Abstract: A voice activity detection apparatus enabling the decision on active interval accurately regardless of time elapse is sought. Apparatus 10 comprises autocorrelation calculating unit 11 calculating autocorrelation value of input signal, delay calculating unit 12 calculating delay for calculated autocorrelation value becoming maximum, noise deciding unit 13 deciding whether input signal is noise or not based on calculated delay, noise estimating unit 14 estimating noise from input signal, activity deciding unit 15 performing activity decision regarding input signal based on result of decision by noise deciding unit 13, noise estimated by noise estimating unit 14, and input signal, and a sound interval detecting unit 16 counting time duration of active interval based on decision result by deciding unit 15. In case of time duration of active interval reaches a predetermined period or more, noise estimating unit 14 changes noise estimating method such that input signal is likely decided as active.

Abstract: It is provided a voice activity decision apparatus capable of accurately performing the decision on the state being associated with a sound interval or a silence interval also in terms of the input signal having many aperiodic components and/or plural mixed different periodic components. The apparatus 1 comprises: an autocorrelation calculating unit 11 for calculating autocorrelation values of an input signal; a delay calculating unit 12 for calculating plural delays at which autocorrelation values calculated by the autocorrelation calculating unit 11 become maximums; a noise deciding unit 13 for deciding whether the input signal is a noise or not based on the plurality of delays calculated by the delay calculating unit 12; and an activity decision unit 14 for performing the activity decision in terms of the input signal based on results of decision by the noise deciding unit 13 and the input signal.

Abstract: The present invention intends to render quantization noise virtually imperceptible for a user and to prevent reduction in frequency resolution and reduction in encoding efficiency.

Abstract: A decoding processing portion 11 of a speech decoder 10 is provided with an emphasis processing portion 15 for performing an emphasis process on signals to be processed (excited signals) SPC generated from coded speech signals BS. A counter portion 17 counts the number of times code errors occurred in successive frames of the coded speech signal BS, and outputs the successive frame error number. When the successive frame error number outputted form the counter portion 17 is less than or equal to a preset reference successive frame error number, a first switch SW1 and second switch SW2 are set to an emphasis processing portion 15 side. Accordingly, the signals to be processed SPC generated from various parameters included in the coded speech signals are supplied through the switch SW1 to the emphasis processing portion 15 of the decoding processing portion 11 to perform an emphasis process.

Abstract: A coding standard selecting method for selecting a coding standard used for communications between a first terminal and a second terminal, each terminal supporting multiple coding standards, is disclosed. The first terminal transmits a first signal representing all the coding standards supported by the first terminal itself to the second terminal. The second terminal selects a set of coding standards that satisfies a predetermined condition from among the coding standards represented by the first signal and among the coding standards supported by the second terminal itself, and transmits a second signal representing the selected set of coding standards to the first terminal. The first terminal selects a coding standard to be used for communication with the second terminal from the set of coding standards represented by the second signal, and transmits a third signal representing the selected coding standard to the second terminal.

Abstract: A variable length frame transmission method making it possible to accurately and easily establish synchronism at the receiver side without redundancy of system under an environment in which a code error easily occurs. In a transmitter, a variable length frame division section 1 divides a variable length frame f into code strings f1 and f2 having a length ratio of 1:1. A first synchronization flag addition section 3-1 adds a synchronization flag S1 to the head of the code string f1 and a second synchronization flag addition section 3-2 adds a synchronization flag S2 to the head of the code string f2. The synchronization flags have contents different from each other, but they have the same length. Code strings having synchronization flags are multiplexed by a changeover switch 4 and formed into a variable length frame. A series of variable length frames obtained from the changeover switch 4 are transmitted to a receiver as serial data.

Abstract: A coding device capable of improving the coding efficiency and a decoding device for decoding a code sequence generated by the coding device are provided. In the coding device., for each of the possible block combinations obtained when dividing a frame, a coding unit encodes each block in the frame block by block at different bit rates, and at the same time, the coding unit decodes the resultant code sequences related to the frame. A calculation unit calculates the error powers of the decoded signals and the input signal. A determination unit selects a code sequence that makes the average bit rate in coding the frame not higher than a specified value and the corresponding error power a minimum. This selected code sequence is output.

Abstract: A speech decoder 10 comprises a decoding processing portion 11 and an amplification process control portion 12. Here, the decoding processing potion 11 is a device for decoding a received coded speech signal (bitstream) BS and outputting a decoded speech signal SP. Additionally, the amplification process control portion 12 monitors the state of occurrence of frame errors in the coded speech signal BS, and when the number of successive frame errors exceeds a predetermined reference frame error number, outputs amplification instructions for a predetermined number of frames after the successive frame errors disappear. As a result, instead of codebook data DCB obtained by a decoding process of the decoding processing portion 11, amplified codebook data DACB are supplied to a synthesis filter portion 17, and is written into the codebook decoder 18 of the decoding processing portion 11 as new original codebook data DCBO.

Abstract: A decoding apparatus is provided. The decoding apparatus has a first decoding part for decoding a code word obtained by encoding an input signal using a Code-Excited Linear Prediction encoding method. A second decoding part decodes a code word obtained by encoding a signal with an encoding method other than the Code-Excited Linear Prediction encoding method. A rising-transition detection and notification part has a detection part that detects the existence of a rising-transition of amplitude of the input signal based on time variation of a gain of excitation vectors obtained by the first decoding part, and a notification part that notifies the second decoding part that the rising-transition of the amplitude exists.

Abstract: The probability of frame destruction is lowered while suppressing the redundancy of the transmission data. On the transmitting side, a predetermined unique word is contained in a frame n for storing the n-th data, and header information n, frame length information and header information n−1 of the frame n−1 one frame before the frame n are subjected to error-correcting coding, contained in the frame n, and transmitted. On the receiving side, the header of the frame n is received. When the frame length information is transmitted with error, the timing is specified by detecting the unique word and header information in the next frame n+1. When the header of the frame n is not successfully decoded, the information data of the frame n is decoded by using the header information n inserted into a predetermined position of the frame n+1.

Abstract: In a voice communication system 1, a gateway server 4 receives IP packets from the Internet, converts PCM voice data in the IP packets into AMR encoded voice data frames, and transmits to a mobile terminal 7. During the propagation to the gateway server 4, there is a possibility of loss of IP packets and crucial bit error in IP packets. In that case, the gateway server 4 puts “No data” data on frames as voice encoded data for the IP packets in question and sends it to the mobile terminal 7. The “No data” data is a target of concealment.

Abstract: A delay unit 103 adds holding time that has been set by a holding time setting unit 104 to a received data. The holding time is computed based on delay time of received data and the minimum delay time of data received up to a certain point for the purpose of reducing a total delay time. The delay time is estimated in a delay time estimating unit 106 from the difference between a reception time of a packet counted based on an internal clock generator 107 and a time designated by a time stamp in the received packet.

Abstract: This invention relates to a double talk detecting method suited to be used for an echo canceler, and more particularly to a method for correctly detecting whether or not the double talk is present, even in the case where a speech transmitting level difference between a far-end talker and a near-end talker is comparatively large. What is assumed in the present invention is an echo canceler which can be employed in a telephone line network for transmitting a speech of a far-end talker sent through a digital line and a speech of a near-end talker sent through an analog line. A double talk detecting method applicable to the echo canceler here extracts predetermined analyzed parameters respectively from a first speech signal corresponding to the speech of the far-end talker sent from the echo canceler to a hybrid circuit and the speech of the near-end talker input to the echo canceler through the hybrid circuit. The analyzed parameters herein used include, for example, speech pitches, frequency components, etc.

Abstract: When errors occur in a received code sequence, an error concealment routine part 130 is actuated and a correct code sequence is generated based on a code sequence received before the errors are detected. A decoding routine part 21' executes a decoding process based on this code sequence and updates the internal state based on the decoded result. Thereafter, when the code errors are recovered, an error recovery routine part 140 is actuated. The error recovery routine part 140 re-estimates a correct code sequence during a time period when the errors are detected, based on a code sequence received before the errors are detected and a code sequence received after no more errors are detected, and generates a second estimated code sequence. The decode process part 21' executes the decoding process based on an internal state information for retaining the second estimated code sequence in an internal state storage part 23', and updates the internal state information based on the decoded result.