User:Ryan Cooley/MPEG1

MPEG-1 was an early standard for lossy compression of video and audio. It was designed to compress VHS-quality raw digital video and CD audio from about 43 Mbit/s down to 1.5 Mbit/s (29:1 compression) without obvious quality loss, making Video CDs, digital cable TV/satellite TV and Digital Audio Broadcasting possible.

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.

Despite it's age, MPEG-1 is not necessarily obsolete or substantially inferior to newer technologies. According to Leonardo Chiariglione (co-founder of MPEG): "the idea that compression technology keeps on improving is a myth."

The MPEG-1 standard is published as ISO/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 encoding formats, and build on H.261 to get better quality through the use of more complex (non-real time) encoding methods.

Development of the MPEG-1 standard began in May 1988. 14 video and 14 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. 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.

After 20 meetings of the full group in various cities around the world, and 4 ¹/₂ years of development and testing, the final standard was approved in early November 1992 and published a few months later. 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. In July 1990, before the first draft of the MPEG-1 standard had even been written, work began on a second standard, MPEG-2, 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. 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.

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). 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.

Applications

• Most popular computer software for video playback includes MPEG-1 decoding, in addition to any other supported formats.
• MPEG-1 video and Layer I/II audio can be implemented without payment of license fees. Due to its age, many of the patents on the technology have expired.
• The popularity of MP3 audio has established a massive installed base of hardware that can playback MPEG-1 audio (all 3 layers).
• Before MPEG-2 became widespread, many digital satellite/cable TV services used MPEG-1 exclusively.
• Millions of portable digital audio players can playback MPEG-1 audio.
• 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 is the exclusive video and audio format used on Video CD (VCD), the first consumer digital video format, and still a very popular format around the world.
• The Super Video CD standard, based on VCD, uses MPEG-1 audio exclusively, as well as MPEG-2 video.
• DVD-Video 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 error resilience and low computational complexity for decoding.

Systems

Part 1 of the MPEG-1 standard covers systems, which is the logical layout of the encoded audio, video, and other bitstream data, and is defined in ISO/IEC-11172-3.

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

Elementary Streams

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

Additionally, elementary streams can be made more robust by packetizing them, ie. dividing them into independent chunks, and adding a cyclic redundancy check (CRC) checksum to each segment for error protection. This is the Packetized Elementary Stream (PES) structure.

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.

System Clock Reference (SCR) is a timing value stored in a 33-bit header of each ES, at a frequency/precision of 90kHz, with an extra 9-bit extension that stores addition timing data with a precision of 27MHz. 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, multiplexing, jitter, and other delay.

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

PTS handling can be problematic. It is required that decoders be able to handle multiple program streams that have been joined sequentially (concatenated/cascaded). 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.

Display 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, ahead of its anchor (P- or I-) frame. Without B-frames in the video, PTS and DTS values are identical.

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 (eg. audio), before it gets enough data to decode the other simultaneous stream (eg. video). The MPEG Video Buffer Verifier (VBV) assists in determining if a multiplexed PS can be decoded by a device with a specified data throughput rate and buffer size. This offers feedback to the muxer and the encoder, so that they can change the mux size or adjust bitrates as needed for compliance.

The PS, additionally, stores aspect ratio information which tells the decoder how much to stretch the height or width of a video when displaying it. This is important because different display devices (such as computer monitors and televisions) have different pixel height/width, which will result in video encoded for one appearing "squished" when played on the other, unless the aspect ratio information in the PS is used to compensate.

Video

Part 2 of the MPEG-1 standard covers video and is defined in ISO/IEC-11172-2.

Color Space

Example of 4:2:0 subsampling. The 2 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 decimated by one half vertically and one half horizontally, to just one quarter the resolution of the video.

Because the human eye is much less sensitive to small changes in color than in brightness, 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/sec.

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). 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.

Frame/Picture/Block Types

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

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.

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. 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 restrictions (color space and quantizer matrix) are followed in the creation of the bitstream.

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. ie. 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.

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 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 intra-frames (B-frames are also intra-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.

If a video drastically changes from one frame to the next (such as a scene change), it can be 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 (ie. 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 makes B-frames very computationally complex, requires larger data buffers, and causes an increased delay on both decoding and during encoding. This also necessitates the display 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 (a second time) in addition to much of the changes being coded in the B-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.

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

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 DC-only (AC coefficients are removed—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.

Given moderately higher-performance decoding equipment, this feature can be approximated by decoding I-frames instead. This provides higher quality previews, and without the need for D-frames taking up space in the stream, yet not improving video quality.

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 and/or width of the video are not exact multiples of 16, a full row of macroblocks must still be encoded (though not displayed) to store the remainder of the picture (macroblock padding). This wastes a significant amount of data in the bitstream, and is to be strictly avoided.

Some decoders will also improperly handle videos with partial macroblocks, resulting in visible artifacts.

Motion Vectors

To decrease the amount of spatial 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.

The encoder compares the current frame with adjacent parts of the video in the anchor frame (previous I- or P- frame) around the area of the current macroblock. It searches in a diamond pattern, up to a (encoder-specific) predefined radius limit.

This greatly helps find redundant information from the previous (anchor) frame, avoiding duplicate encoding of the same information. If a match is found, only the direction and distance (ie. the vector of the motion) from the previous video area to the current macroblock need to be encoded into the intra-frame (P- or B- frame).

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. This error must be seperately encoded in the frame.

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

The reverse of this process, performed by the decoder to reconstruct the picture, is called motion compensation.

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 precision of one half of one pixel, or half-pel. The finer the precision of the motion vectors, the more accurate the match is likely to be, and the more efficient the compression. There are trade-offs and diminishing returns with ever higher precision as well, however, and half-pel was chosen as the best trade-off. The more recent MPEG-4 standard optionally allows quarter-pel precision, but it's use is quite limited due to the drawbacks.

P-frames have 1 motion vector per macroblock, based on the previous anchor frame. B-frames, however, can use 2 motion vectors; one from the previous anchor frame, and one from the future anchor frame.

Partial macroblocks will cause havoc with motion prediction/vectors. The block padding 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 numerous partial macroblocks along the screen border.

This will significantly lower the quality (or require significantly increasing the bitrate) of the video stream. This is also true of black borders/bars that have been encoded into the video, and do not fall exactly on a macroblock boundary. A similar, but even more serious problem exists when macroblocks (typically around the edges of a video) contain significant, random, edge noise as the picture transitions to (typically) black.

Motion vectors are the main cause of compression artifacts that "move" around the screen.

DCT

Each 8x8 block is encoded using the Forward Discrete Cosine Transform (FDCT). This process by itself is theoretically lossless, and is reversed by the Inverse DCT (IDCT) upon playback to produce the original values. In reality, there are some (sometimes large) rounding errors. The minimum allowed accuracy of a DCT implementation is defined by IEEE-1180.

The FDCT process converts the 64 uncompressed pixel values (brightness) into 64 different frequency values. One (large) DC coefficient, which is the average of the entire 8x8 block, and 63 smaller AC coefficients, which are positive or negative values, each relative to the value of the DC coefficient.

An example FDCT encoded 8x8 block:

${\begin{bmatrix}-415&-30&-61&27&56&-20&-2&0\\4&-22&-61&10&13&-7&-9&5\\-47&7&77&-25&-29&10&5&-6\\-49&12&34&-15&-10&6&2&2\\12&-7&-13&-4&-2&2&-3&3\\-8&3&2&-6&-2&1&4&2\\-1&0&0&-2&-1&-3&4&-1\\0&0&-1&-4&-1&0&1&2\end{bmatrix}}$

Since the DC coefficient remains mostly consistent from one block to the next, it can be compressed quite effectively with DPCM-encoding. Only the (smaller) amount of difference between each DC value needs to be stored in the final bitstream. Additionally, this DCT frequency conversion is necessary for 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 (eg. finding the nearest multiple, and discarding the remainder/modulus).

The frame-level quantizer is a number from 1 to 31 (although encoders will often 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:

${\begin{bmatrix}16&11&10&16&24&40&51&61\\12&12&14&19&26&58&60&55\\14&13&16&24&40&57&69&56\\14&17&22&29&51&87&80&62\\18&22&37&56&68&109&103&77\\24&35&55&64&81&104&113&92\\49&64&78&87&103&121&120&101\\72&92&95&98&112&100&103&99\end{bmatrix}}$

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.

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:

${\begin{bmatrix}-26&-3&-6&2&2&-1&0&0\\0&-2&-4&1&1&0&0&0\\-3&1&5&-1&-1&0&0&0\\-4&1&2&-1&0&0&0&0\\1&0&0&0&0&0&0&0\\0&0&0&0&0&0&0&0\\0&0&0&0&0&0&0&0\\0&0&0&0&0&0&0&0\end{bmatrix}}$

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. Since lossless compression aims to remove as much redundancy as possible, it is known as entropy coding in the field of information theory.

RLE

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 bytes. This is stored in a special 2-dimensional Huffman table that codes the run-length and the run-ending character.

Huffman Coding

The data is then analyzed to find strings that repeat often. Those strings are then put into a special Huffman table, with the most frequently repeating data assigned the shortest code. This keeps the data as small as possible with this form of compression.

Once the table is constructed, those strings in the data are replaced with their (much smaller) codes, which references the appropriate entry in the table. This is the final step in the video encoding process, so the result of Huffman coding is known as the MPEG-1 video "bitstream."

 zigzag
 Spacial Complexity*

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 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.

Channel Encoding:

• Mono
• Joint Stereo (intensity encoded)
• Stereo
• Dual (two uncorrelated mono channels)

• Sampling rates: 32000, 44100, and 48000 Hz
• Bitrates: 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 and 384 kbit/s

MPEG-1 Audio is divide into 3 layers. Each higher layer is more computationally complex, and generally more efficient than the previous. The layers are also backwards compatible, so a Layer II decoder can also play Layer I audio, but NOT Layer III audio.

Layer I

MPEG-1 Layer I is nothing more than a simplified version of Layer II. Layer I uses a smaller 384-sample frame size for very low-delay, and finer resolution. 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. With the substantial performance improvements in digital processing since, Layer I quickly became unnecessary and obsolete.

Layer I saw limited adoption in it's time, and most notably was used on the defunct Philips Digital Compact Cassette at 384 kbit/s.

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

Layer II

MPEG-1 Layer II (aka. MP2, and often incorrectly called MUSICAM) is a time-domain encoder. It uses a low-delay 32 sub-band polyphased filter bank for time-frequency mapping; having overlapping ranges (ie. polyphased) to prevent aliasing. The psychoacoustic model is based on auditory masking / simultaneous masking effects and the absolute threshold of hearing (ATH) / masking threshold. The size of a Layer II frame is fixed at 1152-samples (coefficients).

Technical Details

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), allowing avoidance of artifacts like pre-echo.

File:Bande critiche - mpeg.gif

Visualization of the 32 sub-band filter bank used by MPEG-1 Audio. Even I am unsure what it is attempting to show.

The 32 sub-band filter bank returns 32 amplitude coefficients, one for each equal-sized frequency band/segment of the audio, which is about 700Hz wide. The encoder then utilizes the psychoacoustic model to determine which sub-band contains audible information that is less important, and so, where quantization will be in-audible, or at least much less noticeable (higher masking threshold).

Example FFT analysis on an audio wave sample.

The psychoacoustic model is applied using a 1024-point Fast Fourier Transform (FFT). 64 samples at each end of the frequency range, out of the 1152 samples total, are ignored for this analysis. They are presumably not significant enough to change the result. The psychoacoustic model determines which sub-bands contribute more to the masking threshold, and the available bits are assigned to each sub-band accordingly.

Typically, sub-bands are less important if they contain quieter sounds (smaller coefficient) than a neighboring (ie. similar frequency) sub-band with louder sounds (larger coefficient). Also, "noise" components typically have a more significant masking effect than "tonal" components. The less significant sub-bands are then reduced in accuracy by, basically, compressing the frequency range/amplitude, (raising the noise floor) and computing an amplification factor, for the decoder to use to re-expand it to the proper frequency range.

Additionally, Layer II can use intensity stereo coding. This means that both channels (above 6 kHz) are down-mixed into one single (mono) channel, but the ("side channel") information on the relative intensity (volume, amplitude) of each channel is preserved (up to 7 fixed positions) and encoded into the bitstream separately. On playback, the single channel is played through left and right speakers, with the intensity information applied to (pan) each channel to give the illusion of stereo sound. 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 high quality (transparent) audio.

History/MUSICAM

MPEG-1 Layer II was derived from the MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) audio codec, developed by CCETT (Centre commun d'études de télévision et télécommunications), Philips, and IRT (Institut für Rundfunktechnik) 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.

Quality

Subjective audio testing by experts, in the most critical conditions ever implemented, has shown MP2 to offer transparent audio compression at 256kbps for 16-bit 44.1khz CD audio. That (approximately) 1:6 compression ratio for CD audio is particularly impressive since it is quite close to the estimated upper limit of perceptual entropy, at just over 1:8. Achieving much higher compression is simply not possible without discarding some perceptible information.

Despite some 20 years of progress in the field of digital audio coding, MP2 remains the preeminent lossy audio coding standard due to its especially high audio coding performances on highly critical 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, quite the opposite of MP3. More recent testing has shown that MPEG Multichannel (based on MP2), despite being compromised by a significantly inferior matrixed mode, 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 still superior in some cases, namely audience applause).

This is one reason that MP2 audio continues to be used extensively. MP2's especially high quality, low decoder performance requirements, and tolerance of errors also helps makes it a popular choice for applications like digital audio broadcasting (DAB).

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

Layer III/MP3

MP3 is a frequency domain transform audio encoder.

History/ASPEC

The Adaptive Spectral Perceptual Entropy Coding (ASPEC) codec was 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 model, to become Layer III/MP3.

ASPEC itself was based on Multiple adaptive Spectral audio Coding (MSC) by E. F. Schroeder, Optimum Coding in the Frequency domain (OCF) the Ph.D 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 IRT ("CNET").

Even though it utilizes some of the lower layer functions, MP3 is quite different from Layer II/MP2.

In addition to Layer II intensity encoded joint stereo, MP3 can alternatively use mid/side joint stereo.

 With mid/side stereo, a small range of certain key frequencies is stored separately for each channel in addition to intensity information.  This provides higher fidelity joint stereo encoding.

The encoder can chose to switch between a matrixed L/R mode, or full stereo mode, on each frame.

The Layer II 1024 point (FFT) analysis window for spectral estimation (calculating the global and individual masking thresholds) is too small to cover all 1152 samples, so MP3 utilize two sequential passes.

MP3 does not benefit from the 32 sub-band filter bank, instead just using an 18-point MDCT transformation to split the data into 576 frequency components, and processing it in the frequency domain. This extra granularity allows MP3 to much more accurately apply its psychoacoustic model and applying an appropriate masking threshold to each band, than Layer II can, providing much better low-bitrate performance.

MP3 works on 1152 samples like Layer II, but needs to take multiple sample frames before MDCT processing can be effective. It also outputs in larger chunks and spreads the output over a varying number of several Layer I/II-sized output frames. This has caused MP3 to be considered unsuitable for studio applications where editing or other processing needs to take place, and in broadcasting, as small bit errors will spread though the audio over a much longer time period.

Unlike Layers I/II, MP3 uses Huffman coding (after perceptual) to further reduce the bitrate, without any further quality loss. This, however, also makes MP3 even more significantly affected by small errors.

The frequency domain (MDCT) design of MP3 imposes some limitations as well. It causes a factor of 12 - 36 times worse temporal resolution than Layer II, which causes artifacts due to transient sounds like percussive events, with artifacts spread over a larger window. This results in audible smearing and pre-echo. And yet in choosing a fairly small window size, trying to make MP3's temporal response adequate to avoid serious artifacts, MP3 becomes much less efficient in (regular) frequency domain compression.

Being forced to use this type of hybrid time domain (filter bank) and frequency domain (MDCT) model wastes processing time and compromises MP3 quality by introducing additional aliasing artifacts. MP3 has an aliasing compensation stage specifically to mask this problem, instead producing frequency domain energy which is pushed to the top of the frequency range, and causes high frequency distortion.

These issues prevent MP3 from providing transparent quality at any bitrate, and thereby making Layer II sound quality superior to MP3 audio at higher bitrates.

Layer III audio files use the extension .mp3

 No scale factor band for frequencies above 15.5/15.8 kHz"?
 "aliasing compensation"* need more details!
 "If there is a transient, 192 samples are taken instead of 576 to limit the temporal spread of quantization noise"
 ringing
 CBR/VBR

MPEG-2 Audio Extensions

The MPEG-2 standard includes several extensions to MPEG-1 Audio. MPEG-2 Audio is defined in ISO/IEC-13818-3

VBR audio encoding?

MPEG Multichannel ISO/IEC 14496-3. Backward compatible 5.1-channel surround sound.

Sampling rates: 16000, 22050, and 24000 Hz Bitrates: 8, 16, 24, 40, 48, and 144 kbit/s

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. The lower bitrates were introduced because tests showed that MPEG-1 Audio could provide higher quality than any existing (circa 1994) very low bitrate (ie. speech) audio codecs.

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.

Reference Software

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

Simulation: Reference software.

Includes ISO Dist10 audio encoder code which LAME and TooLAME were based upon.

See Also

• MPEG The Moving Picture Experts Group, developers of the MPEG-1 standard
• MP3 More (less technical) detail about MPEG-1 Layer III audio
• MPEG Multichannel Backwards compatible 5.1 channel surround sound extension to Layer II audio
• MPEG-2 The direct successor to the MPEG-1 standard.

Implementations

• Libavcodec includes MPEG-1 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.

External Links

• Official Web Page of the Moving Picture Experts Group (MPEG) a working group of ISO/IEC
• MPEG Industry Forum Organization