摘要:
An encoder and decoder for processing an audio signal including generic audio and speech frames are provided herein. During operation, two encoders are utilized by the speech coder, and two decoders are utilized by the speech decoder. The two encoders and decoders are utilized to process speech and non-speech (generic audio) respectively. During a transition between generic audio and speech, parameters that are needed by the speech decoder for decoding frame of speech are generated by processing the preceding generic audio (non-speech) frame for the necessary parameters. Because necessary parameters are obtained by the speech coder/decoder, the discontinuities associated with prior-art techniques are reduced when transitioning between generic audio frames and speech frames.
摘要:
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.
摘要:
A method and apparatus for encoding a signal is provided herein. During operation a wideband signal that is to be encoded enters a filter bank. A highband signal and a lowband signal are output from the filter bank. Each signal is separately encoded. During the production of the highband signal, a downmixing operation is implemented after preprocessing, and prior to decimating. The downmixing operation greatly reduces system complexity. In fact, it will be observed that the highest sample rate in the prior-art implementation is 64 kHz whereas the sample rate in the system described above remains at 32 kHz or below. This represents a significant complexity saving, as do the reduced number of processing blocks.
摘要:
During operation an input signal to be coded is received and coded to produce a coded audio signal. The coded audio signal is then scaled with a plurality of gain values to produce a plurality of scaled coded audio signals, each having an associated gain value and a plurality of error values are determined existing between the input signal and each of the plurality of scaled coded audio signals. A gain value is then chosen that is associated with a scaled coded audio signal resulting in a low error value existing between the input signal and the scaled coded audio signal. Finally, the low error value is transmitted along with the gain value as part of an enhancement layer to the coded audio signal.
摘要:
An apparatus and method for synchronization between uncoordinated Time Division Duplex (TDD) communication networks includes a first step (300) of measuring an interference level on channels available to a base station. A next step (302) includes choosing the channel having the lowest interference level. A next step (304) includes determining that the interference is from a base station. A next step (306) includes calculating an interference profile over the frame cycle. A next step (308) includes establishing a peak interference level. A next step (310) includes aligning the base station frame timing in response to the peak interference level.
摘要:
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.
摘要:
An apparatus and method for synchronization between uncoordinated Time Division Duplex (TDD) communication networks includes a first step (300) of measuring an interference level on channels available to a base station. A next step (302) includes choosing the channel having the lowest interference level. A next step (304) includes determining that the interference is from a base station. A next step (306) includes calculating an interference profile over the frame cycle. A next step (308) includes establishing a peak interference level. A next step (310) includes aligning the base station frame timing in response to the peak interference level.
摘要:
In a selective signal encoder, an input signal is first encoded using a core layer encoder to produce a core layer encoded signal. The core layer encoded signal is decoded to produce a reconstructed signal and an error signal is generated as the difference between the reconstructed signal and the input signal. The reconstructed signal is compared to the input signal. One of two or more enhancement layer encoders selected dependent upon the comparison and used to encode the error signal. The core layer encoded signal, the enhancement layer encoded signal and the selection indicator are output to the channel (for transmission or storage, for example).
摘要:
An audio signal processing system includes parallel speech and generic audio signal processing paths. One path includes a linear predictive coder and a resampling filter having a non-linear phase characteristic. A phase compensation filter is disposed along the one of the processing paths to compensate for the non-linearity of the resampling filter thereby enabling relatively seamless switching between the coders resulting in a reduction of audio artifacts that would otherwise result from the non-linear phase characteristic of the resampling filter during playback.
摘要:
A method for encoding audio frames by producing a first frame of coded audio samples by coding a first audio frame in a sequence of frames, producing at least a portion of a second frame of coded audio samples by coding at least a portion of a second audio frame in the sequence of frames, and producing parameters for generating audio gap filler samples, wherein the parameters are representative of either a weighted segment of the first frame of coded audio samples or a weighted segment of the portion of the second frame of coded audio samples.