MultimediaWiki - User contributions [en]

DTS-HD

2008-01-17T14:06:45Z

Andoma: It's multichannel

* Company: [[DTS Inc.]]
* Whitepaper: http://www.dtsonline.com/media/DTS-HD_WhitePaper.pdf

DTS-HD is an audio coding technology developed by DTS and targeted for the HD generation of optical discs (namely [[Blu-Ray]] and [[HD-DVD]]). The technology specification embodies various modes and extensions, one of which is XLL, an extension for lossless audio coding.

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

Category:Multichannel Audio Codecs

2008-01-17T14:04:15Z

Andoma:

List of all audio codecs that can deliver multichannel (more than 2 channels) of audio.

Advanced Audio Coding

2008-01-17T14:03:08Z

Andoma:

* FOURCCS: ????
* Company: [[ISO]]

Specification and reference source links.
http://www.3gpp.org/ftp/specs/html-info/26-series.htm
ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg4/

[[Intel]] has source for its IPP based [[AAC]] decoder: http://www.intel.com/cd/software/products/asmo-na/eng/perflib/ipp/219611.htm?prn=y

[[Real|Real Networks]] has source for its Helix AAC fixedpoint decoder here: https://datatype.helixcommunity.org/2005/aacfixptdec

Usually stored in an [[MP4]] container.

See also [[Understanding AAC]].

== Extensions ==

=== AAC+ ===

[http://www.codingtechnologies.com/products/aacPlus.htm AACPlus] (a.k.a. AAC+) is HE-AAC + [http://www.codingtechnologies.com/products/sbr.htm SBR (Spectral Band Replication)] and aacPlus v2 means HE-AAC + [http://www.codingtechnologies.com/products/sbr.htm SBR (Spectral Band Replication)] + [http://www.codingtechnologies.com/products/paraSter.htm PS (Parametric Stereo)]. Standard HE-AAC decoders can decode aacPlus encoded files/streams but without [http://www.codingtechnologies.com/products/sbr.htm SBR] and [http://www.codingtechnologies.com/products/paraSter.htm PS] you do not get the full quality.

* See also
** [[MP3#mp3PRO|MP3PRO]] that also uses SBR.
** [http://www.mpegsurround.com MPEG Surround] that can used for all audio but especially MP3/mp3PRO and AAC/aacPlus. [http://www.codingtechnologies.com/products/mpgsrnd.htm MPEG Surround technology] share similar characteristics with [http://www.codingtechnologies.com/products/sbr.htm SBR (Spectral Band Replication)] and [http://www.codingtechnologies.com/products/paraSter.htm PS (Parametric Stereo)], which [[MP3#mp3PRO|MP3PRO]] and AAC+ decoders also use. [http://www.divx.com DivX Inc.] is one company that uses MPEG Surround technology to achieve 5.1 channel surround sound in smaller files.

=== GAIN headers ===

See http://mp3gain.sourceforge.net and [[MP3#GAIN (MP3Gain) header|MP3 GAIN header]].

[[Category:Audio Codecs]]
[[Category: MDCT Audio Codecs]]
[[Category: Multichannel Audio Codecs]]
[[Category: Formats missing in FFmpeg]]

PCM

2008-01-17T14:02:20Z

Andoma:

PCM stands for pulse code modulation. In the context of audio coding PCM encodes an audio waveform in the time domain as a series of amplitudes.

== Basic Theory ==

'''TODO: add some basic theory and pictures explaining PCM for the uninitiated'''

== PCM Parameters ==

PCM audio is coded using a combination of various parameters.

=== Resolution/Sample Size ===

This parameter specifies the amount of data used to represent each discrete amplitude sample. The most common values are 8 bits (1 byte), which gives a range of 256 amplitude steps, or 16 bits (2 bytes), which gives a range of 65536 amplitude steps. Other sizes, such as 12, 20, and 24 bits, are occasionally seen. Some king-sized formats even opt for 32 and 64 bits per sample.

=== Byte Order ===

When more than one byte is used to represent a PCM sample, the byte order (big endian vs. little endian) must be known. Due to the widespread use of little-endian Intel CPUs, little-endian PCM tends to be the most common byte orientation.

=== Sign ===

It is not enough to know that a PCM sample is, for example, 8 bits wide. Whether the sample is signed or unsigned is needed to understand the range. If the sample is unsigned, the sample range is 0..255 with a centerpoint of 128. If the sample is signed, the sample range is -128..127 with a centerpoint of 0. If a PCM type is signed, the sign encoding is almost always 2's complement. In very rare cases, signed PCM audio is represented as a series of sign/magnitude coded numbers.

=== Channels And Interleaving ===

If the PCM type is monaural, each sample will belong to that one channel. If there is more than one channel, the channels will almost always be interleaved: Left sample, right sample, left, right, etc., in the case of stereo interleaved data. In some rare cases, usually when optimized for special playback hardware, chunks of audio destined for different channels will not be interleaved.

=== Frequency And Sample Rate ===

This parameter measures how many samples/channel are played each second. Frequency is measured in samples/second (Hz). Common frequency values include 8000, 11025, 16000, 22050, 32000, 44100, and 48000 Hz.

=== Integer Or Floating Point ===

Most PCM formats encode samples using integers. However, some applications which demand higher precision will store and process PCM samples using floating point numbers.

Floating-point PCM samples (32- or 64-bit in size) are zero-centred and varies in the interval [-1.0, 1.0], thus signed values.

== PCM Types ==

=== Linear PCM ===

=== Logarithmic PCM ===

Rather than representing sample amplitudes on a linear scale as linear PCM coding does, logarithmic PCM coding plots the amplitudes on a logarithmic scale. Log PCM is more often used in telephony and communications applications than in entertainment multimedia applications.

There are two major variants of log PCM: mu-law (u-law) and A-law. Mu-law coding uses the format number 0x07 in Microsoft multimedia files (WAV/AVI/ASF) and the fourcc 'ulaw' in Apple Quicktime files. A-law coding uses the format number 0x06 is Microsoft multimedia files and the fourcc 'alaw' in Apple Quicktime files.

Every byte of a log PCM data chunk maps to a signed 16-bit linear PCM sample. '''[TODO: Add either the conversion tables or conversion formulas]'''

=== Differential PCM ===

Values are encoded as differences between the current and the previous value. This reduces the number of bits required per audio sample by about 25% compared to PCM.

=== Adaptive DPCM ===

The size of the quantization step is varied to allow further reduction of the required bandwidth for a given signal-to-noise ratio.

== Platform-Specific PCM Identifiers And Characteristics ==

This section describes how different computing platforms store PCM audio data and any format identifiers they use.

=== DOS/Windows ===

The first widely available, PC audio card that could play back PCM audio was the Creative Labs' Sound Blaster. This drove the audio format for a lot of early audio-capable DOS applications and games. The original Sound Blaster could only play mono, unsigned 8-bit PCM data. Later Sound Blaster cards were capable of playing back 16-bit audio data. However, while these cards still played unsigned 8-bit PCM data, 16-bit data needed be signed.

Likely owing to the DOS/Intel little endian architecture, 16-bit PCM for the Sound Blaster also needs to be little endian.

Further, the original Sound Blaster was somewhat limited in the frequencies that it could support. The digital to analog conversion hardware (DAC) had to be programmed with a byte value (frequency divisor) that was processed through the following formula to yield the final playback frequency:

frequency = 1000000 / (256 - frequency_divisor)

A common divisor is 211 which yields an integer frequency of 22222 Hz, a common rate in the days of the Sound Blaster. Note that while very low frequencies (all the way down to 3921 Hz) were supported, frequencies above 45454 Hz were not.

=== Microsoft WAV/AVI/ASF Identifiers ===

Microsoft multimedia file formats such as [[Microsoft Wave|WAV]], [[Microsoft Audio/Video Interleaved|AVI]], and [[Microsoft Advanced Streaming Format|ASF]] all share the [[WAVEFORMATEX]] data structure. The structure defines, among other properties, a 16-bit little endian audio identifier. The following audio identifiers correspond to various PCM formats:
* 0x0001 denotes linear PCM
* 0x0006 denotes A-law logarithmic PCM
* 0x0007 denotes mu-law logarithmic PCM

=== Apple Macintosh ===

Native sample rates of early Apple Macintosh audio hardware included 11127 Hz and 22254 Hz. These sample rates are commonly seen in early [[QuickTime container|QuickTime]] files.

=== Apple QuickTime Identifiers ===

Audio information in [[QuickTime container|QuickTime]] files is stored along with an stsd atom that contains a [[FOURCC]] to indicate the format type. Apple QuickTime accomodates a number of different PCM formats:

* 'raw ' (need space character, ASCII 0x20, to round out FOURCC) denotes unsigned, linear PCM. 16-bit data is stored in little endian format.
* 'twos' denotes signed (i.e. twos-complement) linear PCM. 16-bit data is stored in big endian format.
* 'sowt' ('twos' spelled backwards) also denotes signed linear PCM. However, 16-bit data is stored in little endian format.
* 'in24' denotes 24-bit, big endian, linear PCM.
* 'in32' denotes 32-bit, big endian, linear PCM.
* 'fl32' denotes 32-bit floating point PCM. '''(Presumably IEEE 32-bit; byte order?)'''
* 'fl64' denotes 64-bit floating point PCM. '''(Presumably IEEE 64-bit; byte order?)'''
* 'alaw' denotes A-law logarithmic PCM.
* 'ulaw' denotes mu-law logarithmic PCM.

=== Red Book CD Audio ===

The "Red Book" defines the format of a standard audio [[Compact Disc|compact disc (CD)]]. The audio data on a standard CD consists of 16-bit linear PCM samples stored in little endian format, replayed at 44100 Hz (hence the standard term "CD-quality audio"), with left-right stereo interleaving.

=== Sega CD ===

Games made for the Sega CD, an add-on for the Sega Genesis game console, all seem to use sign-magnitude coding to store PCM information. It is a good guess that the Sega CD unit has custom hardware to play this format natively.

=== Sega Saturn ===

Games made for the Sega Saturn video game console generally seem to store PCM data as signed, 8-bit data or signed, big endian, 16-bit data. The curious property of the PCM, however, is the stereo handling. Generally, multimedia files on Sega Saturn games (most often stored using the [[Sega FILM]] format) would store a block of left channel information followed by a block of right channel information rather than interleaving left and right samples. This is likely due to custom multi-channel audio hardware in which individual channels are assigned pan positions. For playing stereo data, one channel is assigned extreme left and another is assigned extreme right. The correct samples are sent to their respective channels. Interleaved data would require deinterleaving before playback.

=== DVD 20-Bit PCM ===

=== DVD 24-Bit PCM ===

Standard DVDs can store 24-bit, signed, linear PCM. At the very least, the data can be stereo. It is unknown if other channel configurations are supported.

Stereo 24-bit linear PCM is stored in blocks of 12 bytes which pack 4 interleaved stereo samples, denoted here as L0, R0, L1, R1. 24 bits/sample also means 3 bytes/sample. The 3 bytes are denoted here as top, middle, and bottom:

* T = top byte = bits 23..16
* M = middle byte = bits 15..8
* B = bottom byte = bits 7..0

[[Image:Dvd-24bit-pcm.png]]

The first 8 bytes of the block contain the lower 16 bits of each of the 4 samples encoded in big endian format. The last 4 bytes represent the top 8 bits for each of the 4 samples.

== Identifying PCM Data ==

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

'''(TODO: add explanation and example for identifying PCM data in a hex dump)'''

DTS

2008-01-17T14:01:21Z

Andoma:

* Codec ID: 0x2001
* 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 through the [http://pda.etsi.org/pda/queryform.asp ETSI] ([[Mirrored Files|mirrored here]]), search for DTS. Info completing the incomplete specification can be found here: http://webapp.etsi.org/action%5CPU/20021224/ts_102114v010201p.pdf, http://gauss.ffii.org/PatentView/EP864146, and http://ofi.epoline.org/view/GetDossier?dosnum=&pubnum=EP864146&lang=EN# .

The [[VideoLAN]] project has created [http://developers.videolan.org/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.

=== 14-bit words ===
This kind of bitstream is packed into 16-bit words where high two bits contain some auxiliary information (parity?). This coding is used in many DTS-in-[[WAV]] files.

=== 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
|}

== 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 ||
|-
| '''TODO'''
|}
[[Category:Audio Codecs]]
[[Category: QMF Audio Codecs]]
[[Category: Multichannel Audio Codecs]]

Category:Multichannel Audio Codecs

2008-01-17T14:00:03Z

Andoma:

List of all audio codecs that can deliver multichannel (more than 2 channels)

A52

2008-01-17T13:59:02Z

Andoma:

* Codec ID: 0x2000
* Specification for A/52 (revision B): http://www.atsc.org/standards/a_52b.pdf ([[Mirrored Files|mirrored]])

==AC3==
ATSC A/52a is a standard for lossy encoding of audio in digital television broadcasting in the United States. It is the same as Dolby Digital AC3.

==EAC3==
ATSC A/52b is an extension to A/52a and was approved for the [[Blu-ray]] and [[HD DVD]] standard. It is the same as Enhanced AC3 or EAC3 for short. EAC3 is also supposed to be backwards compatible with AC3.

==See also==
* DTS (DTS Coherent Acoustics)

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

User:Andoma

2007-12-04T07:43:12Z

Andoma:

Minor involvement in ffmpeg.

Also working on [http://www.lonelycoder.com/hts this] mediaplayer.

FFmpeg / Libav Summer Of Code

2007-12-04T07:41:50Z

Andoma: /* AAC Decoder */

The [[FFmpeg]] project has been a participant in the [http://code.google.com/soc/ Google Summer of Code] program during the 2006 and 2007 seaons.

* [[FFmpeg Summer Of Code 2006|2006 project page]]
* [[FFmpeg Summer Of Code 2007|2007 project page]]

Each accepted project is developed in its own sandbox, separate from the main FFmpeg codebase. Naturally, the end goal of each of the accepted FFmpeg projects ought to be to have that code in shape for acceptance into the production codebase. This page tracks the status of each project and how well each student did.

== 2006 Projects ==

=== VC-1 Decoder ===
* Student: [[User:Kostya|Kostya Shishkov]]
* Mentor: [[User:Multimedia Mike|Mike Melanson]]
* Student Status: FFmpeg committer
* Code Status: Accepted into the FFmpeg codebase.

=== AMR Decoder ===
* Student: [[User:superdump|Robert Swain]]
* Mentor: [[User:Merbanan|Benjamin Larsson]]
* Student Status: project not finished during SoC but continues working on it
* Code Status: To be expected soon. (Within a few weeks from 20071203 I think.) Narrow band decoding documented on [[AMR-NB]] and floating point code has been implemented up to synthesis.

=== AC3 Decoder ===
* Student: [[User:Cloud9|Kartikey Mahendra BHATT]]
* Mentor: [[User:Merbanan|Benjamin Larsson]]
* Student Status: disappeared, project unfinished
* Code Status: Picked up by [[User:Jruggle|Justin Ruggles]] and committed to FFmpeg.

=== AAC Decoder ===
* Student: Maxim Gavrilov
* Mentor: [[User:ods15|Oded Shimon]]
* Student Status: disappeared, project unfinished
* Code Status: Picked up by [[User:andoma|Andreas Öman]] who is currently preparing code for merge with ffmpeg.

=== Vorbis Encoder ===
* Student: Mathew Philip
* Mentor: [[User:ods15|Oded Shimon]]
* Student Status: disappeared, project barely started
* Code Status: Picked up by [[User:ods15|Oded Shimon]] and committed to FFmpeg.

== 2007 Projects ==

=== RealVideo 4 Decoder ===
* Student: [[User:Kostya|Kostya Shishkov]]
* Mentor: [[User:Multimedia Mike|Mike Melanson]]
* Student Status: FFmpeg committer
* Code Status: in the process of being committed to FFmpeg; the project goal has also morphed to include a RealVideo 3 decoder since the 2 schemes are so similar. Both RV30 and RV40 are decodeable with visual artifacts.

=== QCELP Decoder ===
* Student: [[User:Reynaldo|Reynaldo Verdejo Pinochet]]
* Mentor: [[User:Merbanan|Benjamin Larsson]]
* Student Status: project not finished during SoC but continues working on it
* Code Status: slowly progressing, it's working though

=== Matroska Muxer ===
* Student: David Conrad
* Mentor: [[User:aurel|Aurélien Jacobs]]
* Student Status: FFmpeg committer
* Code Status: Accepted into the FFmpeg codebase.

=== Video Filter API (AKA [[Libavfilter|libavfilter]]) ===
* Student: [[User:Koorogi|Bobby Bingham]]
* Mentor: [[User:Merbanan|Benjamin Larsson]] and Michael Niedermayer
* Student Status: project not finished during SoC but continues working on it
* Code Status: Working code for ffplay and ffmpeg. Colorspace negotiation needs some work. Still in development (albeit slowly) by [[User:Koorogi|Bobby Bingham]] and [[User:Vitor|Vitor]].

=== E-AC3 Decoder ===
* Student: Bartlomiej Wolowiec
* Mentor: [[User:Jruggle|Justin Ruggles]]
* Student Status: project not finished during SoC, (continues working on it?)
* Code Status: Working for most available samples. There are still some unimplemented features though. The code is currently not clean enough for inclusion in FFmpeg.

=== JPEG 2000 Encoder and Decoder ===
* Student: Kamil Nowosad
* Mentor: [[User:pengvado|Loren Merritt]]
* Student Status: disappeared, project unfinished
* Code Status: The code is working but not all features are supported.

=== Dirac Encoder and Decoder ===
* Student: Marco Gerards
* Mentor: [[User:Lu_zero|Luca Barbato]]
* Student Status: project not finished during SoC but continues working on it, just slower than before due to other tasks taking priority.
* Code Status: The decoder is in good shape, the encoder still needs more work.

=== TS Muxer ===
* Student: Xiaohui Sun
* Mentor: [[User:bcoudurier|Baptiste Coudurier]]
* Student Status: disappeared, project unfinished
* Code Status: [[Interesting Patches#PES packetizer by Xiaohui Sun|Changes]] requested during the review process for FFmpeg inclusion were never made.

[[Category:FFmpeg]]