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Total Recorder
Total Recorder is a digital audio editor software from High Criteria, Inc. which is able to record digital sound either directly through the computer's soundcard, or virtually through its software driver. The virtual driver provides the advantage of recording audio reproduced by an external program (including Internet broadcasts) directly in digital format, i.e. without digital-analog-digital conversions leading to loss of quality, and even in those cases when a computer soundcard has no loop-back line (e.g. Stereo Mix, "What you hear"). Total Recorder is a shareware program. The evaluation version of Total Recorder is a fully functional version of the program, with the exception that an audible noise will be inserted about every 60 seconds. Editions Total Recorder is offered in four editions: Standard, Professional, Developer, and VideoPro (beginning with version 8.0 of Total Recorder). Different functionalities are available in different editions. Supported formats Vide ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Audio Editor
Audio editing software is any software or computer program which allows editing and generating audio data. Audio editing software can be implemented completely or partly as a library, as a computer application, as a web application, or as a loadable kernel module. Wave editors are digital audio editors. There are many sources of software available to perform this function. Most can edit music, apply effects and filters, and adjust stereo channels. A digital audio workstation (DAW) is software-based and typically comprises multiple software suite components, all accessible through a unified graphical user interface. DAWs are used for recording or producing music, sound effects and more. Music software capabilities Audio editing software typically offer the following features: *The ability to import and export various audio file formats for editing. *Record audio from one or more inputs and store recordings in the computer's memory as digital audio. *Edit the start ti ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Advanced Systems Format
Advanced Systems Format (formerly Advanced Streaming Format, Active Streaming Format) is Microsoft's proprietary digital audio/digital video container format, especially meant for streaming media Streaming media refers to multimedia delivered through a Computer network, network for playback using a Media player (other), media player. Media is transferred in a ''stream'' of Network packet, packets from a Server (computing), .... ASF is part of the Media Foundation framework. Overview and features ASF is based on serialized '' objects'' which are essentially byte sequences identified by a Globally Unique Identifier, GUID marker. The format does not specify how (i.e. with which codec) the video or audio should be encoded; it just specifies the structure of the video/audio stream. This is similar to the function performed by the QuickTime File Format, Audio Video Interleave, AVI, or Ogg formats. One of the objectives of ASF was to support playback from Streamin ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Timeshifting
In broadcasting, time shifting is the recording of programming to a storage medium to be viewed or listened to after the live broadcasting. Typically, this refers to TV programming but it can also refer to radio shows via podcasts. In recent years, the advent of the digital video recorder (DVR) has made time shifting easier, by using an electronic program guide (EPG) and recording shows onto a hard disk. Some DVRs have other possible time-shifting methods, such as being able to start watching the recorded show from the beginning even if the recording is not yet complete. In the past, time shifting was done with a video cassette recorder (VCR) and its timer function, in which the VCR tunes into the appropriate station and records the show onto video tape. Certain broadcasters transmit timeshifted versions of their channels, usually carrying programming from one hour in the past, to enable those without recording abilities to resolve conflicts and those with recording abilities mor ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Radio Music Ripping
Music radio ripping is the process of digitally extracting and saving songs played on the radio. Ripping is more than simply recording the audio. The key aspect of ripping is disambiguation. Ripped songs can be split into separate tracks or files, and may be tagged. Different software, techniques and cloud services make it possible to extract the songs played on the radio and digitally save them on separate audio tracks. Available techniques make it possible to rip the music from Internet radio broadcasts, satellite radio broadcasts and FM radio broadcasts. Internet radio Popular audio formats for Internet radio include AAC, AAC+, and MP3. Many AM/FM stations simulcast online use the more efficient AAC format while Internet-only streams more typically use MP3. The Shoutcast database featured 34,281 online radio stations covering almost every conceivable music genre. Cloud recording services such as DAR.fm, Quick Record can record from all of these formats. Most PC based Intern ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sound Processing
Audio signal processing is a subfield of signal processing that is concerned with the electronic manipulation of audio signals. Audio signals are electronic representations of sound waves—longitudinal waves which travel through air, consisting of compressions and rarefactions. The energy contained in audio signals or sound power level is typically measured in decibels. As audio signals may be represented in either digital or analog format, processing may occur in either domain. Analog processors operate directly on the electrical signal, while digital processors operate mathematically on its digital representation. History The motivation for audio signal processing began at the beginning of the 20th century with inventions like the telephone, phonograph, and radio that allowed for the transmission and storage of audio signals. Audio processing was necessary for early radio broadcasting, as there were many problems with studio-to-transmitter links. The theory of signal p ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Advanced Audio Coding
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was developed by Dolby, AT&T, Fraunhofer and Sony, originally as part of the MPEG-2 specification but later improved under MPEG-4.ISO (2006ISO/IEC 13818-7:2006 – Information technology — Generic coding of moving pictures and associated audio information — Part 7: Advanced Audio Coding (AAC), Retrieved on 2009-08-06ISO (2006, Retrieved on 2009-08-06 AAC was designed to be the successor of the MP3 format (MPEG-2 Audio Layer III) and generally achieves higher sound quality than MP3 at the same bit rate. AAC encoded audio files are typically packaged in an MP4 container most commonly using the filename extension .m4a. The basic profile of AAC (both MPEG-4 and MPEG-2) is called AAC-LC (''Low Complexity''). It is widely supported in the industry and has been adopted as the default or standard audio format on products including Apple's iTunes Store, Nintendo's Wii, DSi and ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Monkey's Audio
Monkey's Audio is an algorithm and file format for lossless audio data compression. Lossless data compression does not discard data during the process of encoding, unlike lossy compression methods such as Advanced Audio Coding, MP3, Vorbis, and Opus. Similar to other lossless audio codecs, files encoded to Monkey's Audio are typically reduced to about half of the original size, with data transfer time and storage requirements being reduced accordingly. Comparisons Like any lossless compression scheme, Monkey's Audio format takes up several times as much space as lossy compression formats - typically, about twice as much as a 320 kbit/s bitrate MP3 file. The upside is that no data is lost compared to the input file, making lossless codecs suitable for transcoding, or simply taking up approximately half as much space as raw PCM data. Relative to FLAC, Apple Lossless Audio Codec, or WavPack, Monkey's Audio is slow to encode or decode files. While Monkey's Audio can achiev ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FLAC
FLAC (; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of the original audio data. FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC supports metadata tagging, album cover art, and fast seeking. History Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vorbis
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder ( codec) for lossy audio compression, libvorbis. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis. Version 1.0 of Vorbis was released in May 2000. Since 2013, the Xiph.Org Foundation has stated that the use of Vorbis should be deprecated in favor of the Opus codec, an improved and more efficient format that has also been developed by Xiph.Org. Name Vorbis is named after the character Exquisitor Vorbis in the '' Discworld'' novel '' Small Gods'' by Terry Pratchett. The Ogg format is named after ''ogging'', jargon from the computer game '' Netrek''. Development Vorbis is a continuation of audio compression development started in 1993 by Chris Montgomery. Intensive development began following a September 1998 letter from the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Pulse-code Modulation
Pulse-code modulation (PCM) is a method used to digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Digital Sound
Digital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit resolution. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form. Following significant advances in digital audio technology during the 1970s and 1980s, it gradually replaced analog audio technology in many areas of audio engineering, record production and telecommunications in the 1990s and 2000s. In a digital audio system, an analog electrical signal representing the sound is converted with an analog-to-digital converter (ADC) into a digital signal, typically using pulse-code modulation (PCM). This digital signal can then be recorded, edited, modified, and copied using computers, a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
3GP And 3G2
3GP (3GPP file format) is a digital multimedia container format defined by the Third Generation Partnership Project (3GPP) for 3G UMTS multimedia services, largely based on MPEG-4 Part 12. A 3GP container may consist of H.263 or H.264 video codecs or AMR or AAC-LC audio codecs. 3G2 (3GPP2 file format) is a multimedia container format defined by the 3GPP2 for 3G CDMA2000 multimedia services. It is very similar to the 3GP file format but consumes less space and bandwidth, and has some extensions and limitations in comparison to 3GP. Specifications 3GP is defined in the ETSI 3GPP technical specification.ETSI 3GP3GPP TS 26.244; Transparent end-to-end packet switched streaming service (PSS); 3GPP file format (3GP)Retrieved on 2009-06-02. 3GP is a required file format for video and associated speech/audio media types and timed text in ETSI 3GPP technical specifications for IP Multimedia Subsystem (IMS), Multimedia Messaging Service (MMS), Multimedia Broadcast/Multicast ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
