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Moving Picture Experts Group Phase 1 (MPEG-1).mpg,.mpeg,.mp1,.mp2,.mp3,.m1v,.m1a,.m2a,.mpa,.mpv audio/mpeg, video/mpeg Developed by, Initial release created 1988–92 Type of format audio, video, container Extended from, Extended to / 11172 MPEG-1 is a for compression of and. It is designed to compress -quality raw digital video and CD audio down to 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making, digital / TV and (DAB) 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 audio format it introduced.
The MPEG-1 standard is published as / 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:. Systems (storage and synchronization of video, audio, and other data together). Video (compressed video content). Audio (compressed audio content).
Conformance testing (testing the correctness of implementations of the standard). Reference software (example software showing how to encode and decode according to the standard).
Contents. History Modeled on the successful collaborative approach and the compression technologies developed by the and 's (creators of the image compression standard and the standard for respectively), the (MPEG) working group was established in January 1988. MPEG was formed to address the need for video and audio formats, and to build on H.261 to get better quality through the use of more complex encoding methods. It was established in 1988 by the initiative of and. 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 and (human perceived) quality, at data rates of 1.5 Mbit/s. This specific bitrate was chosen for transmission over / lines and as the approximate data rate of. 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 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. 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. The draft standard was publicly available for purchase. The standard was finished with the 6 November 1992 meeting. The Berkeley Plateau Multimedia Research Group developed an MPEG-1 decoder in November 1992. In July 1990, before the first draft of the MPEG-1 standard had even been written, work began on a second standard, intended to extend MPEG-1 technology to provide full broadcast-quality video (as per ) at high bitrates (3–15 Mbit/s) and support for 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, 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 can drastically vary depending on the encoder used, and generally means that newer encoders perform significantly better than their predecessors. The first three parts (Systems, Video and Audio) of ISO/IEC 11172 were published in August 1993. MPEG-1 Parts Part Number First public release date (First edition) Latest correction Title Description Part 1 1993 1999 Systems Part 2 1993 2006 Video Part 3 1993 1996 Audio Part 4 1995 2007 Compliance testing Part 5 1998 2007 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.
The ISO patent database lists one patent for ISO 11172, US 4,472,747, which expired in 2003. The near-complete draft of the MPEG-1 standard was publicly available as ISO CD 11172 by December 6, 1991. Neither the July 2008 Kuro5hin article 'Patent Status of MPEG-1, H.261 and MPEG-2', nor an August 2008 thread on the gstreamer-devel 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. Filed in 1990 and published in 1993, this patent is now expired.
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 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.
Applications. Most popular for video playback includes MPEG-1 decoding, in addition to any other supported formats. The popularity of audio has established a massive of hardware that can play back MPEG-1 Audio (all three layers). 'Virtually all ' can play back MPEG-1 Audio.
Many millions have been sold to-date. Before became widespread, many digital satellite/cable TV services used MPEG-1 exclusively. The widespread popularity of MPEG-2 with broadcasters means MPEG-1 is playable by most digital cable and satellite, and digital disc and tape players, due to backwards compatibility. MPEG-1 was used for full-screen video on, and on (VCD). The standard, based on VCD, uses MPEG-1 audio exclusively, as well as MPEG-2 video. The 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/-only discs. MPEG-1 Layer II audio is still allowed on DVDs, although newer extensions to the format, like, are rarely supported. Most DVD players also support Video CD and playback, which use MPEG-1.
The international (DVB) standard primarily uses MPEG-1 Layer II audio, and MPEG-2 video. The international (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 is specifically designed for storage on media, and transmission over, 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: 'The MPEG-1 Systems design is essentially identical to the MPEG-2 Program Stream structure.' This terminology is more popular, precise (differentiates it from an ) 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 into packets of variable lengths, i.e., divided ES into independent chunks where (CRC) 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.
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. For more details on this topic, see. (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, or a. 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. 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. 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.
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 and are guaranteed to both arrive at the decoder at precisely the same time.
This is a case of. 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. The MPEG (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. 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. 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: ) Color space.
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'=Luma, Cb=Chroma Blue, Cr=Chroma Red). (brightness, resolution) is stored separately from (color, hue, phase) and even further separated into red and blue components. The chroma is also subsampled to, meaning it is reduced by one half vertically and one half horizontally, to just one quarter the resolution of the video. This software algorithm also has analogies in hardware, such as the output from a, 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), is a very effective way to reduce the amount of video data that needs to be compressed. On videos with fine detail (high ) this can manifest as chroma artifacts. Compared to other digital, this issue seems to be very rarely a source of annoyance. Because of subsampling, Y'CbCr video must always be stored using even dimensions ( 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 which is only used in the domain of video signals. Similarly, the terms and 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. MPEG-1 videos are most commonly seen using (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 (CPB), later renamed the 'Low Level' (LL) profile in MPEG-2. This is the minimum video specifications any should be able to handle, to be considered MPEG-1. 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, is I-frame. I-frames I-frame is an abbreviation for, 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 used in animation. I-frames can be considered effectively identical to baseline 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 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. The length between I-frames is known as the (GOP) size. MPEG-1 most commonly uses a GOP size of 15-18.
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 accumulation of IDCT errors in low-precision implementations most common in hardware decoders (See: -1180).
P-frames P-frame is an abbreviation for Predicted-frame. They may also be called forward-predicted frames, or (B-frames are also inter-frames). P-frames exist to improve compression by exploiting the (over time) 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 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 ), 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 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. 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 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. 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. Main article: 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 encoding on the hardware available 1990. Layer I saw limited adoption in its time, and most notably was used on ' at a bitrate of 384 kbit/s.
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. Main article: MPEG-1 Layer II ( MP2—often incorrectly called MUSICAM) is a audio format designed to provide high quality at about 192 kbit/s for stereo sound. Decoding MP2 audio is, relative to MP3, 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 (CCETT), and (IRT/CNET) as part of the 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.
Technical details Layer II/MP2 is a time-domain encoder. It uses a low-delay 32 sub-band for time-frequency mapping; having overlapping ranges (i.e. Polyphased) to prevent aliasing. The psychoacoustic model is based on the principles of, effects, and the (ATH). The size of a Layer II frame is fixed at 1152-samples (coefficients). 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 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 impulses (such as percussive instruments, and applause), offering avoidance of artifacts like pre-echo. The 32 sub-band filter bank returns 32, 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.
Example FFT analysis on an audio wave sample. The psychoacoustic model is applied using a 1024-point (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, and how much quantization noise each can contain without being perceived. Any sounds below the (ATH) are completely discarded.
The available bits are then assigned to each sub-band accordingly. 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. 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.
Layer II can also optionally use 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. 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. 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 using the earliest reference implementation (more recent encoders should presumably perform even better). That (approximately) 1:6 compression ratio for CD audio is particularly impressive because it is quite close to the estimated upper limit of perceptual, at just over 1:8.
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.
More recent testing has shown that (based on MP2), despite being compromised by an inferior matrixed mode (for the sake of backwards compatibility) rates just slightly lower than much more recent audio codecs, such as (AC-3) and (AAC) (mostly within the margin of error—and substantially superior in some cases, such as audience applause). 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. 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. ASPEC 91 in the, 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 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. ASPEC was itself based on Multiple adaptive Spectral audio Coding (MSC) by, Optimum Coding in the Frequency domain (OCF) the by at the, Perceptual Transform Coding (PXFM) by at, and Transform coding of audio signals by and at (IRT/CNET). Technical details MP3 is a frequency-domain audio.
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. 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. This extra 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. 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.
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. Unlike Layers I/II, MP3 uses variable-length (after perceptual) to further reduce the bitrate, without any further quality loss. 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). 128 kbit/s is considered the 'sweet spot' for MP3; meaning it provides generally acceptable quality stereo sound on most music, and there are 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 standard includes several extensions to MPEG-1 Audio.
These are known as MPEG-2 BC – backwards compatible with MPEG-1 Audio. MPEG-2 Audio is defined in ISO/IEC 13818-3. – Backward compatible 5.1-channel.: 16000, 22050, and 24000 Hz.: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144 and 160 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 even-lower bitrates were introduced because tests showed that MPEG-1 Audio could provide higher quality than any existing ( 1994) very low bitrate (i.e. ) audio codecs. 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. 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. Reference code for encoding and decoding of audio and video, as well as multiplexing and demultiplexing. This includes the ISO Dist10 audio encoder code, which and were originally based upon.
File extension.mpg is one of a number of file extensions for MPEG-1 or audio and video compression. MPEG-1 Part 2 video is rare nowadays, and this extension typically refers to an (defined in MPEG-1 and MPEG-2) or (defined in MPEG-2). Other suffixes such as.m2ts also exists 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 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. Note that although it would apply,.mpg does not normally append raw or AAC in.
The.aac extension normally denotes these audio files. See also. The Moving Picture Experts Group, developers of the MPEG-1 standard.
More (less technical) detail about MPEG-1 Layer III audio. Backwards compatible 5.1 channel extension to Layer II audio. The direct successor to the MPEG-1 standard. Implementations.
includes MPEG-1/2 video/audio encoders and decoders. MPEG-1/2 video/audio encoders.
A high quality MPEG-1 Layer II audio encoder. A high quality MP3 (Layer III) audio encoder. A format originally based on MPEG-1 Layer II audio, but now incompatible. References. ^ Adler, Mark; Popp, Harald; Hjerde, Morten (November 9, 1996), retrieved 2016-11-11. ^ Le Gall, Didier (April 1991), (PDF), retrieved 2016-11-11. Chiariglione, Leonardo (October 21, 1989), /, archived from on August 5, 2010, retrieved 2008-04-09.
ISO/IEC JTC 1/SC 29 (2009-10-30). Archived from on 2013-12-31. Retrieved 2009-11-10. Retrieved 2016-11-11. Archived from on 2008-07-08.
Retrieved 2009-10-31. Archived from on 2010-02-21. Retrieved 2009-10-31. Archived from on 2010-04-20.
Retrieved 2009-10-31. Fogg, Chad (April 2, 1996), archived from on August 29, 2000, retrieved 2008-04-09. ^ Fogg, Chad (April 2, 1996), archived from on 2008-06-16, retrieved 2016-11-11. Hans Geog Musmann, (PDF), archived from (PDF) on 2012-01-17, retrieved 2011-07-26. Chiariglione, Leonardo (March 2001), archived from on 2011-07-25, retrieved 2008-04-09. ^ Chiariglione, Leonardo; Le Gall, Didier; Musmann, Hans-Georg; Simon, Allen (September 1990), /, archived from on 2010-02-14, retrieved 2008-04-09., /, archived from on 2010-02-10, retrieved 2008-04-09. ^.
Archived from on 2009-07-23. Retrieved 2008-10-12. Well, then how do I get the documents, like the MPEG I draft? MPEG is a draft ISO standard. It's exact name is ISO CD 11172.
You may order it from your national standards body (e.g. ANSI in the USA) or buy it from companies like OMNICOM. Archived from on 2008-10-06. Retrieved 2008-07-13. Archived from on 2008-06-12.
Retrieved 2008-07-13. A Continuous Media Player, Lawrence A. Rowe and Brian C. Workshop on Network and OS Support for Digital Audio and Video, San Diego CA (November 1992). ^, /, archived from on 2008-07-08, retrieved 2008-04-03.
Chiariglione, Leonardo (November 6, 1992), /, archived from on 12 August 2010, retrieved 2008-04-09. ^ Wallace, Greg (April 2, 1993), /, archived from on August 6, 2010, retrieved 2008-04-09.
^ Popp, Harald; Hjerde, Morten (November 9, 1996), retrieved 2016-11-11. ISO/IEC JTC 1/SC 29 (2010-07-17). Archived from on 2013-12-31. Retrieved 2010-07-18. Retrieved 2016-11-11. Retrieved 2016-11-11.
Retrieved 2016-11-11. Retrieved 2016-11-11. Retrieved 2016-11-11. Ozer, Jan (October 12, 2001), retrieved 2016-11-11., retrieved 2016-11-11., 2003, archived from on 2005-12-14, retrieved 2008-04-09.
Dave Singer (2007-11-09). Retrieved November 11, 2016., Digital Preservation. October 21, 2014. Retrieved 2016-11-11.
Retrieved 2016-11-11. Search for 11172.
Reference 3 in the paper is to Committee Draft of Standard ISO/IEC 11172, December 6, 1991. Retrieved 2016-11-11. 'Digital transmission system using subband coding of a digital signal' Filed: May 31, 1990, Granted May 25, 1993, Expires May 31, 2010?. ^ Grill, B.; Quackenbush, S.
(October 2005), /, archived from on 2010-04-30. Chiariglione, Leonardo, /, retrieved 2016-11-11. ^, retrieved 2016-11-11. Fimoff, Mark; Bretl, Wayne E. (December 1, 1999), retrieved 2016-11-11. Fimoff, Mark; Bretl, Wayne E. (December 1, 1999), retrieved 2016-11-11.
Fimoff, Mark; Bretl, Wayne E. (December 1, 1999), retrieved 2016-11-11. Fimoff, Mark; Bretl, Wayne E.
(December 1, 1999), retrieved 2016-11-11. Acharya, Soam; Smith, Brian (1998), p. 3, retrieved 2016-11-11 - (Requires clever reading: says quantization matrices differ, but those are just defaults, and selectable). ^ Wee, Susie J.; Vasudev, Bhaskaran; Liu, Sam (March 13, 1997), archived from on 2007-08-17, retrieved 2016-11-11.
Archived from on 2009-05-03. Retrieved 2009-05-03. ^ Thom, D.; Purnhagen, H. (October 1998), /, retrieved 2016-11-11., archived from on 2015-02-08, retrieved 2016-11-11.
^ Church, Steve, archived from on 2001-05-08, retrieved 2008-04-09. ^ Pan, Davis (Summer 1995), (PDF), p. 8, archived from (PDF) on 2004-09-19, retrieved 2008-04-09. Smith, Brian (1996), p. 7, retrieved 2008-04-09. Cheng, Mike, retrieved 2016-11-11. Grill, B.; Quackenbush, S. (October 2005), archive.org, archived from on 2008-04-27, retrieved 2016-11-11.
^ Herre, Jurgen (October 5, 2004), (PDF), p. 2, archived from (PDF) on April 5, 2006, retrieved 2008-04-17. C.Grewin, and T.Ryden, Subjective Assessments on Low Bit-rate Audio Codecs, Proceedings of the 10th International AES Conference, pp 91 - 102, London 1991. J. Johnston, Estimation of Perceptual Entropy Using Noise Masking Criteria, in Proc. ICASSP-88, pp. 2524-2527, May 1988.
Johnston, Transform Coding of Audio Signals Using Perceptual Noise Criteria, IEEE Journal Select Areas in Communications, vol. 314-323, Feb. Wustenhagen et al., Subjective Listening Test of Multi-channel Audio Codecs, AES 105th Convention Paper 4813, San Francisco 1998. ^ B/MAE Project Group (September 2007), (PDF), archived from (PDF) on 2008-10-30, retrieved 2008-04-09.
^ Meares, David; Watanabe, Kaoru; Scheirer, Eric (February 1998), (PDF), /, p. 18, archived from (PDF) on April 14, 2008, retrieved 2016-11-11. Painter, Ted; Spanias, Andreas (April 2000), (PDF), archived from (PDF) on September 16, 2006, retrieved 2016-11-11. Amorim, Roberto (September 19, 2006), retrieved 2016-11-11. ISO (October 1998). Retrieved 2016-11-11. Purnhagen, and the MPEG Audio Subgroup (October 1998). Retrieved 2016-11-11.
CS1 maint: Multiple names: authors list. MPEG.ORG. Archived from on 2007-08-31. Retrieved 2009-10-28. ISO (2006-01-15), (PDF), retrieved 2016-11-11. Chiariglione, Leonardo (November 11, 1994), /, archived from on August 8, 2010, retrieved 2008-04-09 External links.
Various MPEG-Animations Various MPEG-Animations NOTE: unless noted otherwise none of the following MPEG-clips include any audio. Clips from various movies. Wallace & Gromit.
A Grand Day Out. (587091 Bytes). (1762084 Bytes). The Wrong Trousers. (1631586 Bytes). (1076091 Bytes). (495147 Bytes).
A Close Shave. (1784395 Bytes). (1340935 Bytes). (627173 Bytes). (394541 Bytes). (2744861 Bytes).
(665381 Bytes). (573440 Bytes). (916615 Bytes).
(390409 Bytes). (847502 Bytes).
Creature Comfort ( Not Wallace & Gromit but also by Nick Park ). Interview with a lion in the zoo.
in format. (10193068 Bytes). in format.
(31168039 Bytes) NOTE: The above MPEG-File (lion.mpg) is a MPEG-System-file (Video and Audio) it was created on a SGI and plays fine with the latest version of SGI's ' movieplayer '.' Mpegplay ' wasn't able to play this file when I tried it. So please make sure that the tool you use to view MPEG-files is able to handle MPEG-system-streams before you download 10MB.
Pink Floyd: The Wall. (426767 Bytes). (694121 Bytes).
(3585752 Bytes). Blade Runner. (1087519 Bytes). (1248464 Bytes). Highlander. (3216275 Bytes). (3231625 Bytes).
Blood Test Mpg
Psycho. (9362694 Bytes). Poltergeist. (1825154 Bytes). In case you're ever in need of a english/german translation, check out 's. In case you don't have a program to view MPEG-movies: For DOS/Windows PC you may try.
Mpg File Player
For Unix Workstations is usually a good choice. NOTE (a message to all companys that sell MPEG-players): These are the only links to MPEG-players I will put on this page. So please spare me the e-mail asking to add a link to your MPEG-player. (I had one too many of those requests recently.) 11.4.1995 13.7.1999.