Showing papers on "Adaptive Multi-Rate audio codec published in 2005"

AMR-WB+: a new audio coding standard for 3rd generation mobile audio services

Abstract: Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase, 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides a unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

Lossless multi-channel audio codec

Abstract: A lossless audio codec segments audio data within each frame to improve compression performance subject to a constraint that each segment must be fully decodable and less than a maximum size. For each frame, the codec selects the segment duration and coding parameters, e.g., a particular entropy coder and its parameters for each segment, that minimizes the encoded payload for the entire frame subject to the constraints. Distinct sets of coding parameters may be selected for each channel or a global set of coding parameters may be selected for all channels. Compression performance may be further enhanced by forming M/2 decorrelation channels for M-channel audio. The triplet of channels (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated, decorrelated) provides two possible pair combinations (basis, correlated) and basis, decorrelated) that can be considered during the segmentation and entropy coding optimization to further improve compression performance.

Spatial Audio Coding: Next-generation efficient and compatible coding of multi-channel audio

Abstract: Reference LCAV-CONF-2005-033 URL: www.aes.org Record created on 2005-10-07, modified on 2017-05-12

Embedded base layer codec for 3D sub-band coding

Abstract: Techniques and tools are described for scalable video coding and decoding. For example, a 3D sub-band video encoder includes an embedded base layer codec as well as temporal sub-band transforms and spatial sub-band transforms. The placement of the base layer codec among the sub-band transforms and the role of the base layer codec in scalable video coding vary depending on implementation. In general, the base layer codec provides efficient compression at low bit rates and produces a base layer compressed video bit stream compatible with existing decoders. At the same time, the 3D sub-band video encoder provides spatial and temporal scalability options at higher bit rates, refining the base layer video. A corresponding 3D sub-band video decoder includes an embedded base layer decoder.

Scalable lossless audio codec and authoring tool

Abstract: An audio codec losslessly encodes audio data into a sequence of analysis windows in a scalable bitstream. This is suitably done by separating the audio data into MSB and LSB portions and encoding each with a different lossless algorithm. An authoring tool compares the buffered payload to an allowed payload for each window and selectively scales the losslessly encoded audio data, suitably the LSB portion, in the non-conforming windows to reduce the encoded payload, hence buffered payload. This approach satisfies the media bit rate and buffer capacity constraints without having to filter the original audio data, reencode or otherwise disrupt the lossless bitstream.

A multi-channel audio compression method with virtual source location information for MPEG-4 SAC

Abstract: Binaural cue coding (BCC) was introduced as an efficient representation method for MPEG-4 SAC (spatial audio coding). However, in a low bit-rate environment, the spectrum of BCC output signals degrades with respect to the perceptual level. The proposed system in this paper estimates VSLI (virtual source location information) as the side information. The VSLI is the angle representation of spatial images between channels on playback layout. The subjective assessment results show that the proposed method provides better audio quality than the BCC method for encoding multi-channel signals.

Interoperability for wireless user devices with different speech processing formats

Abstract: Interoperability is achieved between wireless user communication devices that have different speech processing formats and/or attributes. A first wireless user communication device includes a primary speech codec that encodes a first speech message using a first speech encoding format. The encoded speech is then sent to a second wireless user communications device that includes a primary speech codec supporting a second speech encoding format. The first user device receives from the second user device a second speech message encoded using the second speech encoding format. The second speech message is then decoded by the first user device using a second speech decoder supporting decoding of the second speech encoding format. But the first communication device does not support speech encoding using the second speech encoding format--regardless of whether the first communication device includes or does not includes an encoder for encoding speech using the first speech encoding format.

High-Fidelity Multichannel Audio Coding

Abstract: Preface Audio is one of the fundamental elements in multimedia signals. Audio signal processing has attracted attention from researchers and engineers for several decades. By exploiting unique features of audio signals and common features of all multi-media signals, researchers and engineers have been able to develop more efficient technologies to compress audio data. Although books on digital audio have been available some time, the subject of multichannel audio coding techniques has not yet been addressed in great detail. With many years of teaching and research in the field of digital audio signal processing and digital audio compression, we see a need for an advanced audio coding book that covers recent developments in this field. When we started this book project, we had a smaller scope. Our objective was to present several innovative compression techniques for multichannel audio sources and publish it as a research monograph. However, after the first draft, we received valuable comments from our colleagues and anonymous reviewers. With their encouragement, we decided to extend the coverage of the book by including more background material to make it a senior undergraduate or a graduate level textbook on advanced audio coding techniques. Special thanks also go to Dr. Hongmei Ai for her valuable discussions and suggestions when we developed and tested our new audio coding algorithms. This book includes three parts. The first part covers the basic topics on audio compression, such as quantization, entropy coding, psychoacoustic models, and sound quality assessment. The second part of the book highlights the current most prevalent low-bit-rate high-performance audio coding standard—MPEG-4 Audio. More emphasis is given to the audio standards that are capable of supporting multichannel signals, that is, MPEG Advanced Audio Coding (AAC), including the original MPEG-2 AAC specification, additional MPEG-4 toolsets, and the most recent aacPlus standard. The third part of this book introduces several innovative multichannel audio coding methods, which can further improve the coding performance and expand the available functionalities of MPEG AAC. This section is more suitable for graduate students and researchers.

Transcoding apparatus and method for distributed multimedia transmission network provided with transcoder

Abstract: Disclosed is a transcoding method and apparatus for a mobile communication system provided with a transcoder. The transcoding method includes receiving a request for media content and codec information required for the requested media content, receiving the requested media content from a media content server and determining whether the received codec information exists, transformatting a format of the received media content by a codec of the codec information if the received codec information exists, searching the codec list and installing the codec information and transformatting the format of the received media content using the installed codec, determining whether a transmission bandwidth through which the transformatted media content are transmitted is smaller than a bandwidth of the transformatting codec, transrating the format of the media content by a codec suitable for the transmission bandwidth if the transmission bandwidth is smaller than the bandwidth of the transformatting codec, and transmitting the transrated media content.

Improving coding efficiency for MPEG-4 Audio Scalable Lossless coding

Abstract: The recently introduced MPEG standard for lossless audio coding, MPEG-4 Audio Scalable to Lossless (SLS) coding technology, provides a universal audio format that integrates the functionalities of lossy audio coding, lossless audio coding and fine granular scalable audio coding in a single framework. We propose two coding methods that improve the coding efficiency of SLS, namely, a context-based arithmetic code (CBAC) method and a low energy mode code method. These two coding methods work harmonically with the current SLS framework and preserve all its desirable features, such as fine granular scalability, while successfully improving its lossless compression ratio performance.

Compensation of transient effects in transform coding

Abstract: The present invention provides a method for compensating transient effects in transform coding and decoding of a combined speech and audio in electronic devices by using a transform based time-frequency domain codec. The method can combine, e.g., a CELP (code excited linear prediction) type speech codec and a transform type audio codec. The invention describes a compensation method to handle the transient (e.g., from the CELP coding to the transform coding) in transform coding when the number of quantized transform coding coefficients is lower than in the output of the transform.

Artificial bandwidth expansion method to improve intelligibility and quality of AMR-coded narrowband speech

Abstract: Speech quality suffers from the limited bandwidth of cellular telephone systems, making it sound muffled. In addition, intelligibility is degraded due to missing higher frequency components. The proposed enhancement system is designed to improve both intelligibility and quality of narrowband speech by expanding the bandwidth and creating new spectral components to high frequencies in the receiving end of the transmission link. The algorithm can be used together with conventional narrowband speech codecs and it is designed to be robust in different noise conditions. In addition, the computational load of the algorithm is reasonable.

An assessment of the audio codec performance in voice over WLAN (VoWLAN) systems

Abstract: In this paper we present results of experimental investigation into the performance of three audio codecs (ITU-T G.711, G.723.1, and G.729A) under varying load conditions on a voice over WLAN system utilizing the IEEE 802.11b wireless LAN standard. The analysis is based upon a new technique for estimating user satisfaction of speech quality calculated from packet delay and packet loss/late measurements. We also demonstrate the importance of the de-jitter buffer playout scheme for insuring speech quality. From our results we conclude that the use of the G.711 audio codec in conjunction with the new adaptive playout scheme gives the highest user satisfaction of the voice over WLAN schemes considered.

Method and system for terminal codec setup of multimedia ring back tone service

Abstract: Disclosed is a method for terminal codec setup for a multimedia Ring Back Tone (RBT) service, which allows RBT sound sources previously set in a sound source-providing server by a called subscriber to be reproduced to an originating terminal by means of a Home Location Register (HLR) and the sound source-providing server for storing the RBT sound sources when a calling subscriber telephones the called subscriber, the HLR storing profile information including whether the subscriber has joined the RBT service. The method includes the steps of : (a) receiving a first codec setup message including information (multimedia codec information) regarding a multimedia codec from the called subscriber, after transmitting an ISDN User Part (ISUP) call connection request message including the multimedia codec information to the called subscriber; (b) when the first codec setup message is received, transmitting a second codec setup message for requesting setup of the multimedia codec to a call-side Base Transceiver Station (BTS) , thereby controlling a call-side vocoder located in a call-side Base Station Controller (BSC) to set the multimedia codec; (c) when the first codec setup message is received, transmitting a third codec setup message for requesting setup of the multimedia codec to the originating terminal, thereby controlling the originating terminal to set the multimedia codec; and (d) receiving an RBT sound source selected using the multimedia codec information from the sound source- providing server and transmitting the RBT sound source to the originating terminal.

Interactive Codec Selection

Abstract: A codec selection process is independent of a codec source. A codec table may be used that includes indexed codec references related to codecs stored in a codec source. The codec table and the codec source may be modified without affecting a codec selection process. This feature of an exemplary implementation makes it fairly straightforward to add, change, or otherwise modify codecs stored in the codec source and referenced in the codec table without having to change the codec selection process.

Adaptive thresholds for AMR codec mode selection

Abstract: The speech codecs from the adaptive multi-rate (AMR) codec family enable provisioning of excellent speech quality, at the same time providing a way forward towards state-of-the-art, spectrally efficient, high capacity cellular networks. One straightforward way to characterize the benefit of AMR speech codecs is that the robustness to interference and noise in radio networks is increased and that this advantage over other, nonadaptive, speech codecs can be capitalized on in several different ways, e.g., by enhancing speech quality or improving system capacity. In this paper, improved mode adaptation, where codec mode switching thresholds are adaptive to radio conditions, is discussed. Example simulations show that an adaptive thresholds algorithm applied to GSM can significantly improve objective speech quality. Corresponding improvements were also found in informal listening tests.

A speech coder post-processor controlled by side-information

Abstract: Speech coders provide high speech quality at low rates. However they perform poorly when encoding non-speech signals. This paper proposes a new enhancement algorithm requiring minimum side information to reduce the effect of this shortcoming. The enhancement algorithm consists of post-processing the speech decoder output in the spectral domain. Specifically, some frequency components are reduced or forced to zero when the corresponding frequency content is poorly described by the speech coder. The choice of modifying spectral components is determined at the encoder, thus requiring us to transmit the decision information. Experiments combining the AMR-WB speech codec and the proposed audio enhancement show that the quality for music signals is improved significantly while not affecting the quality for speech inputs.

Performance Analysis for MPEG-4 Video Codec Based on On-Chip Network

Abstract: In this paper, we present a performance analysis for an MPEG-4 video codec based on the on-chip network communication architecture. The existing on-chip buses of system-on-a-chip (SoC) have some limitation on data traffic bandwidth since a large number of silicon IPs share the bus. An on-chip network is introduced to solve the problem of on-chip buses, in which the concept of a computer network is applied to the communication architecture of SoC. We compared the performance of the MPEG-4 video codec based on the on-chip network and Advanced Micro-controller Bus Architecture (AMBA) on-chip bus. Experimental results show that the performance of the MPEG-4 video codec based on the on-chip network is improved over 50% compared to the design based on a multi-layer AMBA bus.

A H.264 compliant stereoscopic video codec

Abstract: Due to the provision of a more natural representation of a scene in the form of left and right eye views, a stereoscopic imaging system provides a more effective method for image/video display. Unfortunately the vast amount of information that needs to be transmitted/stored to represent a stereo image pair/video sequence, has so far hindered its use in commercial applications. However, by properly exploiting the spatial, temporal and binocular redundancy, a stereo image pair or a sequence could be compressed and transmitted through a single monocular channel's bandwidth without unduly sacrificing the perceived stereoscopic image quality. In this paper, we present a new technique for coding stereo video sequences based on H.264 video codec. The proposed codec exploits disparity and worldline correlation in addition to the advance compression techniques inherited by the H.264 standard to achieve a higher video quality especially in the low bit rates. We compare the performance of the proposed CODEC with a DCT-based, modified MPEG-2 stereo video CODEC and ZTE based stereo video CODEC. We show that the proposed CODEC outperforms the benchmark CODECs in coding both main and auxiliary streams by up to 9.0 dB PSNR gain

Proposal on objective speech quality assessment for wideband IP telephony

Abstract: We investigated the performance of the objective speech quality assessment method called "wideband PESQ", which has been proposed to ITU-T for quality assessment of wideband speech, for quality ranging from narrowband to wideband speech with packet-loss degradation. The experimental results show that although the relationship between wideband PESQ and subjective quality depends on the codec for wideband speech, wideband PESQ can be used quite well as an objective quality measurement for speech with packet-loss degradation as long as the same codec is used. This codec dependence, which is especially observed when evaluating wideband codecs, makes it difficult to consistently estimate the quality of speech coded by various kinds of codecs. To avoid this difficulty, we propose correcting objective scores based on a function prepared for each codec. This correcting function is predetermined based on the results of a subjective quality experiment. We show the validity of the proposed method in evaluating the speech quality of various codecs with packet-loss degradation.

Two-channel predictive multiple description coding

Abstract: This paper presents a multiple description (MD) video codec based on the principles side-information coding. In particular, we highlight certain key components of the codec design that contribute significantly to the rate-distortion performance of the proposed codec. These include the use of randomized permutations of the quantization codebook in conjunction with binary LDPC codes for partitioning the available bit-rate among the coefficient bit-planes. Another key component of the proposed codec is the use of pdf estimation for improved decoder reconstruction. Lastly, we use a bank of sequential LDPC decoders to efficiently decode the transmitted coset information. Empirical evaluation demonstrates the superior performance of the proposed codec for the communication of encoded video over packet erasure channels.

Encoding and error correction system for enhanced performance of legacy communications networks

Abstract: An encoding and error correction system and method employs an AMR codec (18) by stripping header data from a plurality of legacy system frames (10) having header and traffic channel (TCH) data blocks. Speech data is then encoded using the AMR to create bits for a data block substantially the same as contained in the plurality of frames. The stripped header data is encoded as a long frame header using a fixed convolution coder (24). The speech data is then convolutionally encoded and the long frame header and encoded speech data are combined as a long frame (32). The long frame is then deconstructed into a plurality of equal segments (106, 110) and the segments are transmitted as TCH data in the legacy system frame format.

Increasing the robustness of CELP-based coders by constrained optimization

Abstract: The adaptive codebook used in CELP-like speech coders is extremely effective on voiced signals. Unfortunately, it is also the main source of error propagation at the decoder when a frame is lost. In this paper, we study several ways of limiting the energy contribution of the adaptive codebook to the synthesized speech signal. We show that a constrained search of the adaptive and innovative codebooks significantly improves the recovery time of the decoder after a lost frame, at the cost of only minor quality degradation in a clear channel. When applied to a standard codec such as the AMR-WB, this constraint only affects the encoder, and the modified codec remains fully interoperable with the standard codec.

Method for codec mode adaptation of adaptive multi-rate codec regarding speech quality

Abstract: Method of codec mode adaptation of a speech codec in dependency of the prevailing channel condition for transmission of speech frames in a telecommunication system comprising the steps of determining a bit error rate (BER) from a estimated carrier to interferer ratio (C/I) per speech burst, generating a frame BER value of speech frame from a plurality of consecutive bursts and determining a critical BER level for a plurality of speech frames by a maximum operation of the frame BER values of the plurality of speech frames.

RTP Payload Format for BroadVoice Speech Codecs

Abstract: This document describes the RTP payload format for the BroadVoice(R) narrowband and wideband speech codecs. The narrowband codec, called BroadVoice16, or BV16, has been selected by CableLabs as a mandatory codec in PacketCable 1.5 and has a CableLabs specification. The document also provides specifications for the use of BroadVoice with MIME and the Session Description Protocol (SDP). [STANDARDS-TRACK]

Codec switching device

Abstract: PROBLEM TO BE SOLVED: To make it impossible that a codec which has developed trouble or the like to become unable to transmit is judged as most suitable for transmission. SOLUTION: The codec switching device 10 includes a plurality of codecs a1 to an and a controller 13. The controller 13 searches its own codec information s1 containing a transmission band of the plurality of the codecs a1 to an and other codec information s2 containing a transmission band of a plurality of codecs b1 to bn of a transmission party as well as network information s3 containing a size of a free bandwidth of a network 30. Based on the searched results, a codec utilized at the time of transmission is selected from among the plurality of codecs a1 to an. On this occasion, a codec with the most large-sized transmission bandwidth is selected from among free bandwidths of the network 30. These operations are implemented also during transmission. COPYRIGHT: (C)2007,JPO&INPIT

AVS Generic Audio Coding

Abstract: AVS Audio Coding Standard is the first standard for Hi-Fi audio in China. The framework of AVS Audio was introduced. Many key technologies are described in details, including long/short window switch decision based on energy and unpredictability, integer MDCT for lossless time-frequency transform, square polar stereo coding, and context-dependent bitplane coding for scalable entropy coding. The informal subject test result is given between AVS audio codec and several dominating audio codecs. It is shown that AVS audio codec is enough for Hi-Fi audio applications.

Signal modification method for variable bit rate wide-band speech coding

Abstract: This paper introduces a novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates. The method is enabled only in stable voiced speech frames, and preserves the original time scale at the end of each frame. This feature helps to avoid artifacts and simplifies an encoder implementation. The signal modification includes a classification algorithm as an integral part. The classification algorithm detects the frames most suitable for signal modification and low bit rate coding, and can be employed in a rate selection module of variable bit rate (VBR) codecs. In this paper, the signal modification method is applied in an experimental VBR wide-band speech codec derived from the 3GPP adaptive multirate wideband (AMR-WB) standard (ITU-T Recommendation G.722.2). The codec fulfills the system requirements of IS-95/CDMA2000 Rate Set II, operating at source coding bit rates 12.65, 6.2, and 1.0 kb/s. The signal modification is used in the 6.2 kb/s mode dedicated for voiced speech frames. Listening test results demonstrate the good performance of the proposed method. The signal modification method is used in the Nokia/VoiceAge codec that was declared in April 2003 as the winner of the selection phase in the 3GPP2 CDMA2000 wide-band speech codec standardization.

Audio codec system and audio signal encoding method using the same

Abstract: An audio codec system and an encoding method using the same are provided. According to the method, encoding and decoding processes are repeatedly performed so as to determine optimized coding parameters when analog audio signals being inputted are encoded. The processes of encoding and decoding inputted analog audio signals using initial coding parameters, and computing new parameters using a differential computed during the encoding process are repeatedly performed.