Abstract: Traditional audio encoders may conserve coding bit-rate by encoding fewer than all spectral coefficients, which can produce a blurry low-pass sound in the reconstruction. An audio encoder using wide-sense perceptual similarity improves the quality by encoding a perceptually similar version of the omitted spectral coefficients, represented as a scaled version of already coded spectrum. The omitted spectral coefficients are divided into a number of sub-bands. The sub-bands are encoded as two parameters: a scale factor, which may represent the energy in the band; and a shape parameter, which may represent a shape of the band. The shape parameter may be in the form of a motion vector pointing to a portion of the already coded spectrum, an index to a spectral shape in a fixed code-book, or a random noise vector. The encoding thus efficiently represents a scaled version of a similarly shaped portion of spectrum to be copied at decoding.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: A mixed lossless audio compression has application to a unified lossy and lossless audio compression scheme that combines lossy and lossless audio compression within a same audio signal. The mixed lossless compression codes a transition frame between lossy and lossless coding frames to produce seamless transitions. The mixed lossless coding performs a lapped transform and inverse lapped transform to produce an appropriately windowed and folded pseudo-time domain frame, which can then be losslessly coded. The mixed lossless coding also can be applied for frames that exhibit poor lossy compression performance.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: An audio encoder implements multi-channel coding decision, band truncation, multi-channel rematrixing, and header reduction techniques to improve quality and coding efficiency. In the multi-channel coding decision technique, the audio encoder dynamically selects between joint and independent coding of a multi-channel audio signal via an open-loop decision based upon (a) energy separation between the coding channels, and (b) the disparity between excitation patterns of the separate input channels. In the band truncation technique, the audio encoder performs open-loop band truncation at a cut-off frequency based on a target perceptual quality measure. In multi-channel rematrixing technique, the audio encoder suppresses certain coefficients of a difference channel by scaling according to a scale factor, which is based on current average levels of perceptual quality, current rate control buffer fullness, coding mode, and the amount of channel separation in the source.

Abstract: Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.

Abstract: An audio encoder encodes side information into a compressed audio bitstream containing encoding parameters used by the encoder for one or more encoding techniques, such as a noise-mask-ratio curve used for rate control. A transcoder uses the encoder generated side information to transcode the audio from the original compressed bitstream having an initial bit-rate into a second bitstream having a new bit-rate. Because the side information is derived from the original audio, the transcoder is able to better maintain audio quality of the transcoding. The side information also allows the transcoder to re-encode from an intermediate decoding/encoding stage for faster and lower complexity transcoding.

Abstract: Quantization matrices facilitate digital audio encoding and decoding. An audio encoder generates and compresses quantization matrices; an audio decoder decompresses and applies the quantization matrices. The invention includes several techniques and tools, which can be used in combination or separately. For example, the audio encoder can generate quantization matrices from critical band patterns for blocks of audio data. The encoder can compute the quantization matrices directly from the critical band patterns, which can be computed from the same audio data that is being compressed. The audio encoder/decoder can use different modes for generating/applying quantization matrices depending on the coding channel mode of multi-channel audio data. The audio encoder/decoder can use different compression/decompression modes for the quantization matrices, including a parametric compression/decompression mode.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: The quality and robustness of audio echo cancellation is enhanced by selectively applying glitch recovery processes based on a quality measurement of the relative offset between capture and render audio streams. For example, large and small glitch detection is enabled for low relative offset variance; large glitch detection is enabled in a medium range of relative offset variance; and neither enabled at high variance. Further, a fast glitch recovery process suspends updating the adaptive filter coefficients of the audio echo cancellation while buffers are re-aligned to recover from the glitch, so as to avoid resetting the adaptive filter. When clock drift exists between capture and render audio streams, a multi-step compensation method is applied to improve AEC output quality in case the drifting rate is low; and a resampler is used to compensate the drift in case the drifting rate is high. An anti-clipping process detects clipping of the signals, and also suspends adaptive filter updating during clipping.

Abstract: Stereophonic teleconferencing system embodiments are described which advantageously employ a microphone array at a remote conference site having multiple conferencees to produce a separate output channel from the each microphone in the array. Audio data streams each representing one of the audio output channels from the microphone array are then sent to a local conference site where a local conferencee is in attendance. The voices of the aforementioned remote conferencees are spatialized within a sound-field of the local site using multiple loudspeakers. Generally, this involves receiving the monophonic audio data streams from the remote site, and processing them to generate an audio signal for each loudspeaker. Each of the generated audio signals is then played through its respective loudspeaker to produce a spatial audio sound-field which is audibly perceived by the local conferencee as having the voice of each of the remote conferencees coming from a different location.

Abstract: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of multi-channel audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder performs a pre-processing multi-channel transform on multi-channel audio data, varying the transform so as to control quality. The encoder groups multiple windows from different channels into one or more tiles and outputs tile configuration information, which allows the encoder to isolate transients that appear in a particular channel with small windows, but use large windows in other channels. Using a variety of techniques, the encoder performs flexible multi-channel transforms that effectively take advantage of inter-channel correlation. An audio decoder performs corresponding processing and decoding. In addition, the decoder performs a post-processing multi-channel transform for any of multiple different purposes.

Abstract: An audio decoder provides a combination of decoding components including components implementing base band decoding, spectral peak decoding, frequency extension decoding and channel extension decoding techniques. The audio decoder decodes a compressed bitstream structured by a bitstream syntax scheme to permit the various decoding components to extract the appropriate parameters for their respective decoding technique.

Abstract: An audio encoder and decoder use architectures and techniques that improve the efficiency of quantization (e.g., weighting) and inverse quantization (e.g., inverse weighting) in audio coding and decoding. The described strategies include various techniques and tools, which can be used in combination or independently. For example, an audio encoder quantizes audio data in multiple channels, applying multiple channel-specific quantizer step modifiers, which give the encoder more control over balancing reconstruction quality between channels. The encoder also applies multiple quantization matrices and varies the resolution of the quantization matrices, which allows the encoder to use more resolution if overall quality is good and use less resolution if overall quality is poor. Finally, the encoder compresses one or more quantization matrices using temporal prediction to reduce the bitrate associated with the quantization matrices. An audio decoder performs corresponding inverse processing and decoding.

Abstract: A communications device is presented for providing bi-directional audio communications between a near-end user and a far-end user via a bidirectional communications channel. The communications device includes an adaptive echo canceller receiving a near-end audio signal and a far-end audio signal and providing an echo-canceled near-end audio signal for transmission to the far-end user via the communications channel. The adaptive echo canceller includes a first bank of analysis filters for filtering the near-end audio signal, a second bank of analysis filters for filtering the far-end audio signal, and a bank of synthesis filters for filtering sub-band echo-canceled signals generated within the adaptive echo canceller. The first and second filter banks have a frequency response optimized to reduce echo residual gain.

Abstract: The disclosed architecture employs signal processing techniques to provide audio perception only, or audio perception that matches the visual perception. This also provides spatial audio reproduction for multiparty teleconferencing such that the teleconferencing participants perceive themselves as if they were sitting in the same room. The solution is based on the premise that people perceive sounds as a reconstructed wavefront, and hence, the wavefronts are used to provide the spatial perceptual cues. The differences between the spatial perceptual cues derived from the reconstructed wavefront of sound waves and the ideal wavefront of sound waves form an objective metric for spatial perceptual quality, and provide the means of evaluating the overall system performance. Additionally, compensation filters are employed to improve the spatial perceptual quality of stereophonic systems by optimizing the objective metrics.

Abstract: An audio encoder encodes a combined channel (e.g., a sum channel) for a group of plural physical audio channels. The encoder determines plural parameters for representing individual physical channels of the group as modified versions of the encoded combined channel. The plural parameters comprise ratios of power in each individual channel to power in the combined channel (e.g., a ratio of the power of a right channel to the power of the combined channel, and a ratio of the power of the left channel to the power of the combined channel). The plural parameters can include a complex parameter. The combined channel and the plural parameters facilitate reconstruction at the audio decoder of source channels. An audio decoder performs a forward complex transform on the multi-channel audio data and reconstructs plural channels from the multi-channel audio data. The decoder can maintain second-order statistics for the source channels.

Abstract: Described techniques and tools include techniques and tools for mapping digital media data (e.g., audio, video, still images, and/or text, among others) in a given format to a transport or file container format useful for encoding the data on optical disks such as digital video disks (DVDs). A digital media universal elementary stream can be used to map digital media streams (e.g., an audio stream, video stream or an image) into any arbitrary transport or file container, including optical disk formats, and other transports, such as broadcast streams, wireless transmissions, etc. The information to decode any given frame of the digital media in the stream can be carried in each coded frame. A digital media universal elementary stream includes stream components called chunks. An implementation of a digital media universal elementary stream arranges data for a media stream in frames, the frames having one or more chunks.