公开(公告)号：DE602004004846D1

公开(公告)日：2007-04-05

申请号：DE602004004846

申请日：2004-09-14

Applicant: ST MICROELECTRONICS ASIA

Inventor： SUDHIR KUMAR KASARGOD ,  KABI PRAKASH PADHI ,  GEORGE SAPNA

IPC: G10L19/025 ,  G10L19/035

Abstract: An MPEG-1 layer 3 audio encoder, including a scalefactor generator for determining first scalefactors for encoding a block of audio data if a temporal masking transient is not detected in said block of audio data; and for selecting the maximum of said scalefactors for encoding said block of audio data if a temporal masking transient is detected in said block of audio data to enable greater compression of said audio data. Increases in quantization error due to use of the maximum scalefactor are pre-masked or post-masked by the temporal masking transient. In cases where the last portion of a block includes a temporal masking transient that masks the preceding portions of the block, the maximum scalefactor is only used to encode the block if the resulting increase in quantization error is less than 30% of the quantization error for the block.

公开(公告)号：DE60033443D1

公开(公告)日：2007-03-29

申请号：DE60033443

申请日：2000-06-23

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR MOHAMMED JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO ANTONIO MARIO

IPC: H03H17/06

公开(公告)号：DE69734782D1

公开(公告)日：2006-01-05

申请号：DE69734782

申请日：1997-09-26

Applicant: ST MICROELECTRONICS ASIA

Inventor： HUI WAI ,  GEORGE SAPNA

IPC: H04H20/88

Abstract: A method and apparatus for decoding a bitstream (100) of transform coded multi-channel audio data. The bitstream is subjected to a block decoding process (101) to obtain for each input audio channel within the multi-channel audio data a corresponding block of frequency coefficients (102). Each block of frequency coefficients (102) is assigned a higher precision inverse transform or a lower precision inverse transform according to predetermined characteristics of the audio data represented by the block. The blocks of frequency coefficients are subsequently subjected to the assigned transform (105, 106) and an output audio signal (108) is generated in response to each of the higher and lower precision inverse transform processes.

公开(公告)号：DE69826529T2

公开(公告)日：2005-09-22

申请号：DE69826529

申请日：1998-04-15

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR MOHAMMED JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO MARIO

IPC: G10L19/002 ,  G10L19/00

Abstract: In a transform encoder for audio data, encoded data in the form of mantissas, exponents and coupling data is packed into fixed length frames in an output bitstream. The fields within the frame for carrying the different forms of data are variable in length, and apace within the frame must be allocated between them to fit all of the required information into the frame. The space required by the various data types depends on certain encoding parameters, which are calculated for a particular frame before the data is encoded, thus ensuring that the encoded data will fit into the frame before the computationally expensive encoding process is carried out. Information in relation to, for example, transform length, coupling parameters and exponent strategy are determined, which allows the space required for the coupling and exponent data to be calculated. The mantissa encoding parameters can then be iteratively determined so that the encoded mantissas will fit into the frame with the other encoded data. The determined encoding parameters are stored and the audio data is encoded according to those parameters after it has been determined that the encoded data will fit into the frame.

公开(公告)号：DE69826529D1

公开(公告)日：2004-10-28

申请号：DE69826529

申请日：1998-04-15

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR MOHAMMED JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO MARIO

IPC: G10L19/002 ,  G10L19/00

Abstract: In a transform encoder for audio data, encoded data in the form of mantissas, exponents and coupling data is packed into fixed length frames in an output bitstream. The fields within the frame for carrying the different forms of data are variable in length, and apace within the frame must be allocated between them to fit all of the required information into the frame. The space required by the various data types depends on certain encoding parameters, which are calculated for a particular frame before the data is encoded, thus ensuring that the encoded data will fit into the frame before the computationally expensive encoding process is carried out. Information in relation to, for example, transform length, coupling parameters and exponent strategy are determined, which allows the space required for the coupling and exponent data to be calculated. The mantissa encoding parameters can then be iteratively determined so that the encoded mantissas will fit into the frame with the other encoded data. The determined encoding parameters are stored and the audio data is encoded according to those parameters after it has been determined that the encoded data will fit into the frame.

公开(公告)号：DE69909849T2

公开(公告)日：2004-05-27

申请号：DE69909849

申请日：1999-11-24

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO MARIO

IPC: G10H1/12 ,  G10H7/12

Abstract: A method for effecting aliasing cancellation in an audio effects algorithm using a delay modulated signal, derived from interpolation of a delay modulator at an instantaneous sampling frequency, including: determining the instantaneous sampling frequency 1/T isf and band limiting an input signal, to which the audio effects algorithm is to be applied to ½ T isf prior to interpolation.

公开(公告)号：DE69722973T2

公开(公告)日：2004-05-19

申请号：DE69722973

申请日：1997-12-19

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO MARIO

IPC: H04B1/66 ,  H04S1/00

Abstract: A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.

公开(公告)号：DE69908433T2

公开(公告)日：2004-04-08

申请号：DE69908433

申请日：1999-10-30

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR MOHAMMED JAVED ,  GEORGE SAPNA

IPC: G10L19/16 ,  G10L19/14 ,  H04B1/66

公开(公告)号：DE69909849D1

公开(公告)日：2003-08-28

申请号：DE69909849

申请日：1999-11-24

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR JAVED ,  GEORGE SAPNA ,  ALVAREZ-TINOCO MARIO

IPC: G10H1/12 ,  G10H7/12

Abstract: A method for effecting aliasing cancellation in an audio effects algorithm using a delay modulated signal, derived from interpolation of a delay modulator at an instantaneous sampling frequency, including: determining the instantaneous sampling frequency 1/T isf and band limiting an input signal, to which the audio effects algorithm is to be applied to ½ T isf prior to interpolation.

公开(公告)号：DE69712230T2

公开(公告)日：2002-10-31

申请号：DE69712230

申请日：1997-05-08

Applicant: ST MICROELECTRONICS ASIA

Inventor： ALVAREZ-TINOCO MARIO ,  GEORGE SAPNA ,  YANG HAIYUN

IPC: H04S1/00

Abstract: An audio decoder solution is here provided where a reduction in computing power is required. The proposed method consists of forcing the multiple output channels to only one type of inverse transformation format. A format of long transform length is more suitable for input signals whose spectrum remains stationary or quasi-stationary. This provides a greater frequency resolution, improved coding performance and a reduction of computing power required. Another format of two or more short transform lengths, possessing greater time resolution, is more desirable for rapidly changing signals with time. The computer power required for two or more short transforms should be higher than for only one transformation. The time versus frequency resolution trade-off should be considered when selecting a transform block length. Advantage is taken of human hearing behaviour to reduce the computing power of a processing engine (e.g. DSP) when downmixing from an M-channel input to a P-channel output is required. The encoder provides spectral information concerning the transmitted audio signal frame. This information corresponds to signals which are stationary/quasi-stationary or changing rapidly with time. Some analysis is required to decide which input channels are forced to long or short block conversion prior to frequency-domain downmixing and transformation.