SampleRate = 32000;

+					wavhdr.ByteRate = 128000;

+					break;

+				default:

+					wavhdr.SampleRate = 48000;

+					wavhdr.ByteRate = 192000;

+				}

+		}

+

+		payloadbytes = DecFS (sampleratecode, framesizecode);

+		payloadbytes *= 2;

+

+		bytesread = fread (&burst[14], 1, payloadbytes - 6, infile);

+		if ((bytesread + 6) < payloadbytes)

+		{

+			printf ("EOF reached (Burst Reading)!\nCurrent position in INFILE: %i\n", ftell(infile));

+			printf ("Frame size: %i  .. Bytes read: %i\n", payloadbytes, bytesread);

+			break;

+		}

+		ac3p.burst_infoMSB = (burst[13] & 7);

+		ac3p.lengthcode = (short)(payloadbytes * 8);

+

+		for (i = 8; i < (payloadbytes + 8); i += 2)

+		{

+			temp = burst[i];

+			burst[i] = burst[i + 1];

+			burst[i + 1] = temp;

+		}

+

+		memcpy (burst, &ac3p, sizeof(struct AC3PREAMBLE));

+

+		fwrite (burst, 1, 6144, outfile);

+

+	}

+

+	printf ("Last Sampleratecode: %i\n", sampleratecode);

+

+	wavhdr.SubChunk2Size = (ftell(outfile) - 44);

+	wavhdr.ChunkSize = wavhdr.SubChunk2Size + 36;

+

+	fseek (outfile, SEEK_SET, 0);

+	fwrite (&wavhdr, sizeof(struct WAVEHEADER), 1, outfile);

+

+	fclose (infile);

+	fclose (outfile);

+

+	return 0;

+}

diff --git a/dts2spdif.c b/dts2spdif.c
new file mode 100755
index 0000000..6aa15bb
--- /dev/null
+++ b/dts2spdif.c
@@ -0,0 +1,177 @@
+

+#include <stdio.h>

+//#include <stdlib.h>

+//#include <string.h>

+#include <memory.h>

+

+struct WAVEHEADER {

+    unsigned long  ChunkID;

+    unsigned long  ChunkSize;

+    unsigned long  Format;

+    unsigned long  SubChunk1ID;

+    unsigned long  SubChunk1Size;

+    unsigned short AudioFormat;

+    unsigned short NumChannels;

+    unsigned long  SampleRate;

+    unsigned long  ByteRate;

+    unsigned short BlockAlign;

+    unsigned short BitsPerSample;

+    unsigned long  SubChunk2ID;

+    unsigned long  SubChunk2Size;

+};

+

+

+unsigned char burst[6144];

+

+FILE *infile, *outfile;

+

+unsigned long bytesread;

+unsigned long pcmbytes;

+unsigned long pcmbytesmultiplier;

+char sampleratecode;

+unsigned long framesize;

+

+struct WAVEHEADER wavhdr = { 0x46464952,

+							0,

+							0x45564157,

+							0x20746D66,

+							16,

+							1,

+							2,

+							0,

+							0,

+							4,

+							16,

+							0x61746164,

+							0 };

+

+unsigned long i;

+char temp;

+

+int main( int argc, char *argv[ ], char *envp[ ] )

+{

+	if (argc < 3)

+	{

+		printf("Wrong syntax. dts2spdif <In.dts> <Out.wav>.\n");

+		return 0;

+	}

+

+	infile = fopen(argv[1], "rb");

+	outfile = fopen(argv[2], "wb");

+	//fseek(outfile, SEEK_SET, 44);

+	fwrite (&wavhdr, sizeof(struct WAVEHEADER), 1, outfile);

+

+

+	for(;;)

+	{

+		memset (burst, 0, 6144);

+		bytesread = fread(burst, 1, 9, infile);

+		if (bytesread < 9)

+		{

+			printf ("EOF reached (Frame Header reading)!\nCurrent position in INFILE: %i\n", ftell(infile));

+			break;

+		}

+		if (((unsigned long*)burst)[0] != 0x0180fe7f)

+		{

+			printf("ERROR: INVALID SYNCWORD !\nCurrent position in INFILE: %i\n", ftell(infile));

+			break;

+		}

+		framesize = (burst[5] & 3);

+		framesize = framesize * 4096;

+		framesize = framesize + (burst[6] * 16);

+		framesize = framesize + ((burst[7] & 240) / 16);

+		framesize = framesize + 1;

+

+		sampleratecode = ((burst[8] & 60) / 4);

+

+		if (wavhdr.SampleRate == 0)

+		{

+				printf ("First Sampleratecode: %i\n", sampleratecode);

+

+				switch (sampleratecode)

+				{

+				case 1:

+					pcmbytesmultiplier = 4;

+				case 2:

+					pcmbytesmultiplier = 2;

+				case 3:

+					pcmbytesmultiplier = 1;

+					wavhdr.SampleRate = 32000;

+					wavhdr.ByteRate = 128000;

+					break;

+				case 6:

+					pcmbytesmultiplier = 4;

+				case 7:

+					pcmbytesmultiplier = 2;

+				case 8:

+					pcmbytesmultiplier = 1;

+					wavhdr.SampleRate = 44100;

+					wavhdr.ByteRate = 176000;

+					break;

+				case 11:

+					pcmbytesmultiplier = 4;

+				case 12:

+					pcmbytesmultiplier = 2;

+				case 13:

+					pcmbytesmultiplier = 1;

+					wavhdr.SampleRate = 48000;

+					wavhdr.ByteRate = 192000;

+					break;

+				default:

+					wavhdr.SampleRate = 0;

+					wavhdr.ByteRate = 0;

+					printf ("Invalid Sampleratecode ! Aborting process...\n");

+					break;

+				}

+		}

+

+		pcmbytes = (burst[4] & 1);

+		pcmbytes = pcmbytes * 64;

+		pcmbytes = pcmbytes + ((burst[5] & 252) / 4);

+		pcmbytes = pcmbytes + 1;

+		pcmbytes = pcmbytes * 128;

+		pcmbytes = pcmbytes * pcmbytesmultiplier;

+

+

+		bytesread = fread (&burst[9], 1, framesize - 9, infile);

+		if ((bytesread + 9) < framesize)

+		{

+			printf ("EOF reached (Burst Reading)!\nCurrent position in INFILE: %i\n", ftell(infile));

+			printf ("Frame size: %i  .. Bytes read: %i\n", framesize, bytesread);

+			break;

+		}

+

+		for (i = 0; i < framesize; i += 2)

+		{

+			temp = burst[i];

+			burst[i] = burst[i + 1];

+			burst[i + 1] = temp;

+		}

+

+

+		fwrite (burst, 1, framesize, outfile);

+		if (pcmbytes > framesize)

+		{

+			fwrite ("", 1, pcmbytes - framesize, outfile);

+		}

+		else

+		{

+			printf("Warning: Frame Size > LPCM Frame Size! Buffer underrun may occur/File damaged?\n");

+			printf("Frame start at INFILE Offset: %i\n", ftell(infile) - framesize);

+		}

+	}

+

+	printf ("Last Sampleratecode: %i\n", sampleratecode);

+

+	wavhdr.SubChunk2Size = (ftell(outfile) - 44);

+	wavhdr.ChunkSize = wavhdr.SubChunk2Size + 36;

+

+	fseek (outfile, SEEK_SET, 0);

+	fwrite (&wavhdr, sizeof(struct WAVEHEADER), 1, outfile);

+

+

+	fclose (infile);

+	fclose (outfile);

+

+	return 0;

+}