Sound Recording Solutions
Sound Card Recorder
Powerful voice activated microphone recorder for Windows. Click here to learn more.MP3
MP3 is a compression format. It provides a representation of pulse-code modulation-encoded (PCM) audio data in a much smaller size by discarding portions that are considered less important to human hearing (similar to JPEG, a lossy compression for images). A number of techniques are employed in MP3 to determine which portions of the audio can be discarded, including psychoacoustics. MP3 audio can be compressed with different bit rates, providing a range of tradeoffs between data size and sound quality. The MP3 format uses a hybrid transformation to transform a time domain signal into a frequency domain signal: 32-band polyphase quadrature filter 36 or 12 tap MDCT; size can be selected independent for sub-band 0...1 and 2...31 Aliasing reduction postprocessing MP3 Surround, a version of the format supporting 5.1 channels for surround sound, was introduced in December 2004. MP3 Surround is backward compatible with standard stereo MP3, and file sizes are similar. In terms of the MPEG specifications, AAC (Advanced audio coding) from MPEG-4 is to be the successor of the MP3 format, although there has been a significant movement to create and popularize other audio formats. Nevertheless, any succession is not likely to happen for a significant amount of time due to MP3's overwhelming popularity (MP3 enjoys extremely wide popularity and support, not just by end-users and software but by hardware such as DVD and CD players). A reference simulation software written in C language known as ISO 11172-5 was developed by the members of the ISO MPEG Audio committee in order to produce bit compliant MPEG Audio files (Layer 1, Layer 2, Layer 3). Working in non real time on a number of operating systems it was able to demonstrate the first real time hardware decoding (DSP based) of compressed audio. Some other real time implementation of MPEG Audio encoders were available for the purpose of digital broadcasting (radio DAB, television DVB) towards consumer receivers and set top boxes. Later on, on July 7, 1994 the Fraunhofer Society released the first software MP3 encoder called l3enc. The filename extension .mp3 was chosen by the Fraunhofer team on July 14, 1995 (previously, the files had been named .bit). With the first real-time software MP3 player Winplay3 (released September 9th, 1995) many people were able to encode and playback MP3 files on their PCs. Because of the relatively small hard drives back in that time (~500 MB) the technology was essential to store music for listening pleasure on a computer. The Internet Underground Music Archive (IUMA) is generally recognized as the start of the on-line music revolution. IUMA was the Internet's first high-fidelity music web site, hosting thousands of authorized MP2 recordings before MP3 or the web was popularized. IUMA was started by Rob Lord (who later headed pioneering Nullsoft) and Jeff Patterson, both from the University of California, Santa Cruz, in 1993. Other founding members include Jon Luini, Brandee Selck, and Ahin Savara. In the first half of 1995 through the late 1990s, MP3 files began flourishing on the Internet. MP3 popularity was mostly due to, and interchangeable with, the successes of companies and software packages like Nullsoft's Winamp (released in 1997), mpg123, and Napster (released in 1999). Those programs made it very easy for the average user to playback, create, share, and collect MP3s. Controversies regarding peer-to-peer file sharing of MP3 files have flourished in recent years - largely because high compression enables sharing of files that would otherwise be too large and cumbersome to share. Due to the vastly increased spread of MP3s through the Internet some major record labels reacted by filing a lawsuit against Napster to protect their Copyrights (see also intellectual property). Commercial online music distribution services (like the iTunes Music Store) usually prefer other/proprietary music file formats that support Digital Rights Management (DRM) to control and restrict the use of digital music. The use of formats that supports DRM is in an attempt to prevent piracy of copyright protected materials, but any computer savvy person can easily rip the DRM from a song file turning it into a file that is not locked to any computer. Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate"-that is, the number of bits of encoded data that are used to represent each second of audio. Typically rates chosen are between 128 and 320 kilobit per second. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1411.2 kbit/s (16 bits/sample ? 44100 samples/second ? 2 channels). MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" (i.e., sounds that were not present in the original recording) may appear in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because of its randomness and sharp attacks, therefore the failings of the encoder are more obvious, and are audible as ringing or pre-echo. As well as the bit rate of the encoded file, the quality of MP3 files depend on the quality of the encoder and the difficulty of the signal being encoded. For average signals with good encoders, some listeners accept the MP3 bit rate of 128 kbit/s and the CD sampling rate of 44.1 kHz as near enough to compact disc quality for them, providing a compression ratio of approximately 11:1. MP3s properly compressed at this ratio can achieve sound quality superior to that of FM radio and cassette tape, primarily due to the limited bandwidth, SNR, and other limitations of these analog media. However, listening tests show that with a bit of practice many listeners can reliably distinguish 128 kbit/s MP3s from CD originals; in many cases reaching the point where they consider the MP3 audio to be of unacceptably low quality. Yet other listeners, and the same listeners in other environments (such as in a noisy moving vehicle or at a party) will consider the quality acceptable. Obviously, imperfections in an MP3 encode will be much less apparent on low-end computer speakers than on a good stereo system connected to a computer or -- especially -- using high-quality headphones. Good encoders produce acceptable quality at 128 to 160 Kibit/s and near-transparency at 160 to 192 kbit/s, while low quality encoders may never reach transparency, not even at 320 kbit/s. It is therefore misleading to speak of 128 kbit/s or 192 kbit/s quality, except in the context of a particular encoder or of the best available encoders. A 128 kbit/s MP3 produced by a good encoder might sound better than a 192 kbit/s MP3 file produced by a bad encoder. It is important to note that quality of an audio signal is subjective. A given bit rate suffices for some listeners but not for others. Individual acoustic perception may vary, so it is not evident that a certain psychoacoustic model can give satisfactory results for everyone. Merely changing the conditions of listening, such as the audio playing system or environment, can expose unwanted distortions caused by lossy compression. The numbers given above are rough guidelines that work for many people, but in the field of lossy audio compression the only true measure of the quality of a compression process is to listen to the results. If your aim is to archive sound files with no loss of quality (or work on the sound files in a studio for example), then you should use Lossless compression algorithms, currently capable of compressing 16-bit PCM audio to 38% while leaving the audio identical to the original, such as Lossless Audio LA, Apple Lossless, TTA, FLAC, Windows Media Audio 9 Lossless (wma) and Monkey's Audio (among others). Lossless formats are strongly preferred for material that will be edited, mixed, or otherwise processed because the perceptual assumptions made by lossy encoders may not hold true after processing. The losses produced by multiple stages of coding may also compound each other, becoming more evident when the signal is reencoded after processing. Lossless formats produce the best possible result, at the expense of a lower compression ratio. Some simple editing operations, such as cutting sections of audio, may be performed directly on the encoded MP3 data without necessitating reencoding. For these operations, the concerns mentioned above are not necessarily relevant, as long as appropriate software (such as mp3DirectCut and MP3Gain) is used to prevent extra decoding-encoding steps. The bit rate is variable for MP3 files. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during play back. In the early days of MP3 encoding, a fixed bit rate was used for the entire file. Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbit/s, and the available sample frequencies are 32, 44.1 and 48 kHz. 44.1 kHz is almost always used (coincides with the sampling rate of compact discs), and 128 kbit/s has become the de facto "good enough" standard, although 192 kbit/s is becoming increasingly popular over peer-to-peer file sharing networks. MPEG-2 and [the non-official] MPEG-2.5 includes some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbit/s. Variable bit rates (VBR) are also possible. Audio in MP3 files are divided into frames (which have their own bit rate) so it is possible to change the bit rate dynamically as the file is encoded (although not originally implemented, VBR is in extensive use today). This technique makes it possible to use more bits for parts of the sound with higher dynamics (more sound movement) and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. This method compares to a sound activated tape recorder that reduces tape consumption by not recording silence. Some encoders utilize this technique to a great extent. Non-standard bitrates up to 640 kbit/s can be achieved with the LAME encoder and the --freeformat option, however only few MP3 players can play those files. The MPEG-1 standard does not include a precise specification for an MP3 encoder. The decoding algorithm and file format, as a contrast, are well defined. Implementers of the standard were supposed to devise their own algorithms suitable for removing parts of the information in the raw audio (or rather its MDCT representation in the frequency domain). During encoding 576 time domain samples are taken and are transformed to 576 frequency domain samples. If there is a transient 192 samples are taken instead of 576. This is done to limit the temporal spread of quantization noise accompanying the transient. This is the domain of psychoacoustics: the study of subjective human perception of sounds. As a result, there are many different MP3 encoders available, each producing files of differing quality. Comparisons are widely available, so it is easy for a prospective user of an encoder to research the best choice. It must be kept in mind that an encoder that is proficient at encoding at higher bitrates (such as LAME, which is in widespread use for encoding at higher bitrates) is not necessarily as good at other, lower bitrates. Decoding, on the other hand, is carefully defined in the standard. Most decoders are "bitstream compliant", meaning that the uncompressed output they produce from a given MP3 file will be the same (within a specified degree of rounding tolerance) as the output specified mathematically in the standard document. The MP3 file has a standard format which is a frame consisting of 384, 576, or 1152 samples (depends on MPEG version and layer) and all the frames have associated header information (32 bits) and side information (9, 17, or 32 bytes, depending on MPEG version and stereo/mono). The header and side information help the decoder to decode the associated huffman encoded data correctly. Therefore, for the most part, comparison of decoders is almost exclusively based on how computationally efficient they are (i.e., how much memory or CPU time they use in the decoding process). A "tag" is data stored in an MP3 (as well as other formats) that contains metadata such as the title, artist, album, track number or other information about the MP3 file to be added to the file itself. The most widespread standard tag formats are currently the ID3 ID3v1 and ID3v2 tags, and the more recent APEv2 tag. APEv2 was originally developed for the MPC file format (see the APEv2 specification). APEv2 can coexist with ID3 tags in the same file, but it can also be used by itself.
Phone Call Recorder
Must have software for voice modem. Record all phone calls automatically, watch Caller ID information, create you own powerful answering machine. Perfect sound quality. Click here to learn more.3d audio effect - 3gp - a-law - aac - acm - adc - aiff - aliasing - amplifier - amr-wb - amr-wb plus - amr - apev2 tag - asf - atrac - audio codec - audio compression - avi - bitrate - bitrate peeling - chord - codec - comfort noise - compact audio cassette - compact disc - compression - compression artifact - compression ratio - contact microphone - container format - dab - data compression - digital audio - digital camera - divx - dolby digital - dolby digital plus - dsp - dvd-audio - dvd - effects unit - equalization - ffmpeg - flanging - fourcc - frequency spectrum - hi-fi - high-end audio - hiln - id3 - joint stereo - laser microphone - line level - lossy - loudspeaker - matroska - mcf - microphone - midi - mixing console - mp2 - mp3 - mp3 surround - mp3 sx - mp3pro - mp4 - mpeg-1 - mpeg-21 - mpeg-3 - mpeg-4 - mpeg-7 - mpeg - mu-law - musepack - music - mxf - nut - osm - parabolic microphone - pcm - perception - phonograph - pink noise - pqf - psychoacoustics - qdesign - quadraphonic - radio receiver - ratdvd - realaudio - red noise - reverberation - rhythm - ribbon microphone - riff - rmvb - sbr - signal processing - sound - sound card - sound effects - sound recording - spdif - speech encoding - speech recognition - speex - stereo - subwoofer - surround sound - synthesizer - tape recorder - tdm - tweeter - video - video compression - vob - voice analysis - vorbis - wav - white noise - wma - woofer