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.

公开(公告)号：DE60332899D1

公开(公告)日：2010-07-22

申请号：DE60332899

申请日：2003-07-23

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR MOHAMMED JAVED ,  GEORGE SAPNA

IPC: G10L19/025

公开(公告)号：DE602004015409D1

公开(公告)日：2008-09-11

申请号：DE602004015409

申请日：2004-09-27

Applicant: ST MICROELECTRONICS ASIA

Inventor： KABI PRAKASH PADHI ,  GEORGE SAPNA

IPC: G10L25/90

Abstract: Pitch detection of speech signals finds numerous applications in karaoke, voice recognition and scoring applications. While most of the existing techniques rely on time domain methods, the invention utilises frequency domain methods. There is provided a method and system for determining the pitch of speech from a speech signal; the method including the steps of: producing or obtaining the speech signal; distinguishing the speech signal into voiced, unvoiced or silence sections using speech signal energy levels; applying a Fourier Transform to the speech signal and obtaining speech signal parameters; determining peaks of the Fourier transformed speech signal; tracking the speech signal parameters of the determined peaks to select partials; and, determining the pitch from the selected partials using a two-way mismatch error calculation.

公开(公告)号：SG142294A1

公开(公告)日：2008-05-28

申请号：SG2007175755

申请日：2007-11-07

Applicant: ST MICROELECTRONICS ASIA

Inventor： ZONG WENBO ,  WU YUAN ,  GEORGE SAPNA

Abstract: ENVIRONMENTAL EFFECTS GENERATOR FOR DIGITAL AUDIO SIGNALS An device and method of generating environmental reverberation effects for digital audio signals is presented. The device includes a reverberation controller. The reverberation controller pre-processes one or more predetermined characteristics of a first audio signal to produce a pre- processed signal and generates a plurality of delayed outputs from the pre- processed signal, each output having a predetermined delay. The reverberation controller also produces a plurality of reflection outputs from the plurality of delayed outputs and combines the plurality of reflection outputs to produce a second audio signal having a desired reverberation response.

公开(公告)号：DE602004004225D1

公开(公告)日：2007-02-22

申请号：DE602004004225

申请日：2004-09-27

Applicant: ST MICROELECTRONICS ASIA

Inventor： KABI PRAKASH PADHI ,  GEORGE SAPNA

IPC: G10L25/78

Abstract: A method for determining whether a data frame of a coded speech signal corresponds to voice or to noise, including the steps of determining the cross-correlation of the data of said data frame; determining the periodicity of the cross-correlation; determining the variance of the periodicity; determining said data frame corresponds to noise if the cross-correlation is lower than a predetermined cross-correlation value; and determining the data corresponds to voice if the variance is less than a predetermined variance value.

公开(公告)号：SG120121A1

公开(公告)日：2006-03-28

申请号：SG200305743

申请日：2003-09-26

Applicant: ST MICROELECTRONICS ASIA

Inventor： PRAKASH PADHI KABI ,  GEORGE SAPNA

IPC: G10L25/90

Abstract: Pitch detection of speech signals finds numerous applications in karaoke, voice recognition and scoring applications. While most of the existing techniques rely on time domain methods, the invention utilises frequency domain methods. There is provided a method and system for determining the pitch of speech from a speech signal; the method including the steps of: producing or obtaining the speech signal; distinguishing the speech signal into voiced, unvoiced or silence sections using speech signal energy levels; applying a Fourier Transform to the speech signal and obtaining speech signal parameters; determining peaks of the Fourier transformed speech signal; tracking the speech signal parameters of the determined peaks to select partials; and, determining the pitch from the selected partials using a two-way mismatch error calculation.

公开(公告)号：SG120118A1

公开(公告)日：2006-03-28

申请号：SG200305637

申请日：2003-09-15

Applicant: ST MICROELECTRONICS ASIA

Inventor： PRAKASH PADHI KABI ,  KUMAR KASARGOD SUDHIR ,  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：DE69928842D1

公开(公告)日：2006-01-12

申请号：DE69928842

申请日：1999-10-30

Applicant: ST MICROELECTRONICS ASIA

Inventor： ABSAR JAVED ,  GEORGE SAPNA

IPC: G10L19/008 ,  G10L19/16 ,  H04B1/66

Abstract: Channel coupling for an AC-3 encoder, using mixed precision computations and 16-bit coupling coefficient calculations for channels with 32-bit frequency coefficients.

公开(公告)号：DE69823557T2

公开(公告)日：2005-02-03

申请号：DE69823557

申请日：1998-02-21

Applicant: ST MICROELECTRONICS ASIA

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

IPC: H04H5/00 ,  H04H20/89 ,  H04H1/00

公开(公告)号：DE69808146T2

公开(公告)日：2003-05-15

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.