MPEG-1

MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively)[1] without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) possible.[2][3]

Today, MPEG-1 has become the most widely compatible lossy audio/video format in the world, and is used in a large number of products and technologies. Perhaps the best-known part of the MPEG-1 standard is the MP3 audio format it introduced.

The MPEG-1 standard is published as ISO/IEC 11172 – Information technology—Coding of moving pictures and associated audio for digital storage media at up to about 1.5 Mbit/s.
The standard consists of the following five Parts:[4][5][6][7][8]

  1. Systems (storage and synchronization of video, audio, and other data together)
  2. Video (compressed video content)
  3. Audio (compressed audio content)
  4. Conformance testing (testing the correctness of implementations of the standard)
  5. Reference software (example software showing how to encode and decode according to the standard)
Moving Picture Experts Group Phase 1 (MPEG-1)
Filename extension.mpg, .mpeg, .mp1, .mp2, .mp3, .m1v, .m1a,.m2a, .mpa, .mpv
Internet media typeaudio/mpeg, video/mpeg
Developed byISO, IEC
Initial releasecreated 1988–92
Type of formataudio, video, container
Extended fromJPEG, H.261
Extended toMPEG-2
StandardISO/IEC 11172

History

Modeled on the successful collaborative approach and the compression technologies developed by the Joint Photographic Experts Group and CCITT's Experts Group on Telephony (creators of the JPEG image compression standard and the H.261 standard for video conferencing respectively), the Moving Picture Experts Group (MPEG) working group was established in January 1988. MPEG was formed to address the need for standard video and audio formats, and to build on H.261 to get better quality through the use of more complex encoding methods.[2][9][10] It was established in 1988 by the initiative of Hiroshi Yasuda (Nippon Telegraph and Telephone) and Leonardo Chiariglione.[11]

Development of the MPEG-1 standard began in May 1988. Fourteen video and fourteen audio codec proposals were submitted by individual companies and institutions for evaluation. The codecs were extensively tested for computational complexity and subjective (human perceived) quality, at data rates of 1.5 Mbit/s. This specific bitrate was chosen for transmission over T-1/E-1 lines and as the approximate data rate of audio CDs.[12] The codecs that excelled in this testing were utilized as the basis for the standard and refined further, with additional features and other improvements being incorporated in the process.[13]

After 20 meetings of the full group in various cities around the world, and 4½ years of development and testing, the final standard (for parts 1–3) was approved in early November 1992 and published a few months later.[14] The reported completion date of the MPEG-1 standard varies greatly: a largely complete draft standard was produced in September 1990, and from that point on, only minor changes were introduced.[2] The draft standard was publicly available for purchase.[15] The standard was finished with the 6 November 1992 meeting.[16] The Berkeley Plateau Multimedia Research Group developed an MPEG-1 decoder in November 1992.[17] In July 1990, before the first draft of the MPEG-1 standard had even been written, work began on a second standard, MPEG-2,[18] intended to extend MPEG-1 technology to provide full broadcast-quality video (as per CCIR 601) at high bitrates (3–15  Mbit/s) and support for interlaced video.[19] Due in part to the similarity between the two codecs, the MPEG-2 standard includes full backwards compatibility with MPEG-1 video, so any MPEG-2 decoder can play MPEG-1 videos.[20]

Notably, the MPEG-1 standard very strictly defines the bitstream, and decoder function, but does not define how MPEG-1 encoding is to be performed, although a reference implementation is provided in ISO/IEC-11172-5.[1] This means that MPEG-1 coding efficiency can drastically vary depending on the encoder used, and generally means that newer encoders perform significantly better than their predecessors.[21] The first three parts (Systems, Video and Audio) of ISO/IEC 11172 were published in August 1993.[22]

MPEG-1 Parts[8][23]
Part Number First public release date (First edition) Latest correction Title Description
Part 1 ISO/IEC 11172-1 1993 1999[24] Systems
Part 2 ISO/IEC 11172-2 1993 2006[25] Video
Part 3 ISO/IEC 11172-3 1993 1996[26] Audio
Part 4 ISO/IEC 11172-4 1995 2007[27] Compliance testing
Part 5 ISO/IEC TR 11172-5 1998 2007[28] Software simulation

Patents

All widely known patent searches suggest that, due to its age, MPEG-1 video and Layer I/II audio is no longer covered by any patents and can thus be used without obtaining a licence or paying any fees.[29][30][31][32][33] The ISO patent database lists one patent for ISO 11172, US 4,472,747, which expired in 2003.[34] The near-complete draft of the MPEG-1 standard was publicly available as ISO CD 11172[15] by December 6, 1991.[35] Neither the July 2008 Kuro5hin article "Patent Status of MPEG-1, H.261 and MPEG-2",[36] nor an August 2008 thread on the gstreamer-devel[37] mailing list were able to list a single unexpired MPEG-1 video and Layer I/II audio patent. A May 2009 discussion on the whatwg mailing list mentioned US 5,214,678 patent as possibly covering MPEG audio layer II.[38] Filed in 1990 and published in 1993, this patent is now expired.[39]

A full MPEG-1 decoder and encoder, with "Layer 3 audio", could not be implemented royalty free since there were companies that required patent fees for implementations of MPEG-1 Layer 3 Audio as discussed in the MP3 article. All patents in the world connected to MP3 expired 30 December 2017, which makes this format totally free for use. Despite this as early as on 23 April 2017 Fraunhofer IIS stopped charging for Technicolor's mp3 licensing program for certain mp3 related patents and software.[40]

Applications

  • Most popular software for video playback includes MPEG-1 decoding, in addition to any other supported formats.
  • The popularity of MP3 audio has established a massive installed base of hardware that can play back MPEG-1 Audio (all three layers).
  • "Virtually all digital audio devices" can play back MPEG-1 Audio.[41] Many millions have been sold to-date.
  • Before MPEG-2 became widespread, many digital satellite/cable TV services used MPEG-1 exclusively.[10][21]
  • The widespread popularity of MPEG-2 with broadcasters means MPEG-1 is playable by most digital cable and satellite set-top boxes, and digital disc and tape players, due to backwards compatibility.
  • MPEG-1 was used for full-screen video on Green Book CD-i, and on Video CD (VCD).
  • The Super Video CD standard, based on VCD, uses MPEG-1 audio exclusively, as well as MPEG-2 video.
  • The DVD-Video format uses MPEG-2 video primarily, but MPEG-1 support is explicitly defined in the standard.
  • The DVD-Video standard originally required MPEG-1 Layer II audio for PAL countries, but was changed to allow AC-3/Dolby Digital-only discs. MPEG-1 Layer II audio is still allowed on DVDs, although newer extensions to the format, like MPEG Multichannel, are rarely supported.
  • Most DVD players also support Video CD and MP3 CD playback, which use MPEG-1.
  • The international Digital Video Broadcasting (DVB) standard primarily uses MPEG-1 Layer II audio, and MPEG-2 video.
  • The international Digital Audio Broadcasting (DAB) standard uses MPEG-1 Layer II audio exclusively, due to MP2's especially high quality, modest decoder performance requirements, and tolerance of errors.

Part 1: Systems

Part 1 of the MPEG-1 standard covers systems, and is defined in ISO/IEC-11172-1.

MPEG-1 Systems specifies the logical layout and methods used to store the encoded audio, video, and other data into a standard bitstream, and to maintain synchronization between the different contents. This file format is specifically designed for storage on media, and transmission over communication channels, that are considered relatively reliable. Only limited error protection is defined by the standard, and small errors in the bitstream may cause noticeable defects.

This structure was later named an MPEG program stream: "The MPEG-1 Systems design is essentially identical to the MPEG-2 Program Stream structure."[42] This terminology is more popular, precise (differentiates it from an MPEG transport stream) and will be used here.

Elementary streams

Elementary Streams (ES) are the raw bitstreams of MPEG-1 audio and video encoded data (output from an encoder). These files can be distributed on their own, such as is the case with MP3 files.

Packetized Elementary Streams (PES) are elementary streams packetized into packets of variable lengths, i.e., divided ES into independent chunks where cyclic redundancy check (CRC) checksum was added to each packet for error detection.

System Clock Reference (SCR) is a timing value stored in a 33-bit header of each PES, at a frequency/precision of 90 kHz, with an extra 9-bit extension that stores additional timing data with a precision of 27 MHz.[43][44] These are inserted by the encoder, derived from the system time clock (STC). Simultaneously encoded audio and video streams will not have identical SCR values, however, due to buffering, encoding, jitter, and other delay.

Program streams

Program Streams (PS) are concerned with combining multiple packetized elementary streams (usually just one audio and video PES) into a single stream, ensuring simultaneous delivery, and maintaining synchronization. The PS structure is known as a multiplex, or a container format.

Presentation time stamps (PTS) exist in PS to correct the inevitable disparity between audio and video SCR values (time-base correction). 90 kHz PTS values in the PS header tell the decoder which video SCR values match which audio SCR values.[43] PTS determines when to display a portion of an MPEG program, and is also used by the decoder to determine when data can be discarded from the buffer.[45] Either video or audio will be delayed by the decoder until the corresponding segment of the other arrives and can be decoded.

PTS handling can be problematic. Decoders must accept multiple program streams that have been concatenated (joined sequentially). This causes PTS values in the middle of the video to reset to zero, which then begin incrementing again. Such PTS wraparound disparities can cause timing issues that must be specially handled by the decoder.

Decoding Time Stamps (DTS), additionally, are required because of B-frames. With B-frames in the video stream, adjacent frames have to be encoded and decoded out-of-order (re-ordered frames). DTS is quite similar to PTS, but instead of just handling sequential frames, it contains the proper time-stamps to tell the decoder when to decode and display the next B-frame (types of frames explained below), ahead of its anchor (P- or I-) frame. Without B-frames in the video, PTS and DTS values are identical.[46]

Multiplexing

To generate the PS, the multiplexer will interleave the (two or more) packetized elementary streams. This is done so the packets of the simultaneous streams can be transferred over the same channel and are guaranteed to both arrive at the decoder at precisely the same time. This is a case of time-division multiplexing.

Determining how much data from each stream should be in each interleaved segment (the size of the interleave) is complicated, yet an important requirement. Improper interleaving will result in buffer underflows or overflows, as the receiver gets more of one stream than it can store (e.g. audio), before it gets enough data to decode the other simultaneous stream (e.g. video). The MPEG Video Buffering Verifier (VBV) assists in determining if a multiplexed PS can be decoded by a device with a specified data throughput rate and buffer size.[47] This offers feedback to the muxer and the encoder, so that they can change the mux size or adjust bitrates as needed for compliance.

Part 2: Video

Part 2 of the MPEG-1 standard covers video and is defined in ISO/IEC-11172-2. The design was heavily influenced by H.261.

MPEG-1 Video exploits perceptual compression methods to significantly reduce the data rate required by a video stream. It reduces or completely discards information in certain frequencies and areas of the picture that the human eye has limited ability to fully perceive. It also exploits temporal (over time) and spatial (across a picture) redundancy common in video to achieve better data compression than would be possible otherwise. (See: Video compression)

Color space

Yuvformats420sampling
Example of 4:2:0 subsampling. The two overlapping center circles represent chroma blue and chroma red (color) pixels, while the 4 outside circles represent the luma (brightness).

Before encoding video to MPEG-1, the color-space is transformed to Y'CbCr (Y'=Luma, Cb=Chroma Blue, Cr=Chroma Red). Luma (brightness, resolution) is stored separately from chroma (color, hue, phase) and even further separated into red and blue components. The chroma is also subsampled to 4:2:0, meaning it is reduced by one half vertically and one half horizontally, to just one quarter the resolution of the video.[1] This software algorithm also has analogies in hardware, such as the output from a Bayer pattern filter, common in digital colour cameras.

Because the human eye is much more sensitive to small changes in brightness (the Y component) than in color (the Cr and Cb components), chroma subsampling is a very effective way to reduce the amount of video data that needs to be compressed. On videos with fine detail (high spatial complexity) this can manifest as chroma aliasing artifacts. Compared to other digital compression artifacts, this issue seems to be very rarely a source of annoyance.

Because of subsampling, Y'CbCr video must always be stored using even dimensions (divisible by 2), otherwise chroma mismatch ("ghosts") will occur, and it will appear as if the color is ahead of, or behind the rest of the video, much like a shadow.

Y'CbCr is often inaccurately called YUV which is only used in the domain of analog video signals. Similarly, the terms luminance and chrominance are often used instead of the (more accurate) terms luma and chroma.

Resolution/bitrate

MPEG-1 supports resolutions up to 4095×4095 (12-bits), and bitrates up to 100 Mbit/s.[10]

MPEG-1 videos are most commonly seen using Source Input Format (SIF) resolution: 352x240, 352x288, or 320x240. These low resolutions, combined with a bitrate less than 1.5 Mbit/s, make up what is known as a constrained parameters bitstream (CPB), later renamed the "Low Level" (LL) profile in MPEG-2. This is the minimum video specifications any decoder should be able to handle, to be considered MPEG-1 compliant. This was selected to provide a good balance between quality and performance, allowing the use of reasonably inexpensive hardware of the time.[2][10]

Frame/picture/block types

MPEG-1 has several frame/picture types that serve different purposes. The most important, yet simplest, is I-frame.

I-frames

I-frame is an abbreviation for Intra-frame, so-called because they can be decoded independently of any other frames. They may also be known as I-pictures, or keyframes due to their somewhat similar function to the key frames used in animation. I-frames can be considered effectively identical to baseline JPEG images.[10]

High-speed seeking through an MPEG-1 video is only possible to the nearest I-frame. When cutting a video it is not possible to start playback of a segment of video before the first I-frame in the segment (at least not without computationally intensive re-encoding). For this reason, I-frame-only MPEG videos are used in editing applications.

I-frame only compression is very fast, but produces very large file sizes: a factor of 3× (or more) larger than normally encoded MPEG-1 video, depending on how temporally complex a specific video is.[2] I-frame only MPEG-1 video is very similar to MJPEG video. So much so that very high-speed and theoretically lossless (in reality, there are rounding errors) conversion can be made from one format to the other, provided a couple of restrictions (color space and quantization matrix) are followed in the creation of the bitstream.[48]

The length between I-frames is known as the group of pictures (GOP) size. MPEG-1 most commonly uses a GOP size of 15-18. i.e. 1 I-frame for every 14-17 non-I-frames (some combination of P- and B- frames). With more intelligent encoders, GOP size is dynamically chosen, up to some pre-selected maximum limit.[10]

Limits are placed on the maximum number of frames between I-frames due to decoding complexing, decoder buffer size, recovery time after data errors, seeking ability, and accumulation of IDCT errors in low-precision implementations most common in hardware decoders (See: IEEE-1180).

P-frames

P-frame is an abbreviation for Predicted-frame. They may also be called forward-predicted frames, or inter-frames (B-frames are also inter-frames).

P-frames exist to improve compression by exploiting the temporal (over time) redundancy in a video. P-frames store only the difference in image from the frame (either an I-frame or P-frame) immediately preceding it (this reference frame is also called the anchor frame).

The difference between a P-frame and its anchor frame is calculated using motion vectors on each macroblock of the frame (see below). Such motion vector data will be embedded in the P-frame for use by the decoder.

A P-frame can contain any number of intra-coded blocks, in addition to any forward-predicted blocks.[49]

If a video drastically changes from one frame to the next (such as a cut), it is more efficient to encode it as an I-frame.

B-frames

B-frame stands for bidirectional-frame. They may also be known as backwards-predicted frames or B-pictures. B-frames are quite similar to P-frames, except they can make predictions using both the previous and future frames (i.e. two anchor frames).

It is therefore necessary for the player to first decode the next I- or P- anchor frame sequentially after the B-frame, before the B-frame can be decoded and displayed. This means decoding B-frames requires larger data buffers and causes an increased delay on both decoding and during encoding. This also necessitates the decoding time stamps (DTS) feature in the container/system stream (see above). As such, B-frames have long been subject of much controversy, they are often avoided in videos, and are sometimes not fully supported by hardware decoders.

No other frames are predicted from a B-frame. Because of this, a very low bitrate B-frame can be inserted, where needed, to help control the bitrate. If this was done with a P-frame, future P-frames would be predicted from it and would lower the quality of the entire sequence. However, similarly, the future P-frame must still encode all the changes between it and the previous I- or P- anchor frame. B-frames can also be beneficial in videos where the background behind an object is being revealed over several frames, or in fading transitions, such as scene changes.[2][10]

A B-frame can contain any number of intra-coded blocks and forward-predicted blocks, in addition to backwards-predicted, or bidirectionally predicted blocks.[10][49]

D-frames

MPEG-1 has a unique frame type not found in later video standards. D-frames or DC-pictures are independent images (intra-frames) that have been encoded using DC transform coefficients only (AC coefficients are removed when encoding D-frames—see DCT below) and hence are very low quality. D-frames are never referenced by I-, P- or B- frames. D-frames are only used for fast previews of video, for instance when seeking through a video at high speed.[2]

Given moderately higher-performance decoding equipment, fast preview can be accomplished by decoding I-frames instead of D-frames. This provides higher quality previews, since I-frames contain AC coefficients as well as DC coefficients. If the encoder can assume that rapid I-frame decoding capability is available in decoders, it can save bits by not sending D-frames (thus improving compression of the video content). For this reason, D-frames are seldom actually used in MPEG-1 video encoding, and the D-frame feature has not been included in any later video coding standards.

Macroblocks

MPEG-1 operates on video in a series of 8x8 blocks for quantization. However, because chroma (color) is subsampled by a factor of 4, each pair of (red and blue) chroma blocks corresponds to 4 different luma blocks. This set of 6 blocks, with a resolution of 16x16, is called a macroblock.

A macroblock is the smallest independent unit of (color) video. Motion vectors (see below) operate solely at the macroblock level.

If the height or width of the video are not exact multiples of 16, full rows and full columns of macroblocks must still be encoded and decoded to fill out the picture (though the extra decoded pixels are not displayed).

Motion vectors

To decrease the amount of temporal redundancy in a video, only blocks that change are updated, (up to the maximum GOP size). This is known as conditional replenishment. However, this is not very effective by itself. Movement of the objects, and/or the camera may result in large portions of the frame needing to be updated, even though only the position of the previously encoded objects has changed. Through motion estimation the encoder can compensate for this movement and remove a large amount of redundant information.

The encoder compares the current frame with adjacent parts of the video from the anchor frame (previous I- or P- frame) in a diamond pattern, up to a (encoder-specific) predefined radius limit from the area of the current macroblock. If a match is found, only the direction and distance (i.e. the vector of the motion) from the previous video area to the current macroblock need to be encoded into the inter-frame (P- or B- frame). The reverse of this process, performed by the decoder to reconstruct the picture, is called motion compensation.

A predicted macroblock rarely matches the current picture perfectly, however. The differences between the estimated matching area, and the real frame/macroblock is called the prediction error. The larger the error, the more data must be additionally encoded in the frame. For efficient video compression, it is very important that the encoder is capable of effectively and precisely performing motion estimation.

Motion vectors record the distance between two areas on screen based on the number of pixels (called pels). MPEG-1 video uses a motion vector (MV) precision of one half of one pixel, or half-pel. The finer the precision of the MVs, the more accurate the match is likely to be, and the more efficient the compression. There are trade-offs to higher precision, however. Finer MVs result in larger data size, as larger numbers must be stored in the frame for every single MV, increased coding complexity as increasing levels of interpolation on the macroblock are required for both the encoder and decoder, and diminishing returns (minimal gains) with higher precision MVs. Half-pel was chosen as the ideal trade-off. (See: qpel)

Because neighboring macroblocks are likely to have very similar motion vectors, this redundant information can be compressed quite effectively by being stored DPCM-encoded. Only the (smaller) amount of difference between the MVs for each macroblock needs to be stored in the final bitstream.

P-frames have one motion vector per macroblock, relative to the previous anchor frame. B-frames, however, can use two motion vectors; one from the previous anchor frame, and one from the future anchor frame.[49]

Partial macroblocks, and black borders/bars encoded into the video that do not fall exactly on a macroblock boundary, cause havoc with motion prediction. The block padding/border information prevents the macroblock from closely matching with any other area of the video, and so, significantly larger prediction error information must be encoded for every one of the several dozen partial macroblocks along the screen border. DCT encoding and quantization (see below) also isn't nearly as effective when there is large/sharp picture contrast in a block.

An even more serious problem exists with macroblocks that contain significant, random, edge noise, where the picture transitions to (typically) black. All the above problems also apply to edge noise. In addition, the added randomness is simply impossible to compress significantly. All of these effects will lower the quality (or increase the bitrate) of the video substantially.

DCT

Each 8x8 block is encoded by first applying a forward discrete cosine transform (FDCT) and then a quantization process. The FDCT process (by itself) is theoretically lossless, and can be reversed by applying an Inverse DCT (IDCT) to reproduce the original values (in the absence of any quantization and rounding errors). In reality, there are some (sometimes large) rounding errors introduced both by quantization in the encoder (as described in the next section) and by IDCT approximation error in the decoder. The minimum allowed accuracy of a decoder IDCT approximation is defined by ISO/IEC 23002-1. (Prior to 2006, it was specified by IEEE 1180-1990.)

The FDCT process converts the 8x8 block of uncompressed pixel values (brightness or color difference values) into an 8x8 indexed array of frequency coefficient values. One of these is the (statistically high in variance) DC coefficient, which represents the average value of the entire 8x8 block. The other 63 coefficients are the statistically smaller AC coefficients, which are positive or negative values each representing sinusoidal deviations from the flat block value represented by the DC coefficient.

An example of an encoded 8x8 FDCT block:

Since the DC coefficient value is statistically correlated from one block to the next, it is compressed using DPCM encoding. Only the (smaller) amount of difference between each DC value and the value of the DC coefficient in the block to its left needs to be represented in the final bitstream.

Additionally, the frequency conversion performed by applying the DCT provides a statistical decorrelation function to efficiently concentrate the signal into fewer high-amplitude values prior to applying quantization (see below).

Quantization

Quantization (of digital data) is, essentially, the process of reducing the accuracy of a signal, by dividing it into some larger step size (i.e. finding the nearest multiple, and discarding the remainder/modulus).

The frame-level quantizer is a number from 0 to 31 (although encoders will usually omit/disable some of the extreme values) which determines how much information will be removed from a given frame. The frame-level quantizer is either dynamically selected by the encoder to maintain a certain user-specified bitrate, or (much less commonly) directly specified by the user.

Contrary to popular belief, a fixed frame-level quantizer (set by the user) does not deliver a constant level of quality. Instead, it is an arbitrary metric that will provide a somewhat varying level of quality, depending on the contents of each frame. Given two files of identical sizes, the one encoded at an average bitrate should look better than the one encoded with a fixed quantizer (variable bitrate). Constant quantizer encoding can be used, however, to accurately determine the minimum and maximum bitrates possible for encoding a given video.

A quantization matrix is a string of 64-numbers (0-255) which tells the encoder how relatively important or unimportant each piece of visual information is. Each number in the matrix corresponds to a certain frequency component of the video image.

An example quantization matrix:

Quantization is performed by taking each of the 64 frequency values of the DCT block, dividing them by the frame-level quantizer, then dividing them by their corresponding values in the quantization matrix. Finally, the result is rounded down. This significantly reduces, or completely eliminates, the information in some frequency components of the picture. Typically, high frequency information is less visually important, and so high frequencies are much more strongly quantized (drastically reduced). MPEG-1 actually uses two separate quantization matrices, one for intra-blocks (I-blocks) and one for inter-block (P- and B- blocks) so quantization of different block types can be done independently, and so, more effectively.[2]

This quantization process usually reduces a significant number of the AC coefficients to zero, (known as sparse data) which can then be more efficiently compressed by entropy coding (lossless compression) in the next step.

An example quantized DCT block:

Quantization eliminates a large amount of data, and is the main lossy processing step in MPEG-1 video encoding. This is also the primary source of most MPEG-1 video compression artifacts, like blockiness, color banding, noise, ringing, discoloration, et al. This happens when video is encoded with an insufficient bitrate, and the encoder is therefore forced to use high frame-level quantizers (strong quantization) through much of the video.

Entropy coding

Several steps in the encoding of MPEG-1 video are lossless, meaning they will be reversed upon decoding, to produce exactly the same (original) values. Since these lossless data compression steps don't add noise into, or otherwise change the contents (unlike quantization), it is sometimes referred to as noiseless coding.[41] Since lossless compression aims to remove as much redundancy as possible, it is known as entropy coding in the field of information theory.

The coefficients of quantized DCT blocks tend to zero towards the bottom-right. Maximum compression can be achieved by a zig-zag scanning of the DCT block starting from the top left and using Run-length encoding techniques.

The DC coefficients and motion vectors are DPCM-encoded.

Run-length encoding (RLE) is a very simple method of compressing repetition. A sequential string of characters, no matter how long, can be replaced with a few bytes, noting the value that repeats, and how many times. For example, if someone were to say "five nines", you would know they mean the number: 99999.

RLE is particularly effective after quantization, as a significant number of the AC coefficients are now zero (called sparse data), and can be represented with just a couple of bytes. This is stored in a special 2-dimensional Huffman table that codes the run-length and the run-ending character.

Huffman Coding is a very popular method of entropy coding, and used in MPEG-1 video to reduce the data size. The data is analyzed to find strings that repeat often. Those strings are then put into a special table, with the most frequently repeating data assigned the shortest code. This keeps the data as small as possible with this form of compression.[41] Once the table is constructed, those strings in the data are replaced with their (much smaller) codes, which reference the appropriate entry in the table. The decoder simply reverses this process to produce the original data.

This is the final step in the video encoding process, so the result of Huffman coding is known as the MPEG-1 video "bitstream."

GOP configurations for specific applications

I-frames store complete frame info within the frame and are therefore suited for random access. P-frames provide compression using motion vectors relative to the previous frame ( I or P ). B-frames provide maximum compression but require the previous as well as next frame for computation. Therefore, processing of B-frames requires more buffer on the decoded side. A configuration of the Group of Pictures (GOP) should be selected based on these factors. I-frame only sequences give least compression, but are useful for random access, FF/FR and editability. I- and P-frame sequences give moderate compression but add a certain degree of random access, FF/FR functionality. I-, P- and B-frame sequences give very high compression but also increase the coding/decoding delay significantly. Such configurations are therefore not suited for video-telephony or video-conferencing applications.

The typical data rate of an I-frame is 1 bit per pixel while that of a P-frame is 0.1 bit per pixel and for a B-frame, 0.015 bit per pixel.[50]

Part 3: Audio

Part 3 of the MPEG-1 standard covers audio and is defined in ISO/IEC-11172-3.

MPEG-1 Audio utilizes psychoacoustics to significantly reduce the data rate required by an audio stream. It reduces or completely discards certain parts of the audio that it deduces that the human ear can't hear, either because they are in frequencies where the ear has limited sensitivity, or are masked by other (typically louder) sounds.[51]

Channel Encoding:

  • Mono
  • Joint Stereo – intensity encoded
  • Joint Stereo – M/S encoded for Layer 3 only
  • Stereo
  • Dual (two uncorrelated mono channels)
  • Sampling rates: 32000, 44100, and 48000 Hz
  • Bitrates for Layer I: 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416 and 448 kbit/s[52]
  • Bitrates for Layer II: 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 and 384 kbit/s
  • Bitrates for Layer III: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s

MPEG-1 Audio is divided into 3 layers. Each higher layer is more computationally complex, and generally more efficient at lower bitrates than the previous.[10] The layers are semi backwards compatible as higher layers reuse technologies implemented by the lower layers. A "Full" Layer II decoder can also play Layer I audio, but not Layer III audio, although not all higher level players are "full".[51]

Layer I

MPEG-1 Layer I is nothing more than a simplified version of Layer II.[13] Layer I uses a smaller 384-sample frame size for very low delay, and finer resolution.[21] This is advantageous for applications like teleconferencing, studio editing, etc. It has lower complexity than Layer II to facilitate real-time encoding on the hardware available circa 1990.[41]

Layer I saw limited adoption in its time, and most notably was used on Philips' defunct Digital Compact Cassette at a bitrate of 384 kbit/s.[1] With the substantial performance improvements in digital processing since its introduction, Layer I quickly became unnecessary and obsolete.

Layer I audio files typically use the extension .mp1 or sometimes .m1a

Layer II

MPEG-1 Layer II (MP2—often incorrectly called MUSICAM)[51] is a lossy audio format designed to provide high quality at about 192 kbit/s for stereo sound. Decoding MP2 audio is computationally simple, relative to MP3, AAC, etc.

History/MUSICAM

MPEG-1 Layer II was derived from the MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and Institut für Rundfunktechnik (IRT/CNET)[10][13][53] as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of digital audio broadcasting.

Most key features of MPEG-1 Audio were directly inherited from MUSICAM, including the filter bank, time-domain processing, audio frame sizes, etc. However, improvements were made, and the actual MUSICAM algorithm was not used in the final MPEG-1 Layer II audio standard. The widespread usage of the term MUSICAM to refer to Layer II is entirely incorrect and discouraged for both technical and legal reasons.[51]

Technical details

Layer II/MP2 is a time-domain encoder. It uses a low-delay 32 sub-band polyphased filter bank for time-frequency mapping; having overlapping ranges (i.e. polyphased) to prevent aliasing.[54] The psychoacoustic model is based on the principles of auditory masking, simultaneous masking effects, and the absolute threshold of hearing (ATH). The size of a Layer II frame is fixed at 1152-samples (coefficients).

Time domain refers to how analysis and quantization is performed on short, discrete samples/chunks of the audio waveform. This offers low delay as only a small number of samples are analyzed before encoding, as opposed to frequency domain encoding (like MP3) which must analyze many times more samples before it can decide how to transform and output encoded audio. This also offers higher performance on complex, random and transient impulses (such as percussive instruments, and applause), offering avoidance of artifacts like pre-echo.

The 32 sub-band filter bank returns 32 amplitude coefficients, one for each equal-sized frequency band/segment of the audio, which is about 700 Hz wide (depending on the audio's sampling frequency). The encoder then utilizes the psychoacoustic model to determine which sub-bands contain audio information that is less important, and so, where quantization will be inaudible, or at least much less noticeable.[41]

Fft-2
Example FFT analysis on an audio wave sample.

The psychoacoustic model is applied using a 1024-point Fast Fourier Transform (FFT). Of the 1152 samples per frame, 64 samples at the top and bottom of the frequency range are ignored for this analysis. They are presumably not significant enough to change the result. The psychoacoustic model uses an empirically determined masking model to determine which sub-bands contribute more to the masking threshold, and how much quantization noise each can contain without being perceived. Any sounds below the absolute threshold of hearing (ATH) are completely discarded. The available bits are then assigned to each sub-band accordingly.[51][54]

Typically, sub-bands are less important if they contain quieter sounds (smaller coefficient) than a neighboring (i.e. similar frequency) sub-band with louder sounds (larger coefficient). Also, "noise" components typically have a more significant masking effect than "tonal" components.[53]

Less significant sub-bands are reduced in accuracy by quantization. This basically involves compressing the frequency range (amplitude of the coefficient), i.e. raising the noise floor. Then computing an amplification factor, for the decoder to use to re-expand each sub-band to the proper frequency range.[55][56]

Layer II can also optionally use intensity stereo coding, a form of joint stereo. This means that the frequencies above 6 kHz of both channels are combined/down-mixed into one single (mono) channel, but the "side channel" information on the relative intensity (volume, amplitude) of each channel is preserved and encoded into the bitstream separately. On playback, the single channel is played through left and right speakers, with the intensity information applied to each channel to give the illusion of stereo sound.[41][53] This perceptual trick is known as stereo irrelevancy. This can allow further reduction of the audio bitrate without much perceivable loss of fidelity, but is generally not used with higher bitrates as it does not provide very high quality (transparent) audio.[41][54][57][58]

Quality

Subjective audio testing by experts, in the most critical conditions ever implemented, has shown MP2 to offer transparent audio compression at 256 kbit/s for 16-bit 44.1 kHz CD audio using the earliest reference implementation (more recent encoders should presumably perform even better).[1][53][54][59] That (approximately) 1:6 compression ratio for CD audio is particularly impressive because it is quite close to the estimated upper limit of perceptual entropy, at just over 1:8.[60][61] Achieving much higher compression is simply not possible without discarding some perceptible information.

MP2 remains a favoured lossy audio coding standard due to its particularly high audio coding performances on important audio material such as castanet, symphonic orchestra, male and female voices and particularly complex and high energy transients (impulses) like percussive sounds: triangle, glockenspiel and audience applause.[21] More recent testing has shown that MPEG Multichannel (based on MP2), despite being compromised by an inferior matrixed mode (for the sake of backwards compatibility)[1][54] rates just slightly lower than much more recent audio codecs, such as Dolby Digital (AC-3) and Advanced Audio Coding (AAC) (mostly within the margin of error—and substantially superior in some cases, such as audience applause).[62][63] This is one reason that MP2 audio continues to be used extensively. The MPEG-2 AAC Stereo verification tests reached a vastly different conclusion, however, showing AAC to provide superior performance to MP2 at half the bitrate.[64] The reason for this disparity with both earlier and later tests is not clear, but strangely, a sample of applause is notably absent from the latter test.

Layer II audio files typically use the extension .mp2 or sometimes .m2a

Layer III/MP3

MPEG-1 Layer III (MP3) is a lossy audio format designed to provide acceptable quality at about 64 kbit/s for monaural audio over single-channel (BRI) ISDN links, and 128 kbit/s for stereo sound.

History/ASPEC

2016-07 ASPEC 91 Bonn
ASPEC 91 in the Deutsches Museum Bonn, with encoder (below) and decoder

Layer III/MP3 was derived from the Adaptive Spectral Perceptual Entropy Coding (ASPEC) codec developed by Fraunhofer as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of digital audio broadcasting. ASPEC was adapted to fit in with the Layer II/MUSICAM model (frame size, filter bank, FFT, etc.), to become Layer III.[13]

ASPEC was itself based on Multiple adaptive Spectral audio Coding (MSC) by E. F. Schroeder, Optimum Coding in the Frequency domain (OCF) the doctoral thesis by Karlheinz Brandenburg at the University of Erlangen-Nuremberg, Perceptual Transform Coding (PXFM) by J. D. Johnston at AT&T Bell Labs, and Transform coding of audio signals by Y. Mahieux and J. Petit at Institut für Rundfunktechnik (IRT/CNET).[65]

Technical details

MP3 is a frequency-domain audio transform encoder. Even though it utilizes some of the lower layer functions, MP3 is quite different from Layer II/MP2.

MP3 works on 1152 samples like Layer II, but needs to take multiple frames for analysis before frequency-domain (MDCT) processing and quantization can be effective. It outputs a variable number of samples, using a bit buffer to enable this variable bitrate (VBR) encoding while maintaining 1152 sample size output frames. This causes a significantly longer delay before output, which has caused MP3 to be considered unsuitable for studio applications where editing or other processing needs to take place.[54]

MP3 does not benefit from the 32 sub-band polyphased filter bank, instead just using an 18-point MDCT transformation on each output to split the data into 576 frequency components, and processing it in the frequency domain.[53] This extra granularity allows MP3 to have a much finer psychoacoustic model, and more carefully apply appropriate quantization to each band, providing much better low-bitrate performance.

Frequency-domain processing imposes some limitations as well, causing a factor of 12 or 36 × worse temporal resolution than Layer II. This causes quantization artifacts, due to transient sounds like percussive events and other high-frequency events that spread over a larger window. This results in audible smearing and pre-echo.[54] MP3 uses pre-echo detection routines, and VBR encoding, which allows it to temporarily increase the bitrate during difficult passages, in an attempt to reduce this effect. It is also able to switch between the normal 36 sample quantization window, and instead using 3× short 12 sample windows instead, to reduce the temporal (time) length of quantization artifacts.[54] And yet in choosing a fairly small window size to make MP3's temporal response adequate enough to avoid the most serious artifacts, MP3 becomes much less efficient in frequency domain compression of stationary, tonal components.

Being forced to use a hybrid time domain (filter bank) /frequency domain (MDCT) model to fit in with Layer II simply wastes processing time and compromises quality by introducing aliasing artifacts. MP3 has an aliasing cancellation stage specifically to mask this problem, but which instead produces frequency domain energy which must be encoded in the audio. This is pushed to the top of the frequency range, where most people have limited hearing, in hopes the distortion it causes will be less audible.

Layer II's 1024 point FFT doesn't entirely cover all samples, and would omit several entire MP3 sub-bands, where quantization factors must be determined. MP3 instead uses two passes of FFT analysis for spectral estimation, to calculate the global and individual masking thresholds. This allows it to cover all 1152 samples. Of the two, it utilizes the global masking threshold level from the more critical pass, with the most difficult audio.

In addition to Layer II's intensity encoded joint stereo, MP3 can use middle/side (mid/side, m/s, MS, matrixed) joint stereo. With mid/side stereo, certain frequency ranges of both channels are merged into a single (middle, mid, L+R) mono channel, while the sound difference between the left and right channels is stored as a separate (side, L-R) channel. Unlike intensity stereo, this process does not discard any audio information. When combined with quantization, however, it can exaggerate artifacts.

If the difference between the left and right channels is small, the side channel will be small, which will offer as much as a 50% bitrate savings, and associated quality improvement. If the difference between left and right is large, standard (discrete, left/right) stereo encoding may be preferred, as mid/side joint stereo will not provide any benefits. An MP3 encoder can switch between m/s stereo and full stereo on a frame-by-frame basis.[53][58][66]

Unlike Layers I/II, MP3 uses variable-length Huffman coding (after perceptual) to further reduce the bitrate, without any further quality loss.[51][54]

Quality

These technical limitations inherently prevent MP3 from providing critically transparent quality at any bitrate. This makes Layer II sound quality actually superior to MP3 audio, when it is used at a high enough bitrate to avoid noticeable artifacts. The term "transparent" often gets misused, however. The quality of MP3 (and other codecs) is sometimes called "transparent," even at impossibly low bitrates, when what is really meant is "good quality on average/non-critical material," or perhaps "exhibiting only non-annoying artifacts."

MP3's more fine-grained and selective quantization does prove notably superior to Layer II/MP2 at lower-bitrates, however. It is able to provide nearly equivalent audio quality to Layer II, at a 15% lower bitrate (approximately).[63][64] 128 kbit/s is considered the "sweet spot" for MP3; meaning it provides generally acceptable quality stereo sound on most music, and there are diminishing quality improvements from increasing the bitrate further. MP3 is also regarded as exhibiting artifacts that are less annoying than Layer II, when both are used at bitrates that are too low to possibly provide faithful reproduction.

Layer III audio files use the extension .mp3.

MPEG-2 audio extensions

The MPEG-2 standard includes several extensions to MPEG-1 Audio.[54] These are known as MPEG-2 BC – backwards compatible with MPEG-1 Audio.[67][68][69][70] MPEG-2 Audio is defined in ISO/IEC 13818-3

These sampling rates are exactly half that of those originally defined for MPEG-1 Audio. They were introduced to maintain higher quality sound when encoding audio at lower-bitrates.[20] The even-lower bitrates were introduced because tests showed that MPEG-1 Audio could provide higher quality than any existing (circa 1994) very low bitrate (i.e. speech) audio codecs.[71]

Part 4: Conformance testing

Part 4 of the MPEG-1 standard covers conformance testing, and is defined in ISO/IEC-11172-4.

Conformance: Procedures for testing conformance.

Provides two sets of guidelines and reference bitstreams for testing the conformance of MPEG-1 audio and video decoders, as well as the bitstreams produced by an encoder.[10][18]

Part 5: Reference software

Part 5 of the MPEG-1 standard includes reference software, and is defined in ISO/IEC TR 11172-5.

Simulation: Reference software.

C reference code for encoding and decoding of audio and video, as well as multiplexing and demultiplexing.[10][18]

This includes the ISO Dist10 audio encoder code, which LAME and TooLAME were originally based upon.

File extension

.mpg is one of a number of file extensions for MPEG-1 or MPEG-2 audio and video compression. MPEG-1 Part 2 video is rare nowadays, and this extension typically refers to an MPEG program stream (defined in MPEG-1 and MPEG-2) or MPEG transport stream (defined in MPEG-2). Other suffixes such as .m2ts also exist specifying the precise container, in this case MPEG-2 TS, but this has little relevance to MPEG-1 media.

.mp3 is the most common extension for files containing MPEG-1 Layer 3 audio. An MP3 file is typically an uncontained stream of raw audio; the conventional way to tag MP3 files is by writing data to "garbage" segments of each frame, which preserve the media information but are discarded by the player. This is similar in many respects to how raw .AAC files are tagged (but this is less supported nowadays, e.g. iTunes).

Note that although it would apply, .mpg does not normally append raw AAC or AAC in MPEG-2 Part 7 Containers. The .aac extension normally denotes these audio files.

See also

Implementations
  • Libavcodec includes MPEG-1/2 video/audio encoders and decoders
  • Mjpegtools MPEG-1/2 video/audio encoders
  • TooLAME A high quality MPEG-1 Layer II audio encoder.
  • LAME A high quality MP3 (Layer III) audio encoder.
  • Musepack A format originally based on MPEG-1 Layer II audio, but now incompatible.

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External links

CDfs

CDfs is a virtual file system for Unix-like operating systems; it provides access to data and audio tracks on Compact Discs. When the CDfs driver mounts a Compact Disc, it represents each track as a file. This is consistent with the Unix convention "everything is a file".

CDfs supports the following track types:

Red Book Compact Disc Digital Audio (CD-DA): Appears as a WAV file; reading from it initiates DAE ripping.

White Book Video CD or Super Video CD video: Appears as a playable MPEG-1 file containing audio and video streams.

Yellow Book CD-ROM data:

Hierarchical File System: Appears as a mountable HFS file system disk image (sans partition table).

ISO 9660: Each session appears as a mountable ISO image file.

El Torito boot file: Appears as a single bootable disk image file.

Comparison of video container formats

This table compares features of container formats (video file formats). To see which multimedia players support which container format, look at comparison of media players.

Digital Satellite Service

Digital Satellite System is the initialism expansion of the DSS digital satellite television transmission system used by DirecTV. Only when digital transmission was introduced did direct broadcast satellite (DBS) television become popular in North America, which has led to both DBS and DSS being used interchangeably to refer to all three commonplace digital transmission formats - DSS, DVB-S and 4DTV. Analog DBS services, however, existed prior to DirecTV and were still operational in continental Europe until April 2012.At the time of DirecTV's launch in 1994, the DVB-S digital satellite system in use in the majority of the world had not yet been standardised, the Thomson developed DSS system was used instead.

While functionally similar in DVB-S - MPEG 2 video, MPEG-1 Layer II or AC3 audio, QPSK modulation, and identical error correction (Reed-Solomon coding and Viterbi forward error correction), the transport stream and information tables are entirely different from those of DVB. Also unlike DVB, all DSS receivers are proprietary DirecTV reception units.

DirecTV is now using a modified version of DVB-S2, the latest version of the DVB-S protocol, for HDTV services off the SPACEWAY-1, SPACEWAY-2, DirecTV-10 and DirecTV-11 satellites; however, huge numbers of DSS encoded channels still remain. The ACM modulation scheme used by DirecTV prevents regular DVB-S2 demodulators from receiving the signal although the data carried are regular MPEG-4 transport streams.

Elementary stream

An elementary stream (ES) as defined by the MPEG communication protocol is usually the output of an audio or video encoder. ES contains only one kind of data (e.g. audio, video, or closed caption). An elementary stream is often referred to as "elementary", "data", "audio", or "video" bitstreams or streams. The format of the elementary stream depends upon the codec or data carried in the stream, but will often carry a common header when packetized into a packetized elementary stream.

G.723

G.723 is an ITU-T standard speech codec using extensions of G.721 providing voice quality covering 300 Hz to 3400 Hz using Adaptive Differential Pulse Code Modulation (ADPCM) to 24 and 40 kbit/s for digital circuit multiplication equipment (DCME) applications. The standard G.723 is obsolete and has been superseded by G.726.

Note that this is a completely different codec from G.723.1.

H.262/MPEG-2 Part 2

H.262 or MPEG-2 Part 2 (formally known as ITU-T Recommendation H.262 and ISO/IEC 13818-2, also known as MPEG-2 Video) is a video coding format developed and maintained jointly by ITU-T Video Coding Experts Group (VCEG) and ISO/IEC Moving Picture Experts Group (MPEG). It is the second part of the ISO/IEC MPEG-2 standard. The ITU-T Recommendation H.262 and ISO/IEC 13818-2 documents are identical. The standard is available for a fee from the ITU-T and ISO.

MPEG-2 Video is similar to MPEG-1, but also provides support for interlaced video (an encoding technique used in analog NTSC, PAL and SECAM television systems). MPEG-2 video is not optimized for low bit-rates (less than 1 Mbit/s), but outperforms MPEG-1 at 3 Mbit/s and above. All standards-conforming MPEG-2 Video decoders are fully capable of playing back MPEG-1 Video streams.

Indeo

Indeo Video (commonly known now simply as "Indeo") is a family of audio and video formats and codecs designed for real-time video playback on desktop CPUs first released in 1992. While its original version was related to Intel's DVI video stream format, a hardware-only codec for the compression of television-quality video onto compact discs, Indeo was distinguished by being one of the first codecs allowing full-speed video playback without using hardware acceleration. Also unlike Cinepak and TrueMotion S, the compression used the same Y'CbCr 4:2:0 colorspace as the ITU's H.261 and ISO's MPEG-1.

Indeo use was free of charge to allow for broadest usage.

MP3

MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is a coding format for digital audio. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit-rates and support for more audio channels—as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG 2.5—extended to better support lower bit rates—is commonly implemented, but is not a recognized standard.

MP3 (or mp3) as a file format commonly designates files containing an elementary stream of MPEG-1 audio and video encoded data, without other complexities of the MP3 standard.

In the aspects of MP3 pertaining to audio compression—the aspect of the standard most apparent to end-users (and for which is it best known)—MP3 uses lossy data-compression to encode data using inexact approximations and the partial discarding of data. This allows a large reduction in file sizes when compared to uncompressed audio. The combination of small size and acceptable fidelity led to a boom in the distribution of music over the Internet in the mid- to late-1990s, with MP3 serving as an enabling technology at a time when bandwidth and storage were still at a premium. The MP3 format soon became associated with controversies surrounding copyright infringement, music piracy, and the file ripping/sharing services MP3.com and Napster, among others. With the advent of portable media players, a product category also including smartphones, MP3 support remains near-universal.

MP3 compression works by reducing (or approximating) the accuracy of certain components of sound that are considered (by psychoacoustic analysis) to be beyond the hearing capabilities of most humans. This method is commonly referred to as perceptual coding or as psychoacoustic modeling. The remaining audio information is then recorded in a space-efficient manner. Compared to CD-quality digital audio, MP3 compression can commonly achieve a 75 to 95% reduction in size. For example, an MP3 encoded at a constant bitrate of 128 kbit/s would result in a file approximately 9% of the size of the original CD audio.The Moving Picture Experts Group (MPEG) designed MP3 as part of its MPEG-1, and later MPEG-2, standards. The first subgroup for audio was formed by several teams of engineers at CCETT, Matsushita, Philips, Sony, AT&T-Bell Labs, Thomson-Brandt, and others. MPEG-1 Audio (MPEG-1 Part 3), which included MPEG-1 Audio Layer I, II and III, was approved as a committee draft for an ISO/IEC standard in 1991, finalised in 1992, and published in 1993 as ISO/IEC 11172-3:1993. A backwards-compatible MPEG-2 Audio (MPEG-2 Part 3) extension with lower sample- and bit-rates was published in 1995 as ISO/IEC 13818-3:1995.

MPEG-1 Audio Layer I

MPEG-1 Audio Layer I, commonly abbreviated to MP1, is one of three audio formats included in the MPEG-1 standard. It is a deliberately simplified version of MPEG-1 Audio Layer II, created for applications where lower compression efficiency could be tolerated in return for a less complex algorithm that could be executed with simpler hardware requirements. While supported by most media players, the codec is considered largely obsolete, and replaced by MP2 or MP3.

For files only containing MP1 audio, the file extension .mp1 is used.

MPEG-1 layer I was also used by the Digital Compact Cassette format, in the form of the PASC audio compression codec. Because of the need of a steady stream of frames per second on a tape-based medium, PASC uses the rarely used (and under-documented) padding bit in the MPEG header to indicate that a frame was padded with 32 extra 0-bits (four 0-bytes) to change a short 416-byte frame into 420 bytes. The varying frame size only occurs when a 44.1 kHz 16-bit stereo audio signal is encoded at 384 kilobits per second, because the bitrate of the uncompressed signal is not an exact multiple of the bitrate of the compressed bit stream.

MPEG-1 Audio Layer II

MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.

MPEG-2

MPEG-2 (a.k.a. H.222/H.262 as defined by the ITU) is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.265/HEVC, backwards compatibility with existing hardware and software means it is still widely used, for example in over-the-air digital television broadcasting and in the DVD-Video standard.

MPEG-2 Part 3

Part 3 of the MPEG-2 standard (formally known as ISO/IEC 13818-3, also known as MPEG-2 Audio or MPEG-2 BC) defines audio coding:

MPEG Multichannel - It enhances MPEG-1's audio by allowing the coding of audio programs with more than two channels, up to 5.1 multichannel. This method is backwards-compatible (also known as MPEG-2 BC), allowing MPEG-1 audio decoders to decode the two main stereo components of the presentation.

MPEG-2 Part 3 also defined additional bit rates and sample rates for MPEG-1 Audio Layer I, MPEG-1 Audio Layer II and MPEG-1 Audio Layer III (a.k.a. MP3).The MPEG-2 Part 3 should not be confused with MPEG-2 Part 7: AAC a.k.a. MPEG-2 NBC (Non-Backward Compatible) - the MPEG-2 Advanced Audio Coding with support for multichannel encoding (up to 48 channels).

MPEG Multichannel

MPEG Multichannel is an extension to the MPEG-1 Layer II audio compression specification, as defined in the MPEG-2 Audio standard (ISO/IEC 13818-3) which allows it provide up to 5.1-channels (surround sound) of audio. To maintain backwards compatibility with the older 2-channel (stereo) audio specification, it uses a channel matrixing scheme, where the additional channels are mixed into the two backwards compatible channels. Extra information in the data stream (ignored by older hardware) contains signals to process extra channels from the matrix.It was originally a mandatory part of the DVD specification for European DVDs, but was dropped in late 1997, and is rarely used as a result.

The Super Video CD (SVCD) standard supports MPEG Multichannel. Player support for this audio format is nearly non-existent however, and it is rarely used.

MPEG Multichannel audio was proposed for use in the ATSC digital TV broadcasting standard, but Dolby Digital (aka. AC-3, A/52) was chosen instead. This is a matter of significant controversy, as it has been revealed that the organizations (The Massachusetts Institute of Technology and Zenith Electronics) behind 2 of the 4 voting board members received tens of millions of dollars of compensation from secret deals with Dolby Laboratories in exchange for their votes.MPEG Multichannel–compatible equipment would bear either the MPEG Multichannel or MPEG Empowered logos.

MPEG program stream

Program stream (PS or MPEG-PS) is a container format for multiplexing digital audio, video and more. The PS format is specified in MPEG-1 Part 1 (ISO/IEC 11172-1) and MPEG-2 Part 1, Systems (ISO/IEC standard 13818-1/ITU-T H.222.0). The MPEG-2 Program Stream is analogous and similar to ISO/IEC 11172 Systems layer and it is forward compatible.Program streams are used on DVD-Video discs and HD DVD video discs, but with some restrictions and extensions. The filename extensions are VOB and EVO respectively.

Macroblock

Macroblock is a processing unit in image and video compression formats based on linear block transforms, such as the discrete cosine transform (DCT). A macroblock typically consists of 16×16 samples, and is further subdivided into transform blocks, and may be further subdivided into prediction blocks. Formats which are based on macroblocks include JPEG, where they are called MCU blocks, H.261, MPEG-1 Part 2, H.262/MPEG-2 Part 2, H.263, MPEG-4 Part 2, and H.264/MPEG-4 AVC. In H.265/HEVC, the macroblock as a basic processing unit has been replaced by the coding tree unit.

Moving Picture Experts Group

The Moving Picture Experts Group (MPEG) is a working group of authorities that was formed by ISO and IEC to set standards for audio and video compression and transmission. It was established in 1988 by the initiative of Hiroshi Yasuda (Nippon Telegraph and Telephone) and Leonardo Chiariglione, group Chair since its inception. The first MPEG meeting was in May 1988 in Ottawa, Canada. As of late 2005, MPEG has grown to include approximately 350 members per meeting from various industries, universities, and research institutions. MPEG's official designation is ISO/IEC JTC 1/SC 29/WG 11 – Coding of moving pictures and audio (ISO/IEC Joint Technical Committee 1, Subcommittee 29, Working Group 11).

TooLAME

TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork (the latest version, TwoLAME 0.3.13, was released January 21, 2011). The name TooLAME is a play on LAME and Layer II.

Video CD

Video CD (abbreviated as VCD, and also known as Compact Disc Digital Video) is a home video format and the first format for distributing films on standard 120 mm (4.7 in) optical discs. The format was widely adopted in Southeast Asia and superseded the VHS and Betamax systems in the region until DVD finally became affordable in the region in the late 2000s.

The format is a standard digital format for storing video on a compact disc. VCDs are playable in dedicated VCD players and widely playable in most DVD players, personal computers and some video game consoles. However, they are less widely playable in some Blu-ray Disc players and video game consoles such as the Sony PlayStation 3/4 due to lack of support for backward compatibility of the older MPEG-1 format.

The Video CD standard was created in 1993

by Sony, Philips, Matsushita, and JVC and is referred to as the White Book standard.

Although they have been superseded by other media, VCDs continue to be retailed as a low-cost video format.

Video file format

A video file format is a type of file format for storing digital video data on a computer system. Video is almost always stored using lossy compression to reduce the file size.

A video file normally consists of a container (e.g. in the Matroska format) containing video data in a video coding format (e.g. VP9) alongside audio data in an audio coding format (e.g. Opus). The container can also contain synchronization information, subtitles, and metadata such as title. A standardized (or in some cases de facto standard) video file type such as .webm is a profile specified by a restriction on which container format and which video and audio compression formats are allowed.

The coded video and audio inside a video file container (i.e. not headers, footers, and metadata) is called the essence. A program (or hardware) which can decode compressed video or audio is called a codec; playing or encoding a video file will sometimes require the user to install a codec library corresponding to the type of video and audio coding used in the file.

Good design normally dictates that a file extension enables the user to derive which program will open the file from the file extension. That is the case with some video file formats, such as WebM (.webm), Windows Media Video (.wmv), Flash Video (.flv), and Ogg Video (.ogv), each of which can only contain a few well-defined subtypes of video and audio coding formats, making it relatively easy to know which codec will play the file. In contrast to that, some very general-purpose container types like AVI (.avi) and QuickTime (.mov) can contain video and audio in almost any format, and have file extensions named after the container type, making it very hard for the end user to use the file extension to derive which codec or program to use to play the files.

The free software FFmpeg project's libraries have very wide support for encoding and decoding video file formats. For example, Google uses ffmpeg to support a wide range of upload video formats for YouTube. One widely used media player using the ffmpeg libraries is the free software VLC media player, which can play most video files that end users will encounter.

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