Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates. Another main improvement is the use of transform coding (transform coded excitation - TCX) additionally to ACELP. This greatly improves the generic audio coding. Automatic switching between transform coding and ACELP provides both good speech and audio quality with moderate bit rates.
As AMR-WB operates at internal sampling rate 12.8 kHz, AMR-WB+ also supports various internal sampling frequencies ranges from 12.8 kHz to 38.4 kHz. AMR-WB uses 16 kHz sampling frequency with a resolution of 14 bits left justified in a 16-bit word. AMR-WB+ uses 16/24/32/48 kHz sampling frequencies with a resolution of 16 bits in a 16-bit word.
3GPP originally developed the AMR-WB+ audio codec for streaming and messaging services in Global System for Mobile communications (GSM) and Third Generation (3G) cellular systems. Its primary target applications are Packet-Switched Streaming service (PSS), Multimedia Messaging Service (MMS) and Multimedia Broadcast and Multicast Service (MBMS).
The AMR-WB+ codec has a wide bit-rate range, from 5.2–48 kbit/s. Mono rates are scalable from 5.2–36 kbit/s, and stereo rates are scalable from 6.2–48 kbit/s, reproducing bandwidth up to 20 kHz (approaching CD quality). Moreover, it provides backward compatibility with AMR wideband.
|Extended Adaptive Multi-Rate - Wideband (AMR-WB+)|
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AMR-WB+ compression incorporate several patents of Nokia Corporation, Telefonaktiebolaget L. M. Ericsson and VoiceAge Corporation. VoiceAge Corporation is the License Administrator for the AMR and AMR-WB+ patent pools. VoiceAge also accepts submission of patents for determination of their possible essentiality to these standards.
Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using similar methodology as algebraic code excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP.
AMR-WB is codified as G.722.2, an ITU-T standard speech codec, formally known as Wideband coding of speech at around 16 kbit/s using Adaptive Multi-Rate Wideband (AMR-WB). G.722.2 AMR-WB is the same codec as the 3GPP AMR-WB. The corresponding 3GPP specifications are TS 26.190 for the speech codec and TS 26.194 for the Voice Activity Detector.The AMR-WB format has the following parameters:
Frequency bands processed: 50-6400 Hz (all modes) plus 6400-7000 Hz (23.85 kbit/s mode only)
Delay frame size: 20 ms
Look ahead: 5 ms
AMR-WB codec employs a bandsplitting filter; the one-way delay of this filter is 0.9375 ms
Complexity: 38 WMOPS, RAM 5.3KWords
Voice activity detection, discontinuous transmission, comfort noise generator
Fixed point: Bit-exact C
Floating point: under work.A common file extension for AMR-WB file format is .awb. There also exists another storage format for AMR-WB that is suitable for applications with more advanced demands on the storage format, like random access or synchronization with video. This format is the 3GPP-specified 3GP container format based on ISO base media file format. 3GP also allows use of AMR-WB bit streams for stereo sound.Adaptive Multi-Rate audio codec
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 and proprietary programs exist (see 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. There also exists another storage format for AMR that is suitable for applications with more advanced demands on the storage format, like random access or synchronization with video. This format is the 3GPP-specified 3GP container format based on ISO base media file format.Enhanced full rate
Enhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quite poor quality of GSM-Full Rate (FR) codec. Working at 12.2 kbit/s the EFR provides wirelike quality in any noise free and background noise conditions. The EFR 12.2 kbit/s speech coding standard is compatible with the highest AMR mode (both are ACELP). Although the Enhanced Full Rate helps to improve call quality, this codec has higher computational complexity, which in a mobile device can potentially result in an increase in energy consumption as high as 5% compared to 'old' FR codec.
Enhanced Full Rate was developed by Nokia and the Université de Sherbrooke (Canada). In 1995, ETSI selected the Enhanced Full Rate voice codec as the industry standard codec for GSM/DCS.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. 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.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. It is recommended to use this codec when the battery is low as it may consume up to 30% less energy. 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.For some Nokia phones one can configure the use of this codec:
To activate HR codec use enter the following code: *4720#
To deactivate HR codec use enter the following code: #4720#RTP audio video profile
The RTP audio/video profile (RTP/AVP) is a profile for Real-time Transport Protocol (RTP) that specifies the technical parameters of audio and video streams. RTP specifies a general-purpose data format, but doesn't specify how encoded data should utilize the features of RTP (what payload type value to put in the RTP header, what sampling rate and clock rate [the rate at which the RTP timestamp increments] to use, etc.). An RTP profile specifies these details. The RTP audio/video profile specifies a mapping of specific audio and video codecs and their sampling rates to RTP payload types and clock rates, and how to encode each data format as an RTP data payload, as well as specifying how to describe these mappings using Session Description Protocol (SDP).Wideband audio
Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 80 Hz to 14 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz or even up to 22 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16-bits to encode samples, also resulting in much better voice quality.
See Compression methods for techniques and Compression software for codecs