Abstract: A hybrid speech encoder detects changes from music-like sounds to speech-like sounds. When the encoder detects music-like sounds (e.g., music), it operates in a first mode, in which it employs a frequency domain coder. When the encoder detects speech-like sounds (e.g., human speech), it operates in a second mode, and employs a time domain or waveform coder. When a switch occurs, the encoder backfills a gap in the signal with a portion of the signal occurring after the gap.

Abstract: Apparatus (119) for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor (119) which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k?1 subsequent frames to values, such that the values of the n bits of the k?1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Abstract: A hybrid speech encoder detects changes from music-like sounds to speech-like sounds. When the encoder detects music-like sounds (e.g., music), it operates in a first mode, in which it employs a frequency domain coder. When the encoder detects speech-like sounds (e.g., human speech), it operates in a second mode, and employs a time domain or waveform coder. When a switch occurs, the encoder backfills a gap in the signal with a portion of the signal occurring after the gap.

Abstract: A method and apparatus provides for encoding an audio signal. A bit rate value is received. A set of energy thresholds based on the bit rate value is selected. The set of energy thresholds is one of a plurality of sets of energy thresholds. The energy thresholds of each set of energy thresholds correspond on a one-to-one basis with a set of sub-bands of the audio signal. The audio signal is received. The energy of each sub-band of the set of sub-bands is determined. A highest frequency sub-band that has an energy exceeding the corresponding threshold is determined. A selected bandwidth of the audio signal is encoded. The selected bandwidth includes only those frequencies of the audio signal that are in the highest frequency sub-band that has an energy exceeding the corresponding threshold, as well as the lower frequencies of the audio signal that are above a high-pass cut-off frequency.

Abstract: A method and apparatus provides for encoding an audio signal. A bit rate value is received. A set of energy thresholds based on the bit rate value is selected. The set of energy thresholds is one of a plurality of sets of energy thresholds. The energy thresholds of each set of energy thresholds correspond on a one-to-one basis with a set of sub-bands of the audio signal. The audio signal is received. The energy of each sub-band of the set of sub-bands is determined. A highest frequency sub-band that has an energy exceeding the corresponding threshold is determined. A selected bandwidth of the audio signal is encoded. The selected bandwidth includes only those frequencies of the audio signal that are in the highest frequency sub-band that has an energy exceeding the corresponding threshold, as well as the lower frequencies of the audio signal that are above a high-pass cut-off frequency.

Abstract: A method for estimating a parameter for low bit rate stereo transmission that includes deriving estimate of any time delay between left and right audio channels in a multi-channel signal from a time delay subsystem. A cross-correlation between the left and right audio channels in the time delay subsystem is employed. Thereafter a normalized cross-correlation within an inter-channel intensity difference (IID) processor is employed before deriving estimate of panning gains for the left and right audio channels from the IID processor.

Abstract: A method for estimating a parameter for low bit rate stereo transmission that includes deriving estimate of any time delay between left and right audio channels in a multi-channel signal from a time delay subsystem. A cross-correlation between the left and right audio channels in the time delay subsystem is employed. Thereafter a normalized cross-correlation within an inter-channel intensity difference (IID) processor is employed before deriving estimate of panning gains for the left and right audio channels from the IID processor.

Abstract: A system, method, and apparatus for separating speech signal from a noisy acoustic environment. The separation process may include directional filtering, blind source separation, and dual input spectral subtraction noise suppressor. The input channels may include two omnidirectional microphones whose output is processed using phase delay filtering to form speech and noise beamforms. Further, the beamforms may be frequency corrected. The omnidirectional microphones generate one channel that is substantially only noise, and another channel that is a combination of noise and speech. A blind source separation algorithm augments the directional separation through statistical techniques. The noise signal and speech signal are then used to set process characteristics at a dual input noise spectral subtraction suppressor (DINS) to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify speech signal.

Abstract: A communication device contains multiple microphones that receive acoustic signals from a user and from the background. The acoustic signals from the user are enhanced using the background acoustic signals to reduce background noise. The enhanced signal are transmitted to an emergency network when an emergency call is made from the communication device. The raw signals are stored in the communication device for later retrieval or are transmitted simultaneously with the enhanced signals. The enhanced signals are transmitted using a circuit-switched voice mode while the raw signals are transmitted using a packet-switched voice mode.

Abstract: Apparatus (119) for encoding at least one parameter associated with a signal source for transmission over k frames to a decoder comprises a processor (119) which is configured in operation to assign a predetermined bit pattern to n bits associated with the at least one parameter of a first frame of k frames and set the n bits associated with the at least one parameter of each of k?1 subsequent frames to values, such that the values of the n bits of the k?1 subsequent frames represent the at least one parameter. The predetermined bit pattern indicates a start of the at least one parameter.

Abstract: A communication device contains multiple microphones that receive acoustic signals from a user and from the background. The acoustic signals from the user are enhanced using the background acoustic signals to reduce background noise. The enhanced signal are transmitted to an emergency network when an emergency call is made from the communication device. The raw signals are stored in the communication device for later retrieval or are transmitted simultaneously with the enhanced signals. The enhanced signals are transmitted using a circuit-switched voice mode while the raw signals are transmitted using a packet-switched voice mode.

Abstract: A system, method, and apparatus for separating speech signal from a noisy acoustic environment. The separation process may include directional filtering, blind source separation, and dual input spectral subtraction noise suppressor. The input channels may include two omnidirectional microphones whose output is processed using phase delay filtering to form speech and noise beamforms. Further, the beamforms may be frequency corrected. The omnidirectional microphones generate one channel that is substantially only noise, and another channel that is a combination of noise and speech. A blind source separation algorithm augments the directional separation through statistical techniques. The noise signal and speech signal are then used to set process characteristics at a dual input noise spectral subtraction suppressor (DINS) to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify speech signal.