摘要:
A stereo sound encoding method and system, for encoding left and right channels of a stereo sound signal, down mix the left and right channels of the stereo sound signal to produce primary and secondary channels and encode the primary and secondary channels. Encoding the primary channel and encoding the secondary channel comprise determining a first bit budget to encode the primary channel and a second bit budget to encode the secondary channel. If the second bit budget is sufficient, the secondary channel is encoded using a four subframes model and, if the second bit budget is insufficient for using the four subframes model, the secondary channel is encoded using a two subframes model.
摘要:
A method and system are implemented in a stereo sound signal encoding system for time domain down mixing right and left channels of an input stereo sound signal into primary and secondary channels. Correlation of the primary and secondary channels of previous frames is determined, and an out-of-phase condition of the left and right channels is detected based on the correlation of the primary and secondary channels of the previous frames. The left and right channels are time domain down mixed, as a function of the detection, to produce the primary and secondary channels using a factor β, wherein the factor β determines respective contributions of the left and right channels upon production of the primary and secondary channels.
摘要:
A stereo sound signal encoding method and system for time domain down mixing right and left channels of an input stereo sound signal into primary and secondary channels, determine normalised correlations of the left channel and right channel in relation to a monophonic signal version of the sound. A long-term correlation difference is determined on the basis of the normalised correlation of the left channel and the normalised correlation of the right channel. The long-term correlation difference is converted into a factor β, and the left and right channels are mixed to produce the primary and secondary channels using the factor β, wherein the factor β determines respective contributions of the left and right channels upon production of the primary and secondary channels.
摘要:
A device and method for quantizing a gain of a fixed contribution of an excitation in a frame, including sub-frames, of a coded sound signal, wherein the gain of the fixed excitation contribution is estimated in a sub-frame using a parameter representative of a classification of the frame. The gain of the fixed excitation contribution is then quantized in the sub-frame using the estimated gain. The device and method is used in jointly quantizing gains of adaptive and fixed contributions of an excitation in a frame of a coded sound signal. For retrieving a quantized gain of a fixed contribution of an excitation in a sub-frame of a frame, the gain of the fixed excitation contribution is estimated using a parameter representative of a classification of the frame, a gain codebook supplies a correction factor in response to a received, gain codebook index, and a multiplier multiplies the estimated gain by the correction factor to provide a quantized gain of the fixed excitation contribution.
摘要:
An audio encoder adapted for encoding frames of a sampled audio signal to obtain encoded frames, wherein a frame includes a number of time domain audio samples. The audio encoder includes a predictive coding analysis stage for determining information on coefficients of a synthesis filter and a prediction domain frame based on a frame of audio samples. The audio encoder further includes a time-aliasing introducing transformer for transforming overlapping prediction domain frames to the frequency domain to obtain prediction domain frame spectra, wherein the time-aliasing introducing transformer is adapted for transforming the overlapping prediction domain frames in a critically-sampled way. Moreover, the audio encoder includes a redundancy reducing encoder for encoding the prediction domain frame spectra to obtain the encoded frames based on the coefficients and the encoded prediction domain frame spectra.
摘要:
An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters. The transform domain path also includes a combiner configured to combine the time-domain representation of the audio content with an aliasing-cancellation synthesis signal to obtain an aliasing reduced time-domain signal.
摘要:
An apparatus for encoding includes a first domain converter, a switchable bypass, a second domain converter, a first processor and a second processor to obtain an encoded audio signal having different signal portions represented by coded data in different domains, which have been coded by different coding algorithms. Corresponding decoding stages in the decoder together with a bypass for bypassing a domain converter allow the generation of a decoded audio signal with high quality and low bit rate.
摘要:
A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.
摘要:
A pitch search method and device for digitally encoding a wideband signal, in particular but not exclusively a speech signal, in view of transmitting, or storing, and synthesizing this wideband sound signal. The new method and device which achieve efficient modeling of the harmonic structure of the speech spectrum uses several forms of low pass filters applied to a pitch codevector, the one yielding higher prediction gain (i.e. the lowest pitch prediction error) is selected and the associated pitch codebook parameters are forwarded.
摘要:
A device and a method for quantizing, in a super-frame including a sequence of frames, LPC filters calculated during the frames of the sequence. The LPC filter quantizing device and method comprises: an absolute quantizer for first quantizing one of the LPC filters using absolute quantization; and at least one quantizer of the other LPC filters using a quantization mode selected from the group consisting of absolute quantization and differential quantization relative to at least one previously quantized filter amongst the LPC filters. For inverse quantizing, at least the first quantized LPC filter is received and an inverse quantizer inverse quantizes the first quantized LPC filter using absolute inverse quantization. If any quantized LPC filter other than the first quantized LPC filter is received, an inverse quantizer inverse quantizes this quantized LPC filter using one of absolute inverse quantization and differential inverse quantization relative to at least one previously received quantized LPC filter.