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Page 1: Multimedia Object - Audio

Multimedia SystemMultimedia SystemAudioAudio

Nyoman Bogi Aditya Karna, ST, MSEESisfo IM Telkom

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Image• Introduction• Compression• Types: GIF/JPEG

Sound• Introduction• Compression• Types: WAV/MPEG

Video• Introduction

• Compression• Types: MPEG

Multimedia Object

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MPEG (Moving Pictures Experts Group) was established 1988 by ISO as a research group to create standard for the coded representation of moving pictures and associated audio to be stored on digital storage media.

Mainly based in German at Fraunhofer Institute (IIS), MPEG submitted its research to ISO :

1993 : MPEG phase 1 (IS 11172-3)

1994 : MPEG phase 2 (IS 13818-3)

1997 : MPEG phase 2.5 (IS 13818-7)

1998 : MPEG phase 4 (IS 14496-3)

2001 : MPEG phase 7

MPEG1 is used in VCD (Video Compact Disc) technology, while Super VCD and DVD (Digital Versatile Disc) are using MPEG2. MPEG4 emphasis on functionality rather than new compression technology, while MPEG7 is a content representation standard.

MPEG - History

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1993 : MPEG1 (IS 11172)• Coding of moving pictures and associated audio for digital storage

media at up to about 1.5 Mbit/s (VCD is using 1.15 Mbit/s)• IS 11172-1 : System, describe synchronization and multiplexing of

audio and video signals• IS 11172-2 Video, describe compression of non-interlaced video signals• IS 11172-3 Sound, describe compression of audio signals• IS 11172-4 Compliance Testing, describe procedures for determining

the characteristics of coded bitstreams and decoding process• IS 11172-5 Software Simulation• Video format : 352x240 SIF (Source Input Format)• Audio : 64/128/192 kbits/channel

MPEG phase 1

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IS 11172-1

Video

Audio

Sync

IS 11172-2

IS 11172-3

MPEG1 handle video (moving picture) and audio separately, since they both have different characteristic and we use different sense to accept both information (eye for video and ear for audio) with their own limitation.

MPEG phase 1

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How MPEG1 Audio Works

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mappingquantizer

and coding

frame packing

psychoacoustic model

Mapping creates a filtered and subsampled representation of the input audio stream.

Psychoacoustic model creates a set of data to control the quantizer and coding.

Quantizer&Coding creates a set of coding symbols from the mapped input sampless

Frame packing assembles the actual bitstream from the output data of the other blocks, and adds other information (e.g. error correction) if necessary.

MPEG1 – Audio Encoding

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frame unpacking

reconstruction inverse mapping

Frame unpacking unpack and decode block as does error detection if error-check is applied in the encoder. The bit stream data are unpacked to recover the various pieces of information.

Reconstruction block reconstructs the quantized version of the set of mapped samples.

Inverse mapping transforms these mapped samples back into uniform PCM

MPEG1 – Audio Decoding

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Depending on the application, different layers of the coding system with increasing encoder complexity and performance can be used. An ISO MPEG Audio Layer N decoder is able to decode bit stream data which has been encoded in Layer N and all layers below N.

Layer I contains the basic mapping of the digital audio input into 32 sub-bands, fixed segmentation to format the data into blocks, a psychoacoustic model to determine the adaptive bit allocation, and quantization using block companding and formatting.

Layer II provides additional coding of bit allocation, scale factors and samples. Different framing is used.

Layer III increased frequency resolution (576 sub-bands) based on a hybrid filterbank (filterbank + MDCT). It adds a different (non-uniform) quantizer, adaptive segmentation and entropy coding of the quantized values.

MPEG1 – Audio Layer

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• All Layers use the same analysis filterbank (polyphase with 32 subbands). Layer-3 adds a MDCT transform to increase the frequency resolution.

• All Layers use the same "header information" in their bitstream, to support the hierarchical structure of the standard.

• All Layers use a bitstream structure that contains parts that are more sensitive to biterrors ("header", "bit allocation", "scalefactors", "side information") and parts that are less sensitive ("data of spectral components").

• All Layers may use 32, 44.1 or 48 kHz sampling frequency.• All Layers are allowed to work with similar bitrates:

• Layer-1: from 32 kbps to 448 kbps• Layer-2: from 32 kbps to 384 kbps• Layer-3: from 32 kbps to 320 kbps

• From Layer-1 to Layer-3:• Complexity increases• Codec Delay increases• Performance increases (sound quality per bitrate)

MPEG1 – Audio Layer

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Layer I and II

Part of the bit stream that is decodable by itself. In Layer I it contains information for 384 samples and in Layer II for 1152 samples. It starts with a syncword, and ends just before the next syncword. It consists of an integer number of slots (four bytes in Layer I, one byte in Layer II).

Layer III

Part of the bit stream that is decodable with the use of previously acquired side and main information. In Layer III it contains information for 1152 samples. Although the distance between the start of consecutive syncwords is an integer number of slots (one byte in Layer III), the audio information belonging to one frame is generally not contained between two successive syncwords

MPEG1 – Audio Frame

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MPEG1 Audio Layer 3

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Inner iteration loop (rate control loop)Huffman code tables assign shorter code words to (more frequent) smaller quantized values. When the code word exceeds the available block, it can be corrected by adjusting quantization step size, leading to smaller quantized values. This adjustment is repeated until the resulting Huffman coding is small enough.This loop is called rate loop because it modifies overall coder rate until it is small enough.

Outer iteration loop (noise control loop)To shape the quantization noise according to the masking threshold (supplied by the perceptual model), scalefactors are applied to each subband. If the quantization noise in a given subband exceed the masking threshold (allowed noise), the scalefactor for this subband is adjusted to reduce quantization noise.Since achieving a smaller quantization noise requires a larger number of quantization steps and thus a higher bit-rate, the rate adjustment loop has to be repeated every time new scalefactors are used.Noise control loop is executed until the quantization noise is below the masking threshold for every scalefactor subband.

Control Loop

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Comparison

CriteriaMPEG1 layer1

MPEG1 layer 2

MPEG1 layer3

MPEG2MPEG2

AAC

input PCM sample

32kHz, 44.1kHz, 48kHz16, 22.05,

24kHz8 – 96 kHz

sound mode mono, dual channel, joint stereo, stereo + 5.1 channelup to 48 channel

filterbank polyphase filterbank + MDCT

decoding active frame active + previous frame

output bit rate for near-CD quality

384 kbps 256 kbps 128 kbps 96 kbps

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