Introduction to MPEG-1/2 L2 Audio 尤信程國立台北科技大學資訊工程系

Transcription

1 Introduction to MPEG-1/2 L2 Audio 尤信程國立台北科技大學資訊工程系

2 Contents Intro to audio coding Psychoacoustics Audio encoding Frame structure Audio decoding Conclusions

3 Intro to audio coding (1) Audio coding uses MPEG-1/2 (part 3) Layer II, originally developed by Philips MPEG-1 1 Layer III audio is known as MP-3 3 (by FhG-IIS) Sampling rate: MPEG-1 1 = 48 ks/s, MPEG-2 2 = 24 ks/s Each audio frame encodes 1152 PCM samples per channel. Audio frame duration: MPEG-1 1 = 24 ms, MPEG-2 2 = 48 ms.

4 Intro to audio coding (2) DAB audio uses MPEG-1 1 Layer II & MPEG-2 LSF Layer II coding DVB-T T audio uses MPEG-1 1 Layer II DVB-T T audio can optionally use other audio coding technique such as MPEG-2 2 MC Layer II & Dolby AC-3 DAB also uses PAD for carrying data DVB-T T does not use PAD as it can transmit data via TS

5 Intro to audio coding (3) MPEG-1 1 audio for higher bitrate, MPEG-2 2 for lower bitrate (lower audio quality). Contrary to common believing Lower sampling freq for low bitrate coding: common practice in audio society MPEG-2 2 Multi-Channel (MC) extension is not currently used in Taiwan MPEG-4 4 HE-AAC could be a future coding scheme

6 Intro to audio coding (4) Features of MPEG-1/2 audio standard psychoacoustic model 32-subband filterbank Bit allocation table SCaleFactor Selection Info (SCFSI) Data grouping Bitrate range: 32 kbps ~ 384 kbps

7 Psychoacoustics (1) Brief intro to psychoacoustics Absolute hearing threshold Frequency masking (main factor) Temporal masking Layer I/II uses psychoacoustic model I Layer III uses psychoacoustic model II

8 Psychoacoustics (2) Absolute hearing threshold is the minimum level of sound human beings can hear Every one has (slightly) different threshold SPL (Sound Pressure Level) = 20 log (P/P( 0 ) db. P 0 = 2 x 10 5 N/m 2 SPL Freq

9 Psychoacoustics (3) Frequency masking: Human ears can be modeled as many bandpass filters. Different signals within the passband of one filter will interfere each other. Stronger one masks weaker one. SPL Freq

10 Psychoacoustics (4) Passband of a bandpass filter is known as one critical band. There are around 25 critical bands. Noise Masks Tone (NMT) ~ -66 db Tone Masks Noise (TMN) ~ -18 db Inter-critical critical-band masking also exists Frequency masking assumes stationary signals. Not correct for transient signals.

11 Psychoacoustics (5) Temporal masking Pre-masking (5 ms) Simultaneous masking Post-masking (200 ms) Pre-masking can be used to mask pre-echoes. echoes. In general, temporal masking is more difficult to use.

12 Audio encoding (1) Simple block diagram of MPEG-1/2 encoding. Digital Audio Input Filter Bank Bit allocation and scalefact or calculatio n Quantized Samples Bitstream Formatting Encoded Bitstream Signal to Mask Ratio Psychoacoustic Model

13 Audio encoding (2) Filter-bank, also called subband analysis, has 32 bandpass filters with the same proto-type. type. Let g(n) ) be the impulse response of proto-type. type. Filter i in the filter bank is obtained by (2i + 1)( n 16) π h i ( n) = g( n) cos( ) 64 Recall MPY in time domain is convolution in frequency domain. Center frequency is shifted according to cosine term.

14 Audio encoding (3) The proto-type type filter is a 512-coefficient FIR filter. Coefficients are in the spec. Since the BW of each BPF is (1/32) of full BW, decimate its output samples by 32. PCM input h 0 (n) 32 h 31 (n) 32 Subband output

15 Audio encoding (4) Number of bits to quantize a subband sample based on PSY model is recorded in Bit Allocation (BA). 3 X 12 = 36 samples from the same subband share one BA info. 12 samples of one subband = 1 part. The amplification ratio of a subband sample in encoding is called SCaleFactor (SCF). It s shared by 12 subband samples.

16 Audio encoding (5) It s s possible to use one, two, or three SCF s s for 36 output samples of one subband. This info is called SCaleFactor Selection Information (SCFSI). Four different cases identified: Three different SCF s First-two two parts share one SCF Last-two two parts share one SCF All three parts share one SCF

17 Audio encoding (6) 1152 PCM samples per ch is packed in one audio frame. Subband samples in bitstream: 3 samples from L ch of SB 0 3 samples form R ch of SB 0 Same for SB 1 till SB 31 Repeat the above 12 times Remember SCF scope is 12 samples from the same SB.

18 Audio encoding (7) Quantized subband samples may be grouped together into one codeword. The following quantization levels use grouping in packing subband samples. 3 levels: use 5 bits 5 levels: use 7 bits 9 levels: use 10 bits

19 Audio frame structure (1) The audio frame in MPEG-1/2 has the following fields: header, CRC, audio data, and ancillary. Audio data has BA field, SCFSI field, SCF field, and sb sample field. DAB uses ancillary to pack F-PAD F (Fixed Program Associated Data), X-PAD X (Extended PAD), and SCF-CRC. CRC.

20 Audio frame structure (2) CRC field in MPEG-1/2 is optional, but is mandatory in DAB. Header CRC Audio data Ancillary BA SCFSCI SCF Subband samples Stuffing X-PAD SCF CRC F-PAD

21 Audio frame structure (3) SCF CRC = CRC for next frame s s SCF. F-PAD contains the following Dynamic Range Control (DRC) X-PAD info (no, short, long) Music/Speech flag ISRC (Int l l Standard Recording Code)

22 Audio frame structure (4) X-PAD has many application types, including label data ITTS (Interactive Text Transmit System) MOT (Multimedia Object Transfer protocol) etc.

23 Audio frame structure (5) Frame header has 32 bits, arranged as follows: AAAAAAAA AAAABCCD EEEEFFGH IIJJKLMM A = sync word = B = ID. 0 = MEPG-2, 1 = MPEG-1 C = layer ID. 10 = Layer 2. (DAB uses it only) D = protection bit. Use 0 (with CRC) for DAB. E = bitrate index. From 32 kbps ~ 384 kbps. F = Sampling frequency. 01 = 48/24 khz. G = padding = 0. No padding required H = private bit. Not used.

24 Audio frame structure (6) AAAAAAAA AAAABCCD EEEEFFGH IIJJKLMM. I = mode. 00 = stereo, 01 = joint stereo, 10 = dual channel, 11 = mono. J = mode extension. Used only if mode = = 4, 01 = 8, 10 = 11 = K = copyright. 0 = no, 1 = yes L = original/copy. 0 = copy, 1 = org. M = emphasis. Use 00 = no emphasis.

25 Audio Decoding (1) Sync frame header by finding Parse frame header to know the coding mode. Bitrate info can also be obtained from FIC. Use assigned BA table to parse BA. Parse SCFSI. No SCFSI for BA field of a SB = 0 Parse SCF based on SCFSI and BA. Parse SB samples. Watch out grouping codeword.

26 Audio Decoding (2) Check if frame CRC and SCF CRC are OK. Do error concealment if necessary. Re-quantize SB samples. Perform subband synthesis via better algorithms than ISO s. Synchronize CODEC s s Fs if necessary.

27 Audio Decoding (3)

28 Audio Decoding (4) Brief introduction to de-grouping and re- quantize the SB samples. 1. De-grouping: for i = 0 to 2 S[i] = c % nlevel c = c div nlevel 2. S S = Invert MSB of S. S S is 2 s-2 complement fractional number. 3. S = = C * (S + D). C & D in tables of spec. 4. S S = S S * SCF

29 Audio Decoding (5) Subband synthesis from ISO s s spec: Input 32 new subband samples, S i, i = 0,.., 31 Shifting: for i =1023 down to 64 do V[i] = V[i-64] Matrixing: : for i = 0 to 63 do + + = 31 (16 i)(2k 1) π V[i] cos k= 0 64 S k

30 Audio Decoding (6) Build a 512 vector U: for i = 0 to 7 do for j = 0 to 31 do U[i*64+j] = V[i*128+j] U[i*64+32+j] = V[i* j] Window by 512-coefficient matrix D: for i = 0 to 511 do W[i] = U[i] * D[i]

31 Audio Decoding (7) Calculate 32 PCM samples: for j = 0 to 31 do s j = 15 i= 0 W[j + 32i] Output 32 reconstructed PCM samples s j.

32 Audio Decoding (8) Subband synthesis algorithm mentioned previously is slow We can avoid data shifting by using circular queue. Matrixing can be more efficient if DCT is in use. Cf. IEEE SPL, vol. 1, no. 2, pp , A fully optimized code can be at least five times faster!

33 Conclusions Introduction to audio coding Brief intro to psychoacoustics Audio encoding flow Audio frame structure Audio decoding, including subband synthesis