公开(公告)号：DE69912860T2

公开(公告)日：2004-11-04

申请号：DE69912860

申请日：1999-09-29

Applicant: ST MICROELECTRONICS ASIA

Inventor： TIAN WENSHUN ,  LEONG YUEN ,  ALVAREZ-TINOCO MARIO

IPC: G06F9/40 ,  G10L19/18

公开(公告)号：DE69808146D1

公开(公告)日：2002-10-24

申请号：DE69808146

申请日：1998-01-12

Applicant: ST MICROELECTRONICS ASIA

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

IPC: H03M7/00 ,  H04B1/66

Abstract: A method and apparatus for coding audio data in a frequency transform digital audio coder employing differential frequency coefficient exponent coding. Differential coding of exponents places constraints on possible values an exponent can take, which can lead to distortion in the decoded and reconstructed audio signal. The method and apparatus herein can overcome this restriction by mapping the input exponent set to a new set of values which satisfy the differential constraint as well as reducing information loss, thereby minimizing overall signal distortion due to coding restrictions.

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：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.

公开(公告)号：DE69712230D1

公开(公告)日：2002-05-29

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.