摘要:
An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.
摘要:
A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.
摘要:
A quantizing method is provided that includes quantizing an input signal by selecting one of a first quantization scheme not using an inter-frame prediction and a second quantization scheme using the inter-frame prediction, in consideration of one or more of a prediction mode, a predictive error and a transmission channel state.
摘要:
A method of processing an audio signal is disclosed. The present invention includes receiving, by an audio processing apparatus, coding identification information indicating whether to apply a first coding scheme or a second coding scheme to a current frame; when the coding identification information indicates that the second coding scheme is applied to the current frame, receiving window type information indicating a particular window for the current frame, from among a plurality of windows; identifying that a current window is a long stop window based on the window type information, wherein the long stop window is followed by only long window of a following frame, wherein the long stop window includes a gentle long stop window and a steep long stop window; and, when the first coding scheme is applied to a previous frame, applying the gentle long stop window to the current frame, wherein: the gentle long stop window comprise an ascending line with first slope, the steep long stop window comprise an ascending line with second slope, and, the first slope is gentler than the second slope.
摘要:
Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.
摘要:
An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter including level information corresponding to at least one object signal, converting the level information corresponding to the object signal to the level information corresponding to an output channel by applying a control parameter to the object parameter, and generating a rendering parameter including the level information corresponding to the output channel to control an object downmix signal resulting from downmixing the object signal.
摘要:
A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.
摘要:
Lost frame reconstruction is described. A previous good or reconstructed frame may be analyzed to determine a category for the lost frame. A percentage Pi may be associated with the determined category of the lost frame. A top Pi percent magnitude samples may be zeroed out in an excitation of the previous good or reconstructed frame to produce a reconstruction excitation. The reconstruction excitation may be applied to one or more linear prediction coefficients for the previous good or reconstructed frame to generate a reconstructed frame.
摘要:
A method and apparatus multiplies a past sample a time lag τ older than a current sample by a quantized multiplier ρ′ on a frame by frame basis, subtracts the multiplication result from the current sample, codes the subtraction result, and codes the time lag using a fixed-length coder if the multiplier ρ′ is smaller than 0.2 or if information about the previous frame is unavailable, or codes the time lag using a variable-length coder if ρ′ is not smaller than 0.2. A multiplier ρ is coded by a multiplier coder and the multiplier ρ′ obtained by decoding the multiplier ρ is outputted. The process is performed for each frame.
摘要:
A decoder configured to generate decoded audio data (e.g., decoded speech data) and including a postfilter coupled and configured to filter encoded audio data in the frequency domain, methods for frequency domain postfiltering of encoded audio data in a decoder, and methods for decoding encoded audio data in a decoder including by postfiltering encoded audio data in the frequency domain in the decoder. In some embodiments, the decoder is configured to decode input encoded audio without performing any time-to-frequency domain transform on encoded audio data to prepare data for postfiltering. Typically, the postfiltering improves the quality of the decoded audio signal by attenuating spectral valley regions thereof to remove excess quantization noise present in the encoded input audio while preserving formants of the decoded audio signal to avoid introducing unnecessary distortion.