TRAU F
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TRAU F
Transcoder and Rate Adaptation Unit or TRAU, performs transcoding function for speech channels and RA (Rate Adaptation) for data channels in the GSM network. The Transcoder/Rate Adaptation Unit (TRAU) is the data rate conversion unit. The PSTN/ISDN switch is a switch for 64 kbit/s voice. Current technology permits to decrease the bit-rate (in GSM radio interface it is 16 kbit/s for full rate and 8 kbit/s for half rate). Since MSC is basically a PSTN/ISDN switch its bit-rate is still 64 kbit/s. That is why a rate conversion is required in between the BSC and MSC... Transcoding is the compression of speech data from 64 kbit/s to 13/12.2/6.5 kbit/s in case FR/EFR/HR (respectively) speech coding. Rate adaptation without transcoding allows Tandem Free Operation (TFO), allowing the original encoded speech data to be carried in a 64 kbit/s channel. TFO offers benefits because transcoding can lead to a degradation of speech quality and requires computational resources. Brief explanati ...
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Transcoding
Transcoding is the direct digital-to-digital conversion of one encoding to another, such as for video data files, audio files (e.g., MP3, WAV), or character encoding (e.g., UTF-8, ISO/IEC 8859). This is usually done in cases where a target device (or workflow) does not support the format or has limited storage capacity that mandates a reduced file size, "Advancements in Compression and Transcoding: 2008 and Beyond", Society of Motion Picture and Television Engineers (SMPTE), 2008, webpageSMPTE-spm or to convert incompatible or obsolete data to a better-supported or modern format. In the analog video world, transcoding can be performed just while files are being searched, as well as for presentation. For example, Cineon and DPX files have been widely used as a common format for digital cinema, but the data size of a two-hour movie is about 8 terabytes (TB). That large size can increase the cost and difficulty of handling movie files. However, transcoding into a JPEG2000 ...
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Tandem Free Operation
Tandem Free Operation (TFO) is a part of ETSI's 3GPP standard specification, which has been included from R99 of the standards specifications onwards. Overview In traditional GSM networks, a call between two Mobile Stations (MS) involve a dual encoding/decoding process. Speech signals are first encoded in the originating MS, converted to G.711 in the local transcoder, converted back to a GSM codec in the remote transcoder and finally converted back to speech at the terminating MS. In this configuration the two transcoders are operating in tandem introducing a voice quality degradation. It is possible to eliminate this problem by removing the two transcoding operations in the voice path if the two MS are using the same codec. Details Broadly, the equipments that are en-route two end mobile sets can be categorized into two types: * Active voice equipments, that does the transcoding operation, either from a GSM/UMTS speech codec (e.g.: GSM-EFR, GSM-AMR) to G.711/PCM or opposite. ...
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Full Rate
Full Rate (FR or GSM-FR or GSM 06.10 or sometimes simply GSM) was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s). The quality of the coded speech is quite poor by modern standards, but at the time of development (early 1990s) it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate. Technology ''GSM-FR'' is specified in ETSI 06.10 (ETS 300 961) and is based on RPE-LTP ( Regular Pulse Excitation - Long Term Prediction) speech coding paradigm. Like many other linear predicti ...
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Half Rate
Half Rate (HR or GSM-HR or GSM 06.20) is a speech coding system for GSM, developed in the early 1990s. Since the codec, operating at 5.6 kbit/s, requires half the bandwidth of the Full Rate codec, network capacity for voice traffic is doubled, at the expense of audio quality. The sampling rate is 8 kHz with resolution 13 bit, frame length 160 samples (20 ms) and subframe length 40 samples (5 ms). GSM Half Rate is specified in ETSI EN 300 969 (GSM 06.20), and uses a form of the VSELP algorithm. Previous specification was in ETSI ETS 300 581–2, which first edition was published in December 1995.ETSI ETS 300 581-2 - Half rate speech transcoding (GSM 06.20 version 4.3.1)
Retrieved on 2009-07-11 For some Nokia phones one can configure the use of this codec: * To a ...
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Adaptive Multi-Rate
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s. AMR was adopted as the standard speech codec by 3GPP in October 1999 and is now widely used in GSM and UMTS. It uses link adaptation to select from one of eight different bit rates based on link conditions. AMR is also a file format for storing spoken audio using the AMR codec. Many modern mobile telephone handsets can store short audio recordings in the AMR format, and both Free software, free and proprietary programs exist (see #Software support, Software support) to convert between this and other formats, although AMR is a speech format and is unlikely to give ideal results for other audio. The common filename extension is .amr. ...
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