DOCSLIB.ORG

User Manual Clipper

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text
User Manual Clipper For a Better Experience User Manual Wireless Receiver for Headphones Clipper Pro www.avantree.com Basic operation Contents 1. AptX Low Latency 2. Product overview & basic operation Button Function Operation Power on/off Hold MFB for 2-3 seconds to turn Clipper Pro ON/OFF, Any questions? Contact: 1. AptX Low Latency 1 Avantree Clipper Pro is a Bluetooth audio receiver incorporating the latest “aptX low you will hear “ Welcome to Avantree” / “ Goodbye” 2. Product overview & basic operation 2 latency” technology, for CD-like quality sound and with virtually no audio delay. From off mode, press & hold for 7 seconds until LED Volume down (Previous track) Volume up (Next track) Enter pairing mode flashes BLUE and RED fast, you will hear “pairing”. Ideal for games, watching TV and movies without lip sync delay. [email protected] 3. What’s in the box? 4 MFB Answer/End a call Press MFB once when a call is incoming or in progress 4. Pairing and connecting 5 Reject a call Press & hold MFB for 2 seconds when a call is incoming 4.1. Connecting to a mobile phone/tablet 5-6 Last number redial Double click MFB To take advantage of the better performance offered by APTX low latency Play/Pause Press once MFB when streaming music 4.2. Connecting to a Bluetooth transmitter 6-7 codec, the other device must also support the same codec (Such as Avantree’s Self-service Support: Volume up Press V+ once 4.3. Connecting to two Bluetooth devices simultaneously 7-8 Saturn Pro, Priva II, Leaf, etc). If the transmitter only supports the normal V+ • More FAQs on support.avantree.com 3.5 mm Audio jack Next track Press and hold • 100+ Step by step video Guide – youtube.com/avantree/playlists 5. How to reset to factory default setting? 8 AptX codec, or the SBC one, these codecs will be used instead. Volume down Press V- once Micro USB port V– 6. LED indications 9 Audio delay by transmitter type - Previous track Press and hold Voice Dial (Siri) Press V+ and V- together once Telephone Support (Mon-Fri): 7. Charging 10 Transmitter support - The audio delay will be LED indicator Multi-function V+& V- Restore to Press and hold V+ and V- together for 7 seconds, the 8. Product Specifications aptX Low Latency codec Around 40ms • USA / CANADA +1 800 232 2078 (EN, PST 9am-5pm) 11 button (MFB) default settings LED flashes BLUE and RED together ( appears PINK) aptX codec Around 180ms • EU / Asia telephone No. refer to http: //www.avantree.com/contact-us Note: Some functions may not work when used with different mobile phones or SBC codec (most standard Bluetooth audio) Around 250ms Microphone other Bluetooth devices. - 1 - - 2 - - 3 - Pair Connect 3 Enjoy 3. What’s in the box? 4. Pairing and connecting Hold MFB 6S, LED Step 4. Connect Clipper Pro to the second Bluetooth device as above. Reconnect: flashes BLUE and Step 5. Once connected, restart Clipper Pro and the first paired device. It will then RED alternately Keep them 4.1. Connecting to a mobile phone/tablet Whenever you turn Clipper Pro on again, it will automatically reconnect to your close & wait automatically reconnect to the two devices simultaneously. Step 1. Ensure Clipper Pro is off. phone. Note: Once disconnected over 10 minutes, Clipper Pro will automatically power Step 2. Set Clipper Pro to PAIRING MODE - Press and hold MFB for 7 seconds itself off. Set transmitter Set Clipper until the LED flashes BLUE and RED alternately and you hear “pairing”. to PAIRING pro to pairing Once connected, the LEDs on Use your existing headphones both devices will slow down. to listen to the TV wirelessly Step 3. Activate Bluetooth on your phone/tablet and select “ Avantree Clipper Pro” , 4.2. Connecting to a Bluetooth transmitter MODE* mode Dual Link LED flash once per 5 seconds once connected, and you hear “connected”. A. B. C. D. **PARING MODE: Discoverable mode for Bluetooth device, normally the LED blinks Pair Connect Enjoy Step 1. Ensure Clipper Pro is off. 3 quickly or flashes between two colors alternately. Step 2. Set Clipper Pro to PAIRING MODE - Press and hold MFB for 7 seconds Settings Bluetooth Bluetooth until the LED flashes BLUE and RED alternately and you hear “pairing”. Devices For a Better Experience Avantree Clipper Pro 4.3. Connecting to two Bluetooth devices simultaneously A. Avantree Clipper Pro Step 3. Set your Bluetooth transmitter to PAIRING MODE**. User Manual Wireless Receiver for Headphones Hold MFB 6S B. Avantree 3.5mm audio splitter *1 Clipper Pro Step 4. Keep both devices close and wait for 5-30 seconds. Once connected, 5. How to reset to factory default setting? Step 1. Connect the first Bluetooth device as above. Then turn off Clipper Pro. C. 3.5mm audio cable *1 LEDs on both devices will slow down and you hear “connected”. www.avantree.com Step 2. Turn off the bluetooth on the first paired device. (If it is a transmitter, turn it off) D. Micro USB charging cable *1 Step 1. Turn on the Clipper Pro E. From off, hold for 6s, LED Activate Bluetooth, search Step 3. Set Clipper Pro to PAIRING MODE again – Press and hold MFB for 6 E. User manual & Quick user guide flashes BLUE and RED and select “Clipper Pro” to Stream music wirelessly Step 2. Press and hold the V+ and V- button for 7 seconds alternately connect. seconds until the LED flashes BLUE and RED alternately. Step 3. The Blue and Red LED will flashes twice then power off at the same time - 4 - - 5 - - 6 - - 7 - - 8 - 6. LED indications 7. Charging 8. Product Specifications 9. Declaration of Conformity This device complies with the essential requirements and other relevant provisions of Directive 1999/5/EC. This Status LED indicator When the LED flashes RED, you should recharge the device. Please remove the • BT version: 4.2 device complies with part 15 of the FCC Rules. Operation is subject to the condition that this device does not cause Power on LED flashes BLUE for 2 seconds micro-USB protector and charge it. It normally takes around 2 hours to fully charge • BT profile: A2DP, AVRCP, HFP, HSP harmful interference (1) this device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. Changes or modifications not Power off LED flashes RED for 2 seconds the battery. The LED will stay RED when charging and will turn off when charging is • Audio codec: aptX Low Latency, aptX, SBC expressly approved by the party responsible for compliance could void the user's authority to operate the equipment. Pairing mode LED flashes BLUE and RED alternately complete. • Operational range: Class II, up to 10 meters NOTE: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to Charging Requirement • Working time: Up to 6 hours Connected LED flashes BLUE once every 10 seconds Part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in You can connect Clipper Pro via the supplied micro USB cable to your own USB • Standby time: 100 hours a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed Disconnected LED flashes BLUE twice every 10 seconds and used in accordance with the instructions, may cause harmful interference to radio communications. However, chargers (wall charger/travel charger/car charger, etc.) with 5V/500mA-1A or PC • Charging time: 2 hours there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful For a Better Experience LED flashes BLUE and RED together twice Restore to default setting USB port. It’s ok to use most mobile phone chargers on the market. • Product weight: approx. 15g interference to radio or television reception, which can be determined by turning the equipment off and on, the user is (appears PINK) encouraged to try to correct the interference by one or more of the following measures: • Product size: 50.5*26*18mm -- Reorient or relocate the receiving antenna. Low battery LED flashes RED once every 30 seconds -- Increase the separation between the equipment and receiver. Charging LED stays RED Note: The talk and standby times may vary when used with different mobile phones -- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected. -- Consult the dealer or an experienced radio/TV technician for help. or other devices and also defendent upon different usage styles, settings and Fully charged Red LED turns off FCC ID: BTHS-AS7 operating environment. Dispose of the packaging and When connected, RED LED flash once per 5s. To maintain compliance with FCC’s RF Exposure guidelines, This equipment should be installed and operated with Low battery minimum distance between 20cm the radiator your body: Use only the supplied antenna. this product in accordance with www.avantree.com When disconnected, RED LED flash once per 2s. the latest provisions. Z-PKMN-AS7L-V5 - 9 - Charge from USB charger - 10 - Charge from PC - 11 -.
Load more

Recommended publications
  • PXC 550 Wireless Headphones
    PXC 550 Wireless headphones Instruction Manual 2 | PXC 550 Contents Contents Important safety instructions ...................................................................................2 The PXC 550 Wireless headphones ...........................................................................4 Package includes ..........................................................................................................6 Product overview .........................................................................................................7 Overview of the headphones .................................................................................... 7 Overview of LED indicators ........................................................................................ 9 Overview of buttons and switches ........................................................................10 Overview of gesture controls ..................................................................................11 Overview of CapTune ................................................................................................12 Getting started ......................................................................................................... 14 Charging basics ..........................................................................................................14 Installing CapTune .....................................................................................................16 Pairing the headphones ...........................................................................................17
    [Show full text]
  • Audio Coding for Digital Broadcasting
    Recommendation ITU-R BS.1196-7 (01/2019) Audio coding for digital broadcasting BS Series Broadcasting service (sound) ii Rec. ITU-R BS.1196-7 Foreword The role of the Radiocommunication Sector is to ensure the rational, equitable, efficient and economical use of the radio- frequency spectrum by all radiocommunication services, including satellite services, and carry out studies without limit of frequency range on the basis of which Recommendations are adopted. The regulatory and policy functions of the Radiocommunication Sector are performed by World and Regional Radiocommunication Conferences and Radiocommunication Assemblies supported by Study Groups. Policy on Intellectual Property Right (IPR) ITU-R policy on IPR is described in the Common Patent Policy for ITU-T/ITU-R/ISO/IEC referenced in Resolution ITU-R 1. Forms to be used for the submission of patent statements and licensing declarations by patent holders are available from http://www.itu.int/ITU-R/go/patents/en where the Guidelines for Implementation of the Common Patent Policy for ITU-T/ITU-R/ISO/IEC and the ITU-R patent information database can also be found. Series of ITU-R Recommendations (Also available online at http://www.itu.int/publ/R-REC/en) Series Title BO Satellite delivery BR Recording for production, archival and play-out; film for television BS Broadcasting service (sound) BT Broadcasting service (television) F Fixed service M Mobile, radiodetermination, amateur and related satellite services P Radiowave propagation RA Radio astronomy RS Remote sensing systems S Fixed-satellite service SA Space applications and meteorology SF Frequency sharing and coordination between fixed-satellite and fixed service systems SM Spectrum management SNG Satellite news gathering TF Time signals and frequency standards emissions V Vocabulary and related subjects Note: This ITU-R Recommendation was approved in English under the procedure detailed in Resolution ITU-R 1.
    [Show full text]
  • (A/V Codecs) REDCODE RAW (.R3D) ARRIRAW
    What is a Codec? Codec is a portmanteau of either "Compressor-Decompressor" or "Coder-Decoder," which describes a device or program capable of performing transformations on a data stream or signal. Codecs encode a stream or signal for transmission, storage or encryption and decode it for viewing or editing. Codecs are often used in videoconferencing and streaming media solutions. A video codec converts analog video signals from a video camera into digital signals for transmission. It then converts the digital signals back to analog for display. An audio codec converts analog audio signals from a microphone into digital signals for transmission. It then converts the digital signals back to analog for playing. The raw encoded form of audio and video data is often called essence, to distinguish it from the metadata information that together make up the information content of the stream and any "wrapper" data that is then added to aid access to or improve the robustness of the stream. Most codecs are lossy, in order to get a reasonably small file size. There are lossless codecs as well, but for most purposes the almost imperceptible increase in quality is not worth the considerable increase in data size. The main exception is if the data will undergo more processing in the future, in which case the repeated lossy encoding would damage the eventual quality too much. Many multimedia data streams need to contain both audio and video data, and often some form of metadata that permits synchronization of the audio and video. Each of these three streams may be handled by different programs, processes, or hardware; but for the multimedia data stream to be useful in stored or transmitted form, they must be encapsulated together in a container format.
    [Show full text]
  • A Multi-Frame PCA-Based Stereo Audio Coding Method
    applied sciences Article A Multi-Frame PCA-Based Stereo Audio Coding Method Jing Wang *, Xiaohan Zhao, Xiang Xie and Jingming Kuang School of Information and Electronics, Beijing Institute of Technology, 100081 Beijing, China; [email protected] (X.Z.); [email protected] (X.X.); [email protected] (J.K.) * Correspondence: [email protected]; Tel.: +86-138-1015-0086 Received: 18 April 2018; Accepted: 9 June 2018; Published: 12 June 2018 Abstract: With the increasing demand for high quality audio, stereo audio coding has become more and more important. In this paper, a multi-frame coding method based on Principal Component Analysis (PCA) is proposed for the compression of audio signals, including both mono and stereo signals. The PCA-based method makes the input audio spectral coefficients into eigenvectors of covariance matrices and reduces coding bitrate by grouping such eigenvectors into fewer number of vectors. The multi-frame joint technique makes the PCA-based method more efficient and feasible. This paper also proposes a quantization method that utilizes Pyramid Vector Quantization (PVQ) to quantize the PCA matrices proposed in this paper with few bits. Parametric coding algorithms are also employed with PCA to ensure the high efficiency of the proposed audio codec. Subjective listening tests with Multiple Stimuli with Hidden Reference and Anchor (MUSHRA) have shown that the proposed PCA-based coding method is efficient at processing stereo audio. Keywords: stereo audio coding; Principal Component Analysis (PCA); multi-frame; Pyramid Vector Quantization (PVQ) 1. Introduction The goal of audio coding is to represent audio in digital form with as few bits as possible while maintaining the intelligibility and quality required for particular applications [1].
    [Show full text]
  • Lossless Compression of Audio Data
    CHAPTER 12 Lossless Compression of Audio Data ROBERT C. MAHER OVERVIEW Lossless data compression of digital audio signals is useful when it is necessary to minimize the storage space or transmission bandwidth of audio data while still maintaining archival quality. Available techniques for lossless audio compression, or lossless audio packing, generally employ an adaptive waveform predictor with a variable-rate entropy coding of the residual, such as Huffman or Golomb-Rice coding. The amount of data compression can vary considerably from one audio waveform to another, but ratios of less than 3 are typical. Several freeware, shareware, and proprietary commercial lossless audio packing programs are available. 12.1 INTRODUCTION The Internet is increasingly being used as a means to deliver audio content to end-users for en­ tertainment, education, and commerce. It is clearly advantageous to minimize the time required to download an audio data file and the storage capacity required to hold it. Moreover, the expec­ tations of end-users with regard to signal quality, number of audio channels, meta-data such as song lyrics, and similar additional features provide incentives to compress the audio data. 12.1.1 Background In the past decade there have been significant breakthroughs in audio data compression using lossy perceptual coding [1]. These techniques lower the bit rate required to represent the signal by establishing perceptual error criteria, meaning that a model of human hearing perception is Copyright 2003. Elsevier Science (USA). 255 AU rights reserved. 256 PART III / APPLICATIONS used to guide the elimination of excess bits that can be either reconstructed (redundancy in the signal) orignored (inaudible components in the signal).
    [Show full text]
  • Improving Opus Low Bit Rate Quality with Neural Speech Synthesis
    Improving Opus Low Bit Rate Quality with Neural Speech Synthesis Jan Skoglund1, Jean-Marc Valin2∗ 1Google, San Francisco, CA, USA 2Amazon, Palo Alto, CA, USA [email protected], [email protected] Abstract learned representation set [11]. A typical WaveNet configura- The voice mode of the Opus audio coder can compress wide- tion requires a very high algorithmic complexity, in the order band speech at bit rates ranging from 6 kb/s to 40 kb/s. How- of hundreds of GFLOPS, along with a high memory usage to ever, Opus is at its core a waveform matching coder, and as the hold the millions of model parameters. Combined with the high rate drops below 10 kb/s, quality degrades quickly. As the rate latency, in the hundreds of milliseconds, this renders WaveNet reduces even further, parametric coders tend to perform better impractical for a real-time implementation. Replacing the di- than waveform coders. In this paper we propose a backward- lated convolutional networks with recurrent networks improved compatible way of improving low bit rate Opus quality by re- memory efficiency in SampleRNN [12], which was shown to be synthesizing speech from the decoded parameters. We compare useful for speech coding in [13]. WaveRNN [14] also demon- two different neural generative models, WaveNet and LPCNet. strated possibilities for synthesizing at lower complexities com- WaveNet is a powerful, high-complexity, and high-latency ar- pared to WaveNet. Even lower complexity and real-time opera- chitecture that is not feasible for a practical system, yet pro- tion was recently reported using LPCNet [15]. vides a best known achievable quality with generative models.
    [Show full text]
  • Cognitive Speech Coding Milos Cernak, Senior Member, IEEE, Afsaneh Asaei, Senior Member, IEEE, Alexandre Hyafil
    1 Cognitive Speech Coding Milos Cernak, Senior Member, IEEE, Afsaneh Asaei, Senior Member, IEEE, Alexandre Hyafil Abstract—Speech coding is a field where compression ear and undergoes a highly complex transformation paradigms have not changed in the last 30 years. The before it is encoded efficiently by spikes at the auditory speech signals are most commonly encoded with com- nerve. This great efficiency in information representation pression methods that have roots in Linear Predictive has inspired speech engineers to incorporate aspects of theory dating back to the early 1940s. This paper tries to cognitive processing in when developing efficient speech bridge this influential theory with recent cognitive studies applicable in speech communication engineering. technologies. This tutorial article reviews the mechanisms of speech Speech coding is a field where research has slowed perception that lead to perceptual speech coding. Then considerably in recent years. This has occurred not it focuses on human speech communication and machine because it has achieved the ultimate in minimizing bit learning, and application of cognitive speech processing in rate for transparent speech quality, but because recent speech compression that presents a paradigm shift from improvements have been small and commercial applica- perceptual (auditory) speech processing towards cognitive tions (e.g., cell phones) have been mostly satisfactory for (auditory plus cortical) speech processing. The objective the general public, and the growth of available bandwidth of this tutorial is to provide an overview of the impact has reduced requirements to compress speech even fur- of cognitive speech processing on speech compression and discuss challenges faced in this interdisciplinary speech ther.
    [Show full text]
  • Codec Is a Portmanteau of Either
    What is a Codec? Codec is a portmanteau of either "Compressor-Decompressor" or "Coder-Decoder," which describes a device or program capable of performing transformations on a data stream or signal. Codecs encode a stream or signal for transmission, storage or encryption and decode it for viewing or editing. Codecs are often used in videoconferencing and streaming media solutions. A video codec converts analog video signals from a video camera into digital signals for transmission. It then converts the digital signals back to analog for display. An audio codec converts analog audio signals from a microphone into digital signals for transmission. It then converts the digital signals back to analog for playing. The raw encoded form of audio and video data is often called essence, to distinguish it from the metadata information that together make up the information content of the stream and any "wrapper" data that is then added to aid access to or improve the robustness of the stream. Most codecs are lossy, in order to get a reasonably small file size. There are lossless codecs as well, but for most purposes the almost imperceptible increase in quality is not worth the considerable increase in data size. The main exception is if the data will undergo more processing in the future, in which case the repeated lossy encoding would damage the eventual quality too much. Many multimedia data streams need to contain both audio and video data, and often some form of metadata that permits synchronization of the audio and video. Each of these three streams may be handled by different programs, processes, or hardware; but for the multimedia data stream to be useful in stored or transmitted form, they must be encapsulated together in a container format.
    [Show full text]
  • Lossy Audio Compression Identification
    2018 26th European Signal Processing Conference (EUSIPCO) Lossy Audio Compression Identification Bongjun Kim Zafar Rafii Northwestern University Gracenote Evanston, USA Emeryville, USA [email protected] zafar.rafi[email protected] Abstract—We propose a system which can estimate from an compression parameters from an audio signal, based on AAC, audio recording that has previously undergone lossy compression was presented in [3]. The first implementation of that work, the parameters used for the encoding, and therefore identify the based on MP3, was then proposed in [4]. The idea was to corresponding lossy coding format. The system analyzes the audio signal and searches for the compression parameters and framing search for the compression parameters and framing conditions conditions which match those used for the encoding. In particular, which match those used for the encoding, by measuring traces we propose a new metric for measuring traces of compression of compression in the audio signal, which typically correspond which is robust to variations in the audio content and a new to time-frequency coefficients quantized to zero. method for combining the estimates from multiple audio blocks The first work to investigate alterations, such as deletion, in- which can refine the results. We evaluated this system with audio excerpts from songs and movies, compressed into various coding sertion, or substitution, in audio signals which have undergone formats, using different bit rates, and captured digitally as well lossy compression, namely MP3, was presented in [5]. The as through analog transfer. Results showed that our system can idea was to measure traces of compression in the signal along identify the correct format in almost all cases, even at high bit time and detect discontinuities in the estimated framing.
    [Show full text]
  • Speech Compression
    information Review Speech Compression Jerry D. Gibson Department of Electrical and Computer Engineering, University of California, Santa Barbara, CA 93118, USA; [email protected]; Tel.: +1-805-893-6187 Academic Editor: Khalid Sayood Received: 22 April 2016; Accepted: 30 May 2016; Published: 3 June 2016 Abstract: Speech compression is a key technology underlying digital cellular communications, VoIP, voicemail, and voice response systems. We trace the evolution of speech coding based on the linear prediction model, highlight the key milestones in speech coding, and outline the structures of the most important speech coding standards. Current challenges, future research directions, fundamental limits on performance, and the critical open problem of speech coding for emergency first responders are all discussed. Keywords: speech coding; voice coding; speech coding standards; speech coding performance; linear prediction of speech 1. Introduction Speech coding is a critical technology for digital cellular communications, voice over Internet protocol (VoIP), voice response applications, and videoconferencing systems. In this paper, we present an abridged history of speech compression, a development of the dominant speech compression techniques, and a discussion of selected speech coding standards and their performance. We also discuss the future evolution of speech compression and speech compression research. We specifically develop the connection between rate distortion theory and speech compression, including rate distortion bounds for speech codecs. We use the terms speech compression, speech coding, and voice coding interchangeably in this paper. The voice signal contains not only what is said but also the vocal and aural characteristics of the speaker. As a consequence, it is usually desired to reproduce the voice signal, since we are interested in not only knowing what was said, but also in being able to identify the speaker.
    [Show full text]
  • Video Source File Specifications
    Video Source File Specifications Limelight recommends the following specifications for all video source files. Adherence to these specifications will result in optimal playback quality and efficient uploading to your account. Edvance360 Best Practices Videos should be under 1 Gig for best results (download speed and mobile devices), but our limit per file is 2 Gig for Lessons MP4 is the optimum format for uploading videos Compress the video to resolution of 1024 x 768 Limelight does compress the video, but it's best if it's done on the original file A resolution is 1080p or less is recommended Recommended frame rate is 30fps Note: The maximum file size for Introduction Videos in Courses is 50MB. This is located in Courses > Settings > Details > Introduction Video. Ideal Source File In general, a source file that represents the following will produce the best results: MP4 file (H.264/ACC-LC) Fast Start (MOOV atom at the front of file) Progressive scan (no interlacing) Frame rate of 24 (23.98), 25, or 30 (29.97) fps A Bitrate between 5,000 - 8,000 Kbps 720p resolution Detailed Recommendations The table below provides detailed recommendations (CODECs, containers, Bitrates, resolutions, etc.) for all video source material uploaded to a Limelight Account: Source File Element Recommendations Video CODEC Recommended CODEC: H.264 Accepted but not Recommended: MPEG-1, MPEG-2, MPEG-4, VP6, VP5, H.263, Windows Media Video 7 (WMV1), Windows Media Video 8 (WMV2), Windows Media Video 9 (WMV3) Audio CODEC Recommended CODEC: AAC-LC Accepted but not Recommended: MP3, MP2, WMA, WMA Pro, PCM, WAV Container MP4 Source File Element Recommendations Fast-Start Make sure your source file is created with the 'MOOV atom' at the front of the file.
    [Show full text]
  • An Audio Codec for Multiple Generations Compression Without Loss of Perceptual Quality
    Frank Kurth An Audio Codec for Multiple Generations Compression without Loss of Perceptual Quality AN AUDIO CODEC FOR MULTIPLE GENERATIONS COMPRESSION WITHOUT LOSS OF PERCEPTUAL QUALITY FRANK KURTH Department of Computer Science V, University of Bonn, Bonn, Germany [email protected] We describe a generic audio codec allowing for multiple, i.e., cascaded, lossy compression without loss of perceptual quality as compared to the first generation of compressed audio. For this sake we transfer encoding information to all subsequent codecs in a cascade. The supplemental information is embedded in the decoded audio signal without causing degradations. The new method is applicable to a wide range of current audio codecs as documented by our MPEG-1 implementation. INTRODUCTION 1. AGEING EFFECTS IN CASCADED CODING Low bit rate high quality coding is used in a wide range of Input Output nowadays audio applications such as digital audio broad- Block- or Subband- Quantization Requantization Inverse casting or network conferencing. Although the decoded Transform Transform versions of the compressed data maintain very high sound quality, multiple- or tandem-coding may result in accu- Spectral Analysis/ mulated coding errors resulting from lossy data reduction Psychoacoustic Model Encoder Decoder schemes. Such multiple or tandem coding, leading to the notion of a signal’s generations, may be decribed as fol- lows. Assuming a coder operation and a corresponding Figure 1: Simplified scheme of psychoacoustic codec. ¢ decoder operation ¡ we shall call, for a given signal , ¦¥ § A general scheme of a psychoacoustic audio codec is ¢ £ ¤ ¡ ¢ ¡ the first generation and, for an integer , given in Fig. 1. In this figure we have omitted optional ¢ the £ -th generation of .
    [Show full text]