The speech encoding apparatus (220) has a first layer encoding section (2201), a first layer decoding section (2202), a delay section (2203), a subtracting section (104), a frequency domain transforming section (101), a second layer encoding section (105) and a multiplexing section (106).

Encoding device, decoding device, and method thereof

The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

Overview of the EVS codec architecture

In accordance with an embodiment, a method of decoding an encoded audio bitstream at a decoder includes receiving the audio bitstream, decoding a low band bitstream of the audio bitstream to get low band coefficients in a frequency domain, and copying a plurality of the low band coefficients to a high frequency band location to generate high band coefficients. The method further includes processing the high band coefficients to form processed high band coefficients. Processing includes modifying an energy envelope of the high band coefficients by multiplying modification gains to flatten or smooth the high band coefficients, and applying a received spectral envelope decoded from the received audio bitstream to the high band coefficients. The low band coefficients and the processed high band coefficients are then inverse-transformed to the time domain to obtain a time domain output signal.

Spectrum flatness control for bandwidth extension

A method for decoding an audio signal includes: obtaining a lower-band signal component of an audio signal corresponding to a received code stream when the audio signal switches from a first bandwidth to a second bandwidth which is narrower than the first bandwidth; extending the lower-band signal component to obtain higher-band information; performing a time-varying fadeout process on the higher-band information to obtain a processed higher-band signal component; and synthesizing the processed higher-band signal component and the obtained lower-band signal component. With the methods provided in the embodiments of the invention, when an audio signal has a switch from broadband to narrowband, a series of processes such as bandwidth detection, artificial band extension, time-varying fadeout process, and bandwidth synthesis, may be performed to make the switch to have a smooth transition from a broadband signal to a narrowband signal so that a comfortable listening experience may be achieved.

Method and apparatus for audio decoding

Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.

Coding of transitional speech frames for low-bit-rate applications

A scalable decoder capable of preventing degradation of the quality of the decoded signal in a disappeared data interpolation in band scalable coding. A core layer decoding section (101) acquires a core layer decoded signal and narrow band spectrum information by decoding. A narrow band spectrum slope computing section (103) computes the slope of an attenuation line of a narrow band spectrum from the narrow band spectrum information. An extended layer disappearance detection section (104) detects whether extended layer coded data has disappeared or not. An extended layer decoding section (105) normally decodes the extended layer coded data. If the extended layer disappears, a parameter required for decoding is interpolated and synthesizes an interpolation decoded signal by the interpolated parameter. The gain of the interpolated data is controlled according to the results of the computation, by the narrow band spectrum slope computing section (103).

Scalable decoder and disappeared data interpolating method

Provided is an audio decoding device capable of suppressing an information amount for a lost frame compensation process and encoding efficiency. In this device, a decoded sound source generation unit ( 203 ) generates a lost frame decoded sound source signal; a pitch pulse information decoding unit ( 204 ) decodes the pitch pulse position information and the pitch pulse amplitude information; a pitch pulse waveform learning unit ( 205 ) learns a pitch pulse learning waveform in the past frame in advance from the lost frame; a convolution unit ( 206 ) amplitude-adjusts the pitch pulse learning waveform according to the pitch pulse amplitude information, and convolutes the pitch pulse waveform into a time axis which has been amplitude-adjusted according to the pitch pulse position information; a sound source signal correction unit ( 207 ) adds or replaces the pitch pulse waveform convoluted into the time axis to the lost frame decoded sound source signal.

Speech decoding apparatus and speech encoding apparatus

A scalable decoder capable of avoiding deterioration in subjective quality of a listener. The scalable decoder for decoding core layer encoding data and extension layer encoding data including an extension layer gain coefficient, wherein a voice analysis section detects variation in power of a core layer decoding voice signal being obtained from the core layer encoding data, a gain attenuation rate calculating section (140) sets the attenuation intensity variable depending on variation in power, and a gain attenuation section (143) attenuates the extension layer gain coefficient in a second period preceding a first period according to a set attenuation intensity when extension layer encoding data in the first period is missing, thus interpolating the extension layer gain coefficient in the first period.

Scalable decoder and scalable decoding method

Disclosed are an audio encoding device and an audio decoding device which reduce degradation of subjective quality of a decoding signal caused by power mismatch of a decoding signal which is generated by a concealing process upon disappearance of a frame. When a frame is lost, a past encoding parameter is used to obtain a concealed LPC of the current frame and a concealed sound source parameter. A normal CELP decoding is performed from the obtained concealed sound source parameter. Correction is performed by using a conceal parameter on the obtained concealed LPC and the concealed sound source signal. The power of the corrected concealed sound source signal is adjusted to match a reference sound source power. A filter gain of the synthesis filter is adjusted so as to adjust the power of a decoded sound signal to the power of a decoded sound signal during an error-free state. Moreover, a synthesis filter gain adjusting coefficient is calculated by using an estimated normalized residual power so that a filter gain of a synthesis filter formed by using a concealed LPC is a filter gain during an error-free state.

Audio encoding device and audio decoding device

This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced into the low bit-rate MDCT coder. Further, small symbol switched Huffman coding is exploited for reducing the bits consumption for quantizing band energies of the spectrum. Finally, the performance of the coder is illustrated with some listening test results.

Low bit rate high-quality MDCT audio coding of the 3GPP EVS standard

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