MultimediaWiki - User contributions [en]

DTS

2024-02-16T15:45:24Z

Rjamorim: Updated the libdca link

* Codec ID: 0x2001 (0x0008 Microsoft DTS)
* Company: [[DTS Inc.]]

DTS Coherent Acoustics is a perceptual audio codec. The main technique used is a QMF filter together with [[Huffman]], [[PCM#Adaptive DPCM|ADPCM]] and [[Vector Quantization|vector quantization]].

Specification can be found [http://webapp.etsi.org/WorkProgram/Report_WorkItem.asp?WKI_ID=35781 here], [http://www.etsi.org/deliver/etsi_ts/102100_102199/102114/01.03.01_60/ts_102114v010301p.pdf direct link] (v1.3.1, 2011-08-09); [[Mirrored Files|mirrored locally]].

The [[VideoLAN]] project has created [http://www.videolan.org/developers/libdca.html libdts/libdca] an open source implementation of DTS. Unfortunately libdts was pushed underground through patent scare tactics for a while.

== Bitstream coding ==
There are four different bitstream formats -- little- or big-endian and 16 or 14 bits per word.

=== Raw bitstream coding ===
Raw bitstream is packed into 16-bit words with possible little- or big-endian. The native file formats, compact (.cpt) and padded (.dts), are all coded in big-endian byte order.

=== 14-bit words ===
This kind of bitstream is packed into 16-bit sample words so that the amplitude is reduced by 12 dB in the event that the data is inadvertently interpreted as uncompressed audio samples. The upper two bits are basically sign bit extension, as defined by twos-complement format. This coding is used in many DTS-in-[[WAV]] files. It is also found on DTS Music CD, where the potential of 1411.2 kbps is reduced to 1234.8 kbps in a tradeoff to reduce speaker damage.

=== How to distinguish different versions ===
Every frame in DTS starts with 32-bit syncword which can be used to distinguish current bitstream encoding:

{| border="1"
! Sequence !! Bitstream type
|-
| 7F FE 80 01 ||raw big-endian
|-
| FE 7F 01 80 ||raw little-endian
|-
| 1F FF E8 00 07 Fx || 14-bit big-endian
|-
|FF 1F 00 E8 Fx 07 ||14-bit little-endian
|}

DTS is natively big-endian byte order. Wave and CD are little-endian formats, so DTS will (usually) be little-endian when found in those media containers.

== Frame format ==

{| border="1"
! Size !! Explanation !! Value
|-
| 32 bits || Sync marker || 0x7FFE8001
|-
| 1 bit || Frame type (or last frame) || 0 - termination frame, 1 - normal frame
|-
| 5 bits || Deficit sample count || number of samples in block - 1 (should be 31 for normal frame)
|-
| 1 bit || CRC present ||
|-
| 7 bits || Number of blocks || 5-127
|-
| 14 bits || Frame size in bytes - 1|| 95-16383
|-
| 6 bits || Channel configuration || values 0-15 are standard, 16-63 are user-defined
|-
| 4 bits || Sample frequency || See table below
|-
| 5 bits || Bitrate || See table below
|-
| 1 bit || Down mix ||
|-
| 1 bit || Dynamic range ||
|-
| 1 bit || Time stamp ||
|-
| 1 bit || Auxiliary data ||
|-
| 1 bit || HDCD ||
|-
| 3 bits || External descr ||
|-
| 1 bit || External coding ||
|-
| 1 bit || ASPF ||
|-
| 2 bits || LFE ||
|-
| 1 bit || Predictor history ||
|-
| 16 bits || Header CRC || if CRC present above is set
|-
| 1 bit || Multirate inter ||
|-
| 4 bits || Version ||
|-
| 2 bits || Copy history ||
|-
| 3 bits || Source PCM resolution ||
|-
| 1 bit || Front sum ||
|-
| 1 bit || Surround sum ||
|-
| 4 bits || Dialog normalization || dB of recovery
|-
| ""SUBFRAMES""
| '''TODO'''
|}
[[Category:Audio Codecs]]
[[Category: QMF Audio Codecs]]
[[Category: Multichannel Audio Codecs]]

AudioZip

2017-10-30T18:20:54Z

Rjamorim:

* Description: [http://www.rarewares.org/rrw/audiozip.php AudioZip]

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

TwinVQ

2017-10-25T17:16:11Z

Rjamorim:

* TwoCCs: Unknown
* Company: [[Nippon TT]]
* Metatag format: http://www.twinvq.org/technology_format.html (Japanese)
* Fileformat: [[VQF]]

Perceptual audio codec. The main technique used is a vector quantizer. The codebook is about 500kb in size, this results
in long encoding times but fast decoding.

http://www.twinvq.org/english/index_en.html
has the official encoder and player for [[Windows]].

http://www.mp3-tech.org/programmer/sources/refsoft990809.tgz
has code for the vqf version that made it into [[AAC]].

http://www.intel.com/cd/software/products/asmo-na/eng/perflib/ipp/219611.htm?prn=y
Intel shares some code for AAC flavored vqf.

http://www.rarewares.org/rrw/nttvqf.php hosts several encoders and decoders. It includes [[NTT]]'s binary-only encoding and decoding libraries and C++ header files.

[[Category:Audio Codecs]]
[[Category: MDCT Audio Codecs]]

QDesign Music Codec

2017-10-25T17:06:37Z

Rjamorim:

* FOURCCs: QDMC, QDM2
* Company: [[QDesign]]
* Samples: http://samples.mplayerhq.hu/A-codecs/QDMC/ (QDMC)
* Samples: http://samples.mplayerhq.hu/A-codecs/QDM2/ (QDM2)
* Technical Analysis: http://multimedia.cx/mirror/qdmc2.pdf
* Description and encoders: http://www.rarewares.org/rrw/qdmc.php

QDesign Music Codec is a perceptual audio codec commonly used in [[MOV]] files between 1998 and 2005, after which it was more or less supplanted by [[AAC]] for the same role. Variants of QDM2 with 32, 16 or 8 subbands exist, 16 and 8 decode with noticeable artifacts (metallic sound).

QDMC was the first version of the codec which was apparently short lived and quickly superceded by the second version, QDM2. Both QDM2 and QDMC are supported in open source software (via [[FFmpeg]]).

=== QDesign Music Codec (v1) ===

This codec codes audio as a set of tones and noise parameters that are later reconstructed with FFT.

For sampling rates < 16kHz subframe should be 64 samples and full frame should be 2048 samples. For 16-32kHz it's 128 and 4096 samples correspondingly. For >=32kHz it's 256 and 8192 samples.

Bits are read from LSB using 16-bit little-endian words.

==== Subblock format ====

===== Noise data =====

for (band = 0; band < noise_bands[mode]; band++) {
v = get_huff(noise_val_tree);
if (v & 1)
v = (v + 1) >> 1;
else
v = -v / 2;
noise[band][0] = v - 1;
lastval = v;
for (idx = 0; idx < 16;) {
len = get_huff_long(noise_seg_len_tree);
v = get_huff(noise_val_tree);
if (v & 1)
v = (v + 1) >> 1;
else
v = -v / 2;
newval = lastval + v;
for (j = 0; j < len; j++)
noise[band][idx + 1 + j] = lastval + v * j / len;
lastval = newval;
idx += len + 1;
}
}

===== Wave data =====

for (group = 0; group < 5; group++) {
group_size = 1 << (frame_bits - group - 1);
group_bits = 4 - group;
freq = 1;
off = 0;
pos2 = 0;
do {
freq += get_huff_long(freq_diff_tree);
while (freq >= group_size - 1) {
freq -= group_size - 2;
off += 1 << group_bits;
pos2 += group_size;
}
if (pos2 >= frame_size)
break;
if (channels > 1)
stereo_mode = get_bits(2);
else
stereo_mode = 0;
amp = get_huff(amp_tree);
phase = get_bits(3);
if (stereo_mode & 2) {
amp2 = amp - get_huff(amp_diff_tree);
phase2 = (phase - get_huff(phase_diff_tree)) & 7;
} else {
amp2 = 0;
phase2 = 0;
}
add tone <off, freq, stereo_mode & 1, amp, phase>
if (stereo_mode & 2)
add tone <off, freq, ~stereo_mode & 1, amp2, phase2>
} while (freq < group_size);
}

===== Huffman code reading =====

Trees are static but the codes can have an additional data afterwards:

int get_huff(tree) {
v = read_code(tree);
if (v)
return v - 1;
else {
v = get_bits(3) + 1;
return get_bits(v);
}
}

int get_huff_long(tree) {
v = read_code(tree);
if (v)
v--;
else
v = get_bits(get_bits(3) + 1);
return code_prefix[v] + get_bits(v >> 2);
}

unsigned code_prefix[] = {
0x0, 0x1, 0x2, 0x3, 0x4, 0x6, 0x8, 0xA,
0xC, 0x10, 0x14, 0x18, 0x1C, 0x24, 0x2C, 0x34,
0x3C, 0x4C, 0x5C, 0x6C, 0x7C, 0x9C, 0xBC, 0xDC,
0xFC, 0x13C, 0x17C, 0x1BC, 0x1FC, 0x27C, 0x2FC, 0x37C,
0x3FC, 0x4FC, 0x5FC, 0x6FC, 0x7FC, 0x9FC, 0xBFC, 0xDFC,
0xFFC, 0x13FC, 0x17FC, 0x1BFC, 0x1FFC, 0x27FC, 0x2FFC, 0x37FC,
0x3FFC, 0x4FFC, 0x5FFC, 0x6FFC, 0x7FFC, 0x9FFC, 0xBFFC, 0xDFFC,
0xFFFC, 0x13FFC, 0x17FFC, 0x1BFFC, 0x1FFFC, 0x27FFC, 0x2FFFC, 0x37FFC,
0x3FFFC
};

==== Tables ====

Noise bands selector (depending on bitrate):

4, 3, 2, 1, 0, 0, 0, 0

Number of noise bands:

19, 14, 11, 9, 4

Noise subbands:

0, 1, 2, 4, 6, 8, 12, 16, 24, 32, 48, 56, 64, 80, 96, 120, 144, 176, 208, 240, 256
0, 2, 4, 8, 16, 24, 32, 48, 56, 64, 80, 104, 128, 160, 208, 256
0, 2, 4, 8, 16, 32, 48, 64, 80, 112, 160, 208, 256
0, 4, 8, 16, 32, 48, 64, 96, 144, 208, 256
0, 4, 16, 32, 64, 256

Levels table:

1.1875, 1.6835938, 2.375, 3.3671875, 4.75, 6.734375, 9.5, 13.46875,
19.0, 26.9375, 38.0, 53.875, 76.0, 107.75, 152.0, 215.5,
304.0, 431.0, 608.0, 862.0, 1216.0, 1724.0, 2432.0, 3448.0,
4864.0, 6896.0, 9728.0, 13792.0, 19456.0, 27584.0, 38912.0, 55168.0,
77824.0, 110336.0, 155648.0, 220672.0, 311296.0, 441344.0, 622592.0, 882688.0,
1245184.0, 1765376.0, 2490368.0, 3530752.0, 4980736.0, 7061504.0

==== Frequencies for the trees ====

Table 0 — noise value

3233, 1195, 1897, 877, 1240, 368, 364, 222, 103, 125, 18, 68, 10, 25, 7, 13,
0, 18, 0, 20, 0, 31, 0, 28, 0, 31, 0, 19, 0, 23, 0, 10, 0, 9, 0, 1,

Table 1 — noise segment length

7647, 1011, 380, 215, 180, 65, 33, 12, 4, 0, 0, 0, 16, 0, 0, 0, 84

Table 2 — amplitude

2436, 1411, 692, 389, 316, 310, 368, 457, 651, 1359, 2563, 4732, 8946, 17150, 29621, 44245,
50156, 45928, 33262, 20474, 9855, 3813, 1378, 514, 154, 82, 3

Table 3 — frequency difference

57884, 27424, 14988, 11027, 17889, 14609, 11790, 9479,
15948, 11581, 7815, 6917, 10486, 6603, 4897, 3983,
5120, 3479, 2949, 2626, 3443, 2984, 3725, 3593,
3307, 3283, 2954, 2384, 1777, 2042, 1641, 798,
769, 863, 776, 239, 162, 104, 63, 43,
49, 30, 6, 1, 0, 4, 1

Table 4 — amplitude difference tree

8392, 14998, 5103, 1797, 648, 237, 42, 7

Table 5 — phase difference tree

8860, 9620, 2138, 897, 618, 834, 1920, 6337

Converting frequencies into codes is left as an exercise to the reader.

[[Category:Audio Codecs]]

Meridian Lossless Packing

2017-10-25T17:05:11Z

Rjamorim:

* Whitepaper: http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF ([[Mirrored Files|mirrored]])
* Samples: http://samples.mplayerhq.hu/A-codecs/lossless/mlp/

Also known as Dolby [[TrueHD]]. This is a lossless audio codec for DVD audio, as stated in the whitepaper, it uses IIR prediction (like TrueAudio or Monkey's Audio) and [[Golomb]] codes residue packing.

You can download samples created by [[User:Rjamorim|rjamorim]] at RareWares:
* http://www.rarewares.org/rja/samples/
they were created with Surcode MLP, and the original samples are also available ([[WavPack]]-encoded). God Save The Queen is a 6ch/96k/24b sample from Queen's A Night At The Opera DVD-A. luckynight is the same sample used [http://samples.mplayerhq.hu/A-codecs/lossless/ elsewhere] in this wiki, and 440hz is, as the name implies, a simple 440 Hz tone.

You can [http://www.rarewares.org/ contact] rjamorim if you want other samples encoded.

== Overview ==

A good overview of the encoding/decoding process is available on Meridian's website:
* http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF

An MLP audio stream can contain multiple substreams - a decoder can optionally decode only a subset of these substreams. Typically stereo files are encoded with a single substream, while surround files have one substream containing a stereo downmix, and another substream containing the additional surround channels and matrixing parameters to extract the front channels from the stereo downmix.

MLP streams can specify that the second substream can have a lower bit depth or sampling frequency than the first substream. TrueHD removes this capability.

== Stream structure ==

A stream is split up into a number of small blocks, or "access units". Each access unit contains coded data for 40 (44100/48000Hz), 80 (88200/96000Hz) or 160 (176400/192000Hz) samples. Periodically an access unit contains a "major sync" header that contains stream metadata (analogous to a sequence header in a video codec).

All values in the stream are big endian.

=== Access unit structure ===

Each access unit starts with a parity nibble (details later), and then a 12-bit length field (in units of two bytes). Following this is a 2-byte value that looks like it may be some form of DTS - in most cases each packet will have a value that is the sum of the previous packet's value and the number of samples per packet, but some values can diverge from this.

After these first four bytes, the major sync header can optionally appear - if the next four bits are all one, then the following 28 bytes (including the four one bits) are a major sync header.

Following this is an index of two- or four-byte entries for each substream. These have the following format:

{| class="wikitable"
|-
! Size !! Meaning
|-
| 1 bit || If set, indicates an extra 16 bits of data are present
|-
| 1 bit || If clear, indicates that the substream data starts with a "restart header"
|-
| 1 bit || If set, indicates that the substream data contains extra parity information
|-
| 1 bit || Unknown
|-
| 12 bits || Offset, in units of two bytes, of the end of the data for this substream from the start of the start of the substream data block (ie length of this substream's data plus all previous)
|-
| 16 bits || (Optional, present if first bit was set) Unknown
|-
|}

Following these entries is the data for each substream.

The first nibble of the access unit is a parity check on the access unit header - XOR'ing together the nibbles of the first four bytes and the substream entry table above (excluding the major sync header if present and the substream data) should give a result of 0xF.

=== Major sync header ===

The major sync header is 28 bytes long, and starts with the three bytes F8 72 6F, followed by BB for MLP streams, or BA for TrueHD streams. This contains details like the number of substreams, the channel arrangement, bit depth and sampling frequency. In MLP streams major syncs must occur between every 8 and 32 access units. In TrueHD streams spacings of 128 access units have been observed.

The major sync header for MLP streams has the following format:

{| class="wikitable"
|-
! Size !! Meaning
|-
| 24 bits || SYNC_MAJOR (0xf8726f)
|-
| 8 bits || SYNC_MLP (0xbb)
|-
| 4 bits || Bit-depth (substream 0)
|-
| 4 bits || Bit-depth (substream 1)
|-
| 4 bits || Sample-rate (substream 0)
|-
| 4 bits || Sample-rate (substream 1)
|-
| 11 bits || Unknown
|-
| 5 bits || Channel arrangement
|-
| 16 bits || MAJOR_SYNC_INFO_SIGNATURE (0xb752)
|-
| 16 bits || Info flags
|-
| 16 bits || Unknown
|-
| 1 bit || Set to 1 if VBR
|-
| 15 bits || Coded peak bitrate
|-
| 4 bits || Number of substreams
|-
| 4 bits || Unknown (set to 0x1)
|-
| 8 bits || Substream info
|-
| 5 bits || Unknown (some substream info)
|-
| 5 bits || Sample wordlength (no. of bits)
|-
| 6 bits || Channel occupancy
|-
| 3 bits || Unknown
|-
| 10 bits || Speaker layout?
|-
| 3 bits || Copy protection
|-
| 16 bits || Unknown (set to 0x8080)
|-
| 7 bits || Unknown
|-
| 4 bits || Source format
|-
| 5 bits || Summary info
|-
|}

=== Substream data block ===

A substream data block contains one or more chunks of data.

A data chunk may start with a restart header followed by a decoding parameter block, just a decoding parameter block, or may just go straight into data.

A decoding parameter block is present if the first bit is '1'. If the next bit is also '1', a restart header is also present, before the decoding parameter block.

The encoded residual data then follows. If data_check_present was set in the last restart header for this substream, the data is preceded by a 16-bit count indicating how many bits of data should be present, and followed by an 8-bit value (currently unknown, maybe a CRC). The bit count includes the 16 count bits, but not the 8 extra bits.

A single bit marks the end of a chunk - this is set to 1 if it is the last chunk in the data block.

After the last chunk, padding bits are inserted to align the bitstream to a 16-bit boundary. The 32-bit value D2 34 D2 34 may then optionally be present, which indicates that this was the last access unit of the audio stream.

If substream parity information was indicated in the substream index earlier, the data for the substream is terminated by a parity check byte and a CRC byte. The parity check is such that XORing all the bytes of the substream up to (and including) the parity byte should give the value 0xA9.

=== Restart header ===

The restart header contains information necessary for the decoding of the audio stream. It is generally present in packets containing a major sync header. The restart header starts with a 14-bit code, either 0x31EA or 0x31EB. MLP streams only use 0x31EA headers, TrueHD streams use either/both. The difference lies mostly in the way rematrixing is done.

TODO: Document more

[[Category:Audio Codecs]]
[[Category:Lossless Audio Codecs]]
[[Category: Multichannel Audio Codecs]]

Split2000

2017-10-25T17:04:00Z

Rjamorim:

*Description: [http://www.rarewares.org/rrw/split2000.php Split2000]

Some interesting info from the page above:
It was only on Linux, and the source code to the program is available (it was for a very old kernel).
There's also a binary for newer systems and a diff file as well.

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category:Formats missing in FFmpeg]]

DSP Group Truespeech

2017-10-25T17:00:49Z

Rjamorim:

* Format tag: 0x22
* Company: [[DSP Group]]
* Description and binaries: http://www.rarewares.org/rrw/truespeech.php
* Samples: http://samples.mplayerhq.hu/A-codecs/truespeech/

This is the codec developed by [[DSP Group]]. It operates on 8000 Hz, 16-bit mono [[PCM]] and have some fixed bitrates, Windows version supports 15:1 compression (~1 Kbps) only.

This codec employs LPC (Linear Predictive Coding) as almost every speech codec does.

== Bitstream format ==

Frame (240 samples) is divided into 4 subframes.

Packed frame data holds 8-order filter used in synthesis, 4x7 pulse positions and values, offset in previously decoded frame.

All data is stored in 32-bit little-endian words and bits are read from the LSB.

=== Frame format ===

// 1st word
1 bit - previous filter selection mode
the rest are 8 filter coefficients with the next bit allocation: 5, 5, 4, 4, 4, 3, 3, 3

// 2nd word
4x7 bits - offset2 values 0-3
4 bits - high 4 bits of offset1[0]

// 3d word
2x14 bits - pulse values 0 and 1
4 bits - low 4 bits of offset1[1]

// 4th word
2x14 bits - pulse values 2 and 3
4 bits - high 4 bits of offset1[1]

// 5th word
4 bits - pulse offset 0
27 bits - pulse position 0
1 bit - offset1[0] bit 0

// 6th word
4 bits - pulse offset 1
27 bits - pulse position 1
1 bit - offset1[0] bit 1

// 7th word
4 bits - pulse offset 2
27 bits - pulse position 2
1 bit - offset1[0] bit 2

// 8th word
4 bits - pulse offset 3
27 bits - pulse position 3
1 bit - offset1[0] bit 3

previous filter selection mode flag - when set to one, first subframe uses filter average 2/3*prev_filt[] + 1/3*cur_filt[] and 1/3*prev_filt[]+2/3*cur_filt[] for the second subframe, otherwise previous filter coefficients are used for both; the third and fourth subframes always use current filter.

== Technical details (for standard scheme) ==

Decoding flow:

unpack frame data
reconstruct packed filter
merge this filter with previous to create 4 filters for each subframe
for each subframe {
apply twopoint filter on some saved data
place pulses (7 pulses for each subframe max)
update saved filter
apply main filter to subframe data (mostly pulses)
}

[[Category: Audio Codecs]]

VocPack

2017-10-25T16:58:03Z

Rjamorim:

*Description: [http://www.rarewares.org/rrw/vocpack.php VocPack]

From the description, this appears to be one of the earliest lossless codecs: it dates from 1993.
Two versions of the encoder/decoder are available on the site listed above

*Identifying file:
Hex: 4E 46 56 50 20
ASCII: NFVP
from [http://filext.com/detaillist.php?extdetail=VP File Extension Details for VP]

[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

RK Audio

2017-10-25T16:56:17Z

Rjamorim:

* Extension: rka
* Website: [http://www.msoftware.co.nz/downloads_page.php http://www.msoftware.co.nz/downloads_page.php]
* Description: http://www.rarewares.org/rrw/rkau.php
* Company: [[M Software]]
* Samples: http://samples.mplayerhq.hu/A-codecs/lossless/ (luckynight.rka)

RK Audio is a lossless audio coding algorithm.

[[Category:Container Formats]]
[[Category:Audio Codecs]]
[[Category:Lossless Audio Codecs]]
[[Category:Undiscovered Audio Codecs]]

Pegasus Sound Processing Software

2017-10-25T16:54:49Z

Rjamorim:

Pegasus Sound Processing Software (SPS) was developed by KrishnaSoft as a general purpose audio editor, that featured a lossless mode based on Pegasus' ELS (Entropy Logarithmic Scale) compression algorithm.

==External links==
* [http://www.krishnasoft.com/pages/1/index.htm] (official website)
* Description: http://www.rarewares.org/rrw/pegasussps.php

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

NCT Soft ALF2CD ACM

2017-10-25T16:52:36Z

Rjamorim:

* Codec ID: 0x1FC4
* Website: http://www.nctsoft.com/
* Description: http://www.rarewares.org/rrw/alf2cd.php
* Author: Lev Bannov / leksei Myasnikov

Almost nothing is known about this audio codec.

[[Category:Audio Codecs]]
[[Category:Undiscovered Audio Codecs]]

MUSICompress

2017-10-25T16:52:13Z

Rjamorim:

* Extensions: mcp
* Company: Soundspace Audio
* Homepage: [http://members.aol.com/sndspace Soundspace]
* Description: http://www.rarewares.org/rrw/wavezip.php
* [https://groups.google.com/forum/?fromgroups=#!topic/comp.music.research/uCu3xAi2k7o Introductory letter from author]
* Publication: A. Wegener, “MUSICompress: Lossless, lowMIPS audio compression in software and hardware”, in Proc. Int. Conf. Signal Processing Applications and Technology, San Diego, CA, USA, 1997 September 14-17.

Implementation also called WaveZip.

[[Category:Audio Codecs]]
[[Category:Undiscovered Audio Codecs]]
[[Category:Formats missing in FFmpeg]]

Marian's A-pac

2017-10-25T16:51:34Z

Rjamorim:

* Description: [http://www.rarewares.org/rrw/apac.php apac]
* Homepage: [http://www.marian.de/english/downloads.htm#APAC APAC]

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

Lossless Predictive Audio Coding

2017-10-25T16:51:13Z

Rjamorim:

* Extension: pac
* Website: http://www.nue.tu-berlin.de/fileadmin/fg97/Ueber_uns/Team/Liebchen/LPAC/lpacarc.html
* Samples: http://samples.mplayerhq.hu/A-codecs/lossless/ (luckynight.pac)
* Description: http://www.rarewares.org/rrw/lpac.php

Lossless Predictive Audio Coding (LPAC) is a lossless audio coding method. According to the official website an "improved version" was chosen to be the lossless audio method ([[MPEG-4 Audio Lossless Coding (ALS)|ALS]]) for the MPEG-4 standard.

"The Linux and Solaris versions are executable (binary) files. It is not possible to release the source code (yet) due to copyright issues - sorry for that! Maybe there will be a GPL solution in the future."

[[Category:Container Formats]]
[[Category:Audio Codecs]]
[[Category:Lossless Audio Codecs]]
[[Category:Undiscovered Audio Codecs]]

LiteWave

2017-10-25T16:50:16Z

Rjamorim:

* Description: [http://www.rarewares.org/rrw/litewave.php Litewave]
Description has link to encoder, and the homepage for the product has a link to the free encode.

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

Dakx Wav

2017-10-25T16:49:46Z

Rjamorim:

* Description: [http://www.rarewares.org/rrw/daxwav.php Dakx Wav]
* Website: http://www.dakx.com/
* Patent: [http://www.google.com/patents?vid=USPAT5825830 USPAT 5,825,830]
* Simple decoder/encoder: http://www.dakx.com/software/simdaxg.c

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

AUPECg2

2017-10-25T16:49:06Z

Rjamorim:

* Website: http://web.archive.org/web/20020306212541/students.washington.edu/mvinton/downloads.html
* Description: http://www.rarewares.org/rrw/aupec.php

[[Category:Audio Codecs]]
[[Category:Undiscovered Audio Codecs]]

AudioZip

2017-10-25T16:48:24Z

Rjamorim:

* Description: [http://www.rarewares.org/rrw/audiozip.php AudioZip] [http://web.tiscali.it/mupuxeddu/software/old/audiozip.html AudioZip #2]

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

Advanced Digital Audio

2017-10-25T16:48:04Z

Rjamorim:

*Description: [http://www.rarewares.org/rrw/ada.php ADA]

[[Category: Audio Codecs]]
[[Category: Lossless Audio Codecs]]
[[Category: Undiscovered Audio Codecs]]

Microcosm

2008-06-22T15:20:42Z

Rjamorim:

* FourCCs: ?
* Company: [http://www.digitalanarchy.com/micro/main.html Digital Anarchy]

Microcosm is a [[QuickTime]] codec targeted at lossless encoding of 64BPP RGBA movies. This codec was designed as an editing format and doesn't seem to be used for media distribution. Digital Anarchy offers a demo version for Mac and Windows that watermarks the video data, but doesn't seem to have time expiration.

They also developed a freeware uncmpressed codec for 64BPP RGBA movies called [http://www.digitalanarchy.com/micro/free.html None16].

[[Category:Video Codecs]]
[[Category:Undiscovered Video Codecs]]

Microcosm

2008-06-22T15:20:03Z

Rjamorim:

* FourCCs: ?
* Company: [http://www.digitalanarchy.com/micro/main.html Digital Anarchy]

Microcosm is a [[QuickTime]] codec targeted at lossless encoding of 64BPP RGBA movies. This codec was designed as an editing format and is rarely used for media distribution. Digital Anarchy offers a demo version for Mac and Windows that watermarks the video data, but doesn't seem to have time expiration.

They also developed a freeware uncmpressed codec for 64BPP RGBA movies called [http://www.digitalanarchy.com/micro/free.html None16].

[[Category:Video Codecs]]
[[Category:Undiscovered Video Codecs]]

Meridian Lossless Packing

2006-09-16T22:08:30Z

Rjamorim:

Also known as Dolby TrueHD. This is a lossless audio codec for DVD audio, as it states [http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF here], it uses IIR prediction (like TrueAudio or Monkey's Audio) and Golomb codes residue packing.

You can download samples created by [[User:Rjamorim|rjamorim]] at RareWares:
* http://www.rarewares.org/rja/samples/
they were created with Surcode MLP, and the original samples are also available ([[WavPack]]-encoded). God Save The Queen is a 6ch/96k/24b sample from Queen's A Night At The Opera DVD-A. luckynight is the same sample used [http://samples.mplayerhq.hu/A-codecs/lossless/ elsewhere] in this wiki, and 440hz is, as the name implies, a simple 440 Hz tone.

You can [http://www.rjamorim.com/contact-en.html contact] rjamorim if you want other samples encoded.

[[Category:Undiscovered Audio Codecs]]
[[Category:Lossless Audio Codecs]]
[[Category:Audio Codecs]]

User:Rjamorim

2006-09-06T01:27:39Z

Rjamorim:

http://www.rjamorim.com

Hello, and welcome to my info page.
My real name is Roberto José de Amorim.

I'm known in the audiocoding community from

* Being the admin/maintainer of [http://www.rarewares.org/ RareWares]
* Being the admin/maintainer of [http://www.rjamorim.com/rrw/ ReallyRareWares]
* Conducing several [http://www.rjamorim.com/test/ Listening Tests]
* Being member of [http://www.hydrogenaudio.org Hydrogenaudio]
* Formerly being an AAC freak.
* Being hated by Dolby with all their [http://pessoal.onda.com.br/rjamorim/Dolby.txt might]

I'm 23 years old, Brazilian and live in Curitiba.

Studying Computing Science at University.

My e-mail address: rjamorim@yahoo.com

ICQ number: 19724800

MSN contact: rjamorim@hotmail.com

Meridian Lossless Packing

2006-09-06T00:46:44Z

Rjamorim:

This is lossless audio codec for DVD audio, as it states [http://www.meridian-audio.com/w_paper/mlp_jap_new.PDF here], it uses IIR prediction (like TrueAudio or Monkey's Audio) and Golomb codes residue packing.

You can download samples created by [[User:Rjamorim]] at RareWares:
* http://www.rarewares.org/rja/samples/
they were created with Surcode MLP, and the original samples are also available ([[WavPack]]-encoded). God Save The Queen is a 6ch/96k/24b sample from Queen's A Night At The Opera DVD-A. luckynight is the same sample used [http://samples.mplayerhq.hu/A-codecs/lossless/ elsewhere] in this wiki, and 440hz is, as the name implies, a simple 440 Hz tone.

You can [http://www.rjamorim.com/contact-en.html contact] rjamorim if you want other samples encoded.

[[Category:Undiscovered Audio Codecs]]
[[Category:Lossless Audio Codecs]]
[[Category:Audio Codecs]]