User:Ryan Cooley/MPEG1

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MPEG-1 articles (MPEG-1, MP1, MP2, MP3) on wikipedia are complete crap. Disorganized, slanted, incomplete, misconstrued, etc. It's far easier to start from scratch than try to fix all the individual existing ones, and will give far better end results; I will use some small bits of content from the existing articles.

Do not make any changes to this page for now. This is my mind-dump and accommodating others before I'm done will just make much, much more work for me. Put any suggestions on the Talk page, and I will eventually address them. -RC


MPEG-1 was an early standard for lossy compression of video and audio. It was designed to compress raw video and CD audio to 1.5Mb/s without discernible quality loss, making Video CDs and Digital Video Broadcasting possible.

Perhaps the most well-known part of the MPEG-1 standard today is the MP3 audio format it introduced.

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 Photographics Expert Group and CCITT's Experts Group on Telephony (creators of the JPEG image compression standard and the H.261 standard for video conferencing over ISDN lines respectively) the 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-realtime) encoding methods. [1]

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 (combined video+audio) data rates of 1.5Mbps. 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. [2]

After 20 meetings of the full group in various cities around the world, and 4 1/2 years of development and testing, (a draft standard was produced September 1990, and only minor changes were introduced) the final standard was approved in early November 1992. [3] Before the MPEG-1 standard had even been finalized, work began on a second standard, MPEG-2, intended to extend MPEG-1 technology to provide full broadcast-quality at high bitrates (3 - 15 Mbps), and support for interlaced video. [4] Due in part to the similarity between the two codecs, the MPEG-2 standard included full backwards compatibility with MPEG-1 video.

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 they did provide a reference implementation). 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

  • Today, MPEG-1 is by far the most widely compatible lossy audio/video format in the world.
  • Due to its age, most patents on MPEG-1 Video and Layer I/II audio technology have expired (MP3 remains patent encumbered), and can be implemented without payment of license fees in almost all countries.
  • Most computer software for video playback includes MPEG-1 decoding, in addition to any other supported formats.
  • The immense popularity of MP3 audio has established a massive installed base of hardware that can playback all 3 layers of MPEG-1 audio.
  • Millions of portable digital audio players (such as iPods) can playback MPEG-1 audio.
  • The widespread popularity of MPEG-2 (mostly with broadcasters) means MPEG-1 is playable by most digital cable/satellite set-top-boxes, and digital disc and tape players.
  • MPEG-1 video and audio is the exclusive format used on Video CD (VCD), the first consumer digital video format, and still very popular around the world.
  • The Super Video CD standard, based on VCD uses MPEG-1 audio exclusively, as well as MPEG-2 video.
  • DVD video use primarily MPEG-2 video, but MPEG-1 support is explicitly specified in the standard.
  • DVD video originally required MPEG-1 Layer II audio for PAL countries, but was changed to allow Dolby Digital AC-3. MPEG-1 Layer II audio is still allowed on DVDs, although the MPEG-2 additions, like MPEG multichannel and VBR, are rarely supported. Most DVD players also support Video CD and MP3 CD playback.
  • The international Digital Video Broadcasting (DVB) standard primarily uses MPEG-1 Layer II audio, as well as MPEG-2 video.
  • The international Digital Audio Broadcasting (DAB) standard uses MPEG-1 Layer II audio exclusively due to error resilience and low complexity of decoding.
  • MPEG-1 Layer II audio, with MPEG multichannel extensions, was proposed for use in the North American ATSC 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 at least 2 (The Massachusetts Institute of Technology and Zenith) of the 4 voting board members received millions of dollars of compensation from Dolby Laboratories in exchange for their votes. [5]


Video

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

Color Space

Before encoding 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) and even further separated into red and blue components. The chroma is also subsampled to 4:2:0, meaning it is decimated by half vertically and half horizontally, or 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/complex details this can manifest as chroma aliasing artifacts. Compared to other digital compression artifacts, this issue seems to be very minor/rare source of annoyance.

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

Resolution

MPEG-1 supports resolutions up to 4096 × 4096.

MPEG-1 videos are most commonly found using (SIF) resolutions: 352x240, 352x288, or 320x240. These low resolutions, combined with a bitrate less than 1.5Mb/s, makes up what is known as a constrained parameters bitstream. This is the commonly accepted minimum video specifications any decoder should be able to play, 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.

Bitrate ?


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 used/referenced by I, P or B frames. D-frames are only useful 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 processing I-frames, and discarding the AC coefficients before display.

Macroblocks

MPEG-1 operate on video in a series of 8x8 blocks for quantization, motion estimation, etc. Because chroma is subsampled by 4, however. you need 4 luma blocks to correspond to 1 chroma block. This gives us the 16x16 macroblock as the smallest independent unit in video.

It is very important to maintain video resolutions that are multiples of 16.

The same problem can be seen where black bars do not fall on a macroblock boundary.


DCT

Each 8x8 block is encoded using the Forward Discrete Cosign Transform (FDCT). This process by itself is lossless, and is reversed by the Inverse DCT (IDCT) upon playback to produce the original values.

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

The (large) DC coefficient remains mostly consistent from one block to the next, and can be compressed quite effectively with DPCM, so only the amount of difference between each DC value needs to be stored. A significant number of the AC coefficients will be near 0, which can then be more efficiently compressed in a later step. Additionally, the frequency conversion is necessary for quantization.

Quantization

Quantization (of digital data) is, essentially, the process of reducing the accuracy of a signal.

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

Each of the 64 frequency values of the DCT block are divided by their corresponding values in the quantization table. This reduces or completely eliminates the information in some frequency components of the video, deemed less visually important. This quantization process usually reduces a significant number of the AC coefficients to zero.

Quantization eliminates a large amount of data, and is the main lossy processing step in MPEG-1 video encoding. This is also the source of most MPEG-1 video compression artifacts, like blockiness, color banding, noise, ringing, discoloration, et al. when video is encoded with an insufficient bitrate.

Lossless Data Compression

Several steps in the encoding of MPEG-1 video are lossless, meaning they will be reversed on decoding to produce exactly the same values. Since these lossless data compression steps don't add noise into or otherwise change the video (unlike quantization), it is often called noiseless coding.

Huffman Coding

After perceptual coding, 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 to keep the data as small as possible.

Once the table is constructed, those strings are in the data are replaced with their (much smaller) codes, which references the appropriate entry in the table.

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.

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


 Datarate

 I-frames (Intraframe) 
   Seeking
 P-frames (Predicted)
 B-frames (Bidirectional)
   Complexity (memory)
   Delay
 GOP
   Keyframe placement

 Quantization*
   Ringing (large coefficients in high frequency sub-bands)
   zigzag
 Motion Vectors/Estimation
   Black borders/Noise
   pel precision (half pixel IIRC)
   Two MV per macroblock (forward/backward pred)
   Prediction error
   DPCM encoded, just like DC coeffs
   Blockiness
 CBR/VBR
 Spacial Complexity
 Temporal 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 (impulse encoded)
  • Stereo
  • Dual (two uncorrelated mono channels)
  • Sampling rates: 32, 44.1 and 48 kHz
  • Bitrates: 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 and 384 kbit/s
 mono, stereo, joint stereo (impulse, m/s), dual.*
 efficient time-domain concealment characteristics

Layer I

MPEG-1 Layer I is nothing more than a simplified version of Layer II, designed for low-delay and low complexity to facilitate real-time encoding on the hardware available in 1990 for applications like teleconferencing and studio editing. With the substantial performance improvements in digital processing since, it has now been long obsolete.

It saw limited adoption in it's time, and most notably was used on the defunct Digital Compact Cassette at 384 kbps. Layer I audio files use the extension .mp1

Layer II

MPEG-1 Layer I/II are time-domain encoders that utilize an empirically determined/defined psychoacoustic model based on the absolute threshold of hearing, as well as auditory masking. A low-delay 32 sub-band polyphased filter bank is used for time-frequency mapping, with overlapping ranges to prevent aliasing.


??( with the global masking threshold determined using a 1024 point FFT???) Or Just for MP3???

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 high quality percussive sounds (impulses) like triangle, glockenspiel and audience applause. Testing has shown MP2 to be equivalent or superior to than much more recent audio codecs, such as Dolby Digital AC-3. [6]

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. [7] That (approximately) 1:6 compression ratio for CD audio is particularly impressive since it's quite close to upper theoretical limit of perceptual entropy, at just over 1:8. [8] [9] Achieving much higher compression is simply not possible without discarding some perceptible information. 

 audio broadcasting
 error resilient
 Musicam


Layer III/MP3

The hybrid (filter bank + MDCT) design of MP3 imposes some limitations as well. It causes a factor of 12 - 36 times worse temporal resolution than MP2, which can cause artifacts due to transients sounds like percussive events. This results in audible smearing and pre-echo. [10]

The hybrid design also introduces aliasing artifacts, which are compensated for, but that produces artifacts (energy?) encoded in the high frequency range.

The combination of MP3's 1152-sample data frames. Model 2 uses a 1024-point window for spectral estimation introduces further limitations.

Because of these issues, MP2 sound quality is actually superior to MP3 above 112 kbps/channel 

 "The combination of the two filter banks creates aliasing issues that are only partially handled by the "aliasing compensation" stage, but that create excess energy to be coded in the frequency domain, thereby decreasing coding efficiency.  Frequency resolution is limited by the small long block window size, decreasing coding efficiency

No scale factor band for frequencies above 15.5/15.8 kHz" 


 9 months?
 ASPEC (Fraunhoffer) 
 freq transform encoder 
 entropy coding (Huffman)
 Hybrid filtering
   MDCT (overlapping DCT)
   pre-echo worse
   aliasing issues
   "aliasing compensation"?
 mid/side (or impulse) joint stereo 
 576 frequency components
 selectivity
 "If there is a transient, 192 samples are taken instead of 576 to limit the temporal spread of quantization noise"?
 psychoacoustic model and frame format from MP1/2
 ringing
 CBR/VBR
 Frames are not independent

Systems

Part 1 of the MPEG-1 standard covers systems which is the logical layout of the encoded audio, video, and other bitstream data.

"The MPEG-1 Systems design is essentially identical to the MPEG-2 Program Stream structure." [11] 

 Program Stream
 Interleaving
 PES
   Wrap-around
 DTS
 Timebase correction
 Pixel/Display Aspect Ratio


See Also

  • MPEG The Moving Picture Experts Group
  • MP3 The Cultural Phenomenon in Music

References

  1. ↑ http://www.cis.temple.edu/~vasilis/Courses/CIS750/Papers/mpeg_6.pdf pp.2
  2. ↑ http://www.chiariglione.org/mpeg/meetings/santa_clara90/santa_clara_press.htm
  3. ↑ http://www.chiariglione.org/mpeg/meetings.htm
  4. ↑ http://www.chiariglione.org/mpeg/meetings/london/london_press.htm
  5. ↑ http://www-tech.mit.edu/V122/N54/54hdtv.54n.html
  6. ↑ Wustenhagen et al, Subjective Listening Test of Multi-channel Audio Codecs, AES 105th Convention Paper 4813, San Francisco 1998
  7. ↑ http://www.faqs.org/faqs/mpeg-faq/part1/ "You can compress the same stereo program down to 256 Kbits/s with no loss in discernable quality." (the original papers would be much, much better refs, but I can't seem to find them! This just proves they exist!)
  8. ↑ J. Johnston, Estimation of Perceptual Entropy Using Noise Masking Criteria, in Proc. ICASSP-88, pp. 2524-2527, May 1988.
  9. ↑ 6. J. Johnston, Transform Coding of Audio Signals Using Perceptual Noise Criteria, IEEE J. Sel. Areas in Comm., pp. 314-323, Feb. 1988.
  10. ↑ http://www.cs.columbia.edu/~coms6181/slides/6R/mpegaud.pdf pp.8
  11. ↑ http://www.chiariglione.org/mpeg/faq/mp1-sys/mp1-sys.htm

External Links

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