PCMU (G.711U)
type: Audio
makers: ITU-T
Required bandwidth: 64Kbps (90.4)
Features: Both PCMU and PCMA can provide better voice quality, but they occupy a higher bandwidth and require 64kbps.
Advantages: excellent voice quality
Disadvantages: high bandwidth
usage Application field: voip
royalty method: Free
Remarks: Both PCMU and PCMA can achieve CD sound quality, but they also consume the most bandwidth (64kbps). If the network bandwidth is relatively low, you can choose a low bit rate encoding method, such as G.723 or G.729, these two encoding methods can also achieve the sound quality of traditional long-distance calls, but require very little bandwidth (G723 requires 5.3/ 6.3kbps, G729 requires 8kbps). If the bandwidth is sufficient and better voice quality is required, PCMU and PCMA can be used, and even wideband encoding method G722 (64kbps) can be used, which can provide high fidelity sound quality.
PCMA (G.711A)
type: Audio
makers: ITU-T
Required bandwidth: 64Kbps (90.4)
Features: Both PCMU and PCMA can provide better voice quality, but they occupy a higher bandwidth and require 64kbps.
Advantages: excellent voice quality
Disadvantages: high bandwidth
usage Application field: voip
royalty method: Free
Remarks: Both PCMU and PCMA can achieve CD sound quality, but they also consume the most bandwidth (64kbps). If the network bandwidth is relatively low, you can choose a low bit rate encoding method, such as G.723 or G.729, these two encoding methods can also achieve the sound quality of traditional long-distance calls, but require very little bandwidth (G723 requires 5.3/ 6.3kbps, G729 requires 8kbps). If the bandwidth is sufficient and better voice quality is required, PCMU and PCMA can be used, and even wideband encoding method G722 (64kbps) can be used, which can provide high fidelity sound quality.
ADPCM (Adaptive Differential Pulse Code Modulation)
Type: Audio
Formulator: ITU-T
Required Bandwidth: 32Kbps
Features: ADPCM (adaptive difference pulse code modulation) combines the adaptive characteristics of APCM and the differential characteristics of the DPCM system, and is a performance Better waveform encoding. Its core idea is:
①Use the idea of self-adaptation to change the size of the quantization step, that is, use a small quantization step (step-size) to encode a small difference, and use a large quantization step to encode a large difference;
②Use Past sample values estimate the predicted value for the next input sample such that the difference between the actual sample value and the predicted value is always minimized.
Advantages: low algorithm complexity, small compression ratio (CD sound quality > 400kbps), shortest codec delay (compared to other technologies) Disadvantages:
average sound quality
Application field: voip
royalty method: Free
Remarks: ADPCM (ADPCM Adaptive Differential Pulse Code Modulation ), is a lossy compression algorithm for 16bit (or higher?) sound waveform data, it stores the 16bit data of each sample in the sound stream as 4bit, so the compression ratio is 1:4. And the compression/decompression The compression algorithm is very simple, so it is a good way to obtain high-quality sound with low space consumption.
LPC (Linear Predictive Coding, Linear Predictive Coding)
Type: Audio
Formulator:
Required Bandwidth: 2Kbps-4.8Kbps
Features: Large compression ratio, large amount of calculation, low sound quality, cheap
Advantages: Large compression ratio, cheap
Disadvantages: Calculation The volume is large, the voice quality is not very good, and the naturalness is low.
Application field: voip
Royalty method: Free
Remarks: Parameter coding is also called sound source coding, which is to extract characteristic parameters of the source signal in the frequency domain or other orthogonal transform domains. and transform it into digital code for transmission. Decoding is its inverse process. The received digital sequence is transformed to restore the characteristic parameters, and then the speech signal is reconstructed according to the characteristic parameters. Specifically, parametric coding is to extract and encode the characteristic parameters of the speech signal, trying to make the reconstructed speech signal as accurate as possible, but the waveform of the reconstructed signal may be quite different from that of the original speech signal. Such as: linear predictive coding (LPC) and various other improved types belong to parameter coding. The encoding bit rate can be compressed to 2Kbit/s-4.8Kbit/s, or even lower, but the voice quality can only reach a medium level, especially the low naturalness.
CELP (Code Excited Linear Prediction, Code Excited Linear Prediction Coding)
type: Audio
makers: European Telecommunications Standards Institute (ETSI)
Required bandwidth: 4-16Kbps rate
Features: Improve voice quality:
① Perceived weighting of error signals , using the masking characteristics of human hearing to improve the subjective quality of speech;
② using fractional delay to improve pitch prediction, making the expression of voiced sounds more accurate, especially improving the quality of female speech;
③ using the modified MSPE criterion to find the "best" delay, which makes the shape of the pitch cycle delay smoother;
④According to the efficiency of long-term prediction, adjust the size of the random excitation vector to improve the subjective quality of speech; ⑤Use an adaptive smoother based on channel error rate estimation, Higher natural speech can also be synthesized at a higher bit rate.
Conclusions:
① CELP algorithm can obtain satisfactory compression effect in low-rate coding environment;
② Using fast algorithm can effectively reduce the complexity of CELP algorithm, so that it can be realized in real time;
③ CELP can successfully compress various Different types of speech signals are encoded, and this adaptability is more important for real environments, especially when background noise exists.
Advantages: Provides clearer voice with very low bandwidth
Disadvantages:
Application field: voip
royalty method: Free
Remarks: In 1999, the European Telecommunications Standards Institute (ETSI) launched the third-generation mobile communication speech coding standard Adaptive Multi-Rate Speech Coder (AMR) based on code-excited linear predictive coding (CELP), the lowest rate of which is 4.75kb/s , to achieve communication quality. CELP Code Excited Linear Predictive Coding is the abbreviation of Code Excited Linear Prediction. CELP is the most successful speech coding algorithm in the past 10 years.
The CELP speech coding algorithm uses linear prediction to extract the vocal tract parameters, uses a codebook containing many typical excitation vectors as the excitation parameter, and searches for an optimal excitation vector in this codebook every time encoding, the encoding of this excitation vector The value is the sequence number in the codebook for this sequence.
CELP has been adopted by many speech coding standards, and the American federal standard FS1016 adopts CELP coding method, which is mainly used for high-quality narrowband speech confidential communication. CELP (Code-Excited Linear Prediction) This is a simplified LPC algorithm, known for its low bit rate (4800-9600Kbps), with very clear voice quality and high background noise immunity. CELP is a speech compression coding scheme widely used in medium and low rates.
G.711
Type: Audio
Formulator: ITU-T
Required Bandwidth: 64Kbps
Features: Small Algorithm Complexity, General Sound Quality
Advantages: Low Algorithm Complexity, Small Compression Ratio (CD Sound Quality > 400kbps), Shortest Codec Delay ( Relative to other technologies)
Disadvantages: high bandwidth occupied
Application field: voip
royalty method: Free
Remarks: G.711 64kb/s pulse code modulation PCM announced by CCITT in the 1970s.
G.721
Type: Audio
Formulator: ITU-T
Required Bandwidth: 32Kbps
Features: Compared with PCMA and PCMU, its compression ratio is higher, and it can provide a compression ratio of 2:1.
Advantages: Large compression ratio
Disadvantages: Average sound quality
Application field: voip
royalty method: Free
Remarks: Sub-band ADPCM (SB-ADPCM) technology. The G.721 standard is a code conversion system. It uses ADPCM conversion technology to realize mutual conversion between 64 kb/s A-law or μ-law PCM rate and 32 kb/s rate.
G.722
Type: Audio
Formulator: ITU-T
Required Bandwidth: 64Kbps
Features: G722 can provide high-fidelity voice quality
Advantages: Good sound quality
Disadvantages: High bandwidth requirements
Application Field: voip
Royalty Method: Free
Remarks: Sub-band ADPCM (SB-ADPCM) technology
G.723 (Low Bit Rate Speech Coding Algorithm)
Type: Audio
Formulator: ITU-T
Required Bandwidth: 5.3Kbps/6.3Kbps
Features: The voice quality is close to good, the bandwidth requirement is low, efficient implementation, easy for multi-channel expansion, and can be Realize 53coder by utilizing 16kRAM in C5402 chip. Reach the voice quality required by ITU-TG723, and the performance is stable. It can be used for IP phone voice source coding or high-efficiency voice compression storage.
Advantages: low bit rate, small bandwidth requirements. And achieve the voice quality required by ITU-TG723, the performance is stable.
Disadvantages: General sound quality
Application field: voip
Royalty method: Free
Remarks: G.723 voice encoder is a dual-bit-rate encoding scheme for multimedia communication with encoding rates of 5.3kbits/s and 6.3kbit/s. The G.723 standard is an integral part of the multimedia communication standard formulated by the International Telecommunication Union (ITU), and can be applied to systems such as IP phones. Among them, the 5.3kbits/s code rate encoder adopts multi-pulse maximum likelihood quantization technology (MP-MLQ), and the 6.3kbits/s code rate encoder adopts algebraic code excitation linear prediction technology.
G.723.1 (Double-rate Speech Coding Algorithm)
Type: Audio
Maker: ITU-T
Required Bandwidth: 5.3Kbps (22.9)
Features: It can compress and decompress music and other audio signals, but it is for voice signals is optimal. G.723.1 employs silence compression that performs discontinuous transmission, which means that artificial noise is added to the bitstream during periods of silence. In addition to reserving bandwidth, this technique keeps the modem of the transmitter in continuous operation and avoids the on and off of the carrier signal.
Advantages: low bit rate, small bandwidth requirements. And achieve the voice quality required by ITU-TG723, stable performance, avoiding the on and off of the carrier signal.
Disadvantages: general voice quality
Application field: voip
royalty method: Free
Remarks: G.723.1 algorithm is a compression algorithm recommended by ITU-T for voice or other audio signals in low-rate multimedia services, and its target application systems include H.323, H.324 and other multimedia communication systems. At present, this algorithm has become one of the mandatory algorithms in the IP telephone system.
G.728
Type: Audio
Formulator: ITU-T
Required Bandwidth: 16Kbps/8Kbps
Features: Used in many fields such as IP telephone, satellite communication, and voice storage. G.728 is a low-latency coder, but it is more complex than other coders because the 50-order LPC analysis must be repeated in the coder. G.728 also uses an adaptive post-filter to improve its performance.
Advantages: backward adaptive, using adaptive post-filter to improve its performance
Disadvantages: more complex than other encoders
Application field: voip
royalty method: Free
Remarks: G.728 16kb/s short-delay codebook excitation linear Predictive Coding (LD-CELP). In 1996, ITU announced the CS-ACELP algorithm of G.728 8kb/s, which can be used in many fields such as IP telephone, satellite communication, and voice storage. 16 kbps G.728 low-latency code-excited linear prediction.
G.728 is a hybrid of a low-bit linear predictive analysis-by-synthesis coder (G.729 and G.723.1) and a backward ADPCM coder. G.728 is an LD-CELP encoder, which only processes 5 samples at a time. For low-rate (56~128 kbps) Integrated Services Digital Network (ISDN) videophones, G.728 is a proposed speech coder. Because of its backward adaptive nature, G.728 is a low-delay coder, but it is more complex than other coders because the 50-order LPC analysis must be repeated in the coder. G.728 also uses an adaptive post-filter to improve its performance.
G.729
Type: Audio
Formulator: ITU-T
Required Bandwidth: 8Kbps
Features: To achieve long-distance call quality under good channel conditions, in the case of random bit errors, frame loss and multiple transfers, etc. To have good robustness etc. This voice compression algorithm can be used in a wide range of fields, including IP telephony, wireless communications, digital satellite systems and digital dedicated lines.
The G.729 algorithm adopts the "conjugate structure algebraic codebook excitation linear predictive coding scheme" (CS-ACELP) algorithm. This algorithm combines the advantages of waveform coding and parameter coding, and is based on adaptive predictive coding technology, and uses vector quantization, compositional analysis and sensory weighting.
The G.729 encoder is designed for low-latency applications. Its frame length is only 10ms, and the processing delay is also 10ms. In addition, the look-ahead of 5ms makes the point-to-point delay generated by G.729 as 25ms with a bit rate of 8 kbps.
Advantages: good voice quality, wide range of applications, using vector quantization, synthesis analysis and perceptual weighting, providing a hidden processing mechanism for frame loss and packet loss Disadvantages: poor performance in dealing with random bit errors
.
Application field: voip
Royalty method: Free
Note: The International Telecommunication Union (ITU-T) formally passed G.729 in November 1995. The ITU-T recommendation G.729 is also called "Conjugate Structure Algebraic Codebook Excited Linear Predictive Coding Scheme" (CS-ACELP), which is a newer speech compression standard at present. G.729 was jointly developed by several well-known international telecommunications entities in the United States, France, Japan, and Canada.
G.729A
Type: Audio
Formulator: ITU-T
Required Bandwidth: 8Kbps (34.4)
Features: The complexity is lower than G.729, and the performance is worse than G.729.
Advantages: good voice quality, reduced computational complexity for real-time implementation, and provides a hidden processing mechanism for frame loss and packet loss
Disadvantages: poorer performance than G.729
Application field: voip
royalty method: Free
Remarks: ITU in 1996 -T also formulated G.729A, a simplified scheme of G.729, which mainly reduces the complexity of calculation for real-time implementation, so G.729A is currently used.
GIPS
Type: Audio
Developer: Sweden Global IP Sound Company
Required Bandwidth:
Features: GIPS technology can automatically adjust the encoding bit rate according to the bandwidth condition, providing low bit rate high-quality audio. The core technology of GIPS (network adaptive algorithm, packet loss compensation algorithm and echo cancellation algorithm) can well solve the problem of voice delay and echo, bring perfect sound quality, and provide a voice call effect that is clearer than a telephone.
Advantages: Solve the problem of voice delay and echo very well, bring perfect sound quality, and provide clearer voice call effect than telephone Disadvantages:
Not Free
Application field: voip
royalty method: Pay a fee for the right to use every year Remarks
: GIPS audio technology is The voice compression engine system dedicated to the Internet is provided by "GLOBAL IP SOUND", the world's top voice processing high-tech company from Sweden. GIPS technology can automatically adjust the encoding bit rate according to the bandwidth condition, providing low bit rate high-quality audio. The core technology of GIPS (network adaptive algorithm, packet loss compensation algorithm and echo cancellation algorithm) can well solve the problem of voice delay and echo, bring perfect sound quality, and provide a voice call effect that is clearer than a telephone.
The encoding technology GIPS used by QQ makes the sound clearer in actual use, but the sound is a bit bad compared to SIP encoding. May I ask, how much bandwidth does GIPS theoretically occupy? Can the GIPS encoding method be added to SIP?
The voice coding technology used by GIPS is iLBC coding. GIPS technology can automatically adjust the encoding bit rate according to the bandwidth condition, providing low bit rate high-quality audio.
The core technology of GIPS (network adaptive algorithm, packet loss compensation algorithm and echo cancellation algorithm) can well solve the problem of voice delay and echo, bring perfect sound quality, and provide a voice call effect that is clearer than a telephone.
Regarding the iLBC used by GIPS, there is a special explanation in the previous blog
Apt-X
type: Audio
Developer: Audio Processing Technology Company
Required bandwidth: 10Hz to 22.5 kHz, 56kbit/s to 576 kbit/s (16 bit 7.5 kHz mono to 24-bit, 22.5kHz stereo)
Features: Mainly used Provide high-quality audio in the field of professional audio. Its characteristics are:
①Adopt 4:1:4 compression and amplification scheme;
②Low hardware complexity;
③Extremely low encoding delay;
④Realized by a single chip;
⑤Mono or stereo codec;
⑥Only need a single device 22.5kHz dual-channel stereo can be realized;
⑦Sampling frequency up to 48kHz; ⑧Good fault tolerance
; ⑨Complete
AUTOSYNC™ codec synchronization scheme; Low Disadvantage: Not Free Application field: voip Royalty method: One-time payment Note: Sub-band ADPCM (SB-ADPCM) technology
NICAM (Near Instantaneous Companded Audio Multiplex quasi-instantaneous companded audio multiplex)
Type: Audio
Formulator: British BBC Broadcasting Corporation
Required bandwidth: 728Kbps
Features: It has a wide range of applications, and it can be used for stereo or bilingual broadcasting
Advantages: Application range It is widely used, with high signal-to-noise ratio, wide dynamic range, and sound quality comparable to CDs, so it is named NICAM, so NICAM is also called NICAM
Disadvantages: Not Free, high bandwidth requirements
Application field: voip
Royalty method: One-time payment
Remarks: NICAM is also called NICAM, which is the abbreviation of Near-Instantaneously Companded Audio Multiplex in English, and its meaning is quasi-instantaneously companded audio multiplexing.
In layman's terms, NICAM technology is actually two-channel digital sound technology, and its application scope is extremely wide. Spectrum resource of the channel. This can be achieved without adding much investment to conventional television broadcasting. When broadcasting in stereo, it improves the signal quality of the audio, bringing it closer to CD quality. Moreover, NICAM technology can also be used for high-speed data broadcasting and other data transmission multiplication services, which seems to be particularly important in today's information society!
MPEG-1 audio layer 1
type: Audio
Maker: MPEG
required bandwidth: 384kbps (compressed 4 times)
Features: simple encoding, used for digital cassette tapes, 2 channels, the audio compression scheme used in VCD is MPEG -1 layer I.
Advantages: The compression method is much more complicated than the time-domain compression technology, and the coding efficiency and sound quality are also greatly improved, and the coding delay is correspondingly increased. Can achieve "completely transparent" sound quality (EBU sound quality standard)
Disadvantages: high bandwidth requirements
Application field: voip
royalty method: Free
Remarks: MPEG-1 sound compression coding is the first international high-fidelity sound data compression international Standard, it is divided into three levels:
--Layer 1 (Layer 1): simple encoding, used for digital audio cassette tapes
--Layer 2 (Layer 2): medium algorithm complexity, used for digital audio broadcasting (DAB) And VCD, etc.--
Layer 3 (Layer 3): complex encoding, used for transmission of high-quality sound on the Internet, such as MP3 music compression 10 times
MUSICAM (MPEG-1 audio layer 2, that is, MP2)
type: Audio
makers: MPEG
required bandwidth: 256 ~ 192kbps (compression 6 ~ 8 times)
features: algorithm complexity is medium, used for digital audio broadcasting (DAB) and VCD, etc., 2-channel, and MUSICAM is widely used in the production, exchange, storage, and transmission of digital programs such as digital studios, DAB, and DVB due to its appropriate complexity and excellent sound quality.
Advantages: The compression method is much more complicated than the time-domain compression technology, and the coding efficiency and sound quality are also greatly improved, and the coding delay is correspondingly increased. Can achieve "completely transparent" sound quality (EBU sound quality standard)
Disadvantages:
Application field: voip
Royalty method: Free
Remarks: Same as MPEG-1 audio layer 1
MP3 (MPEG-1 audio layer 3)
type: Audio
makers: MPEG
required bandwidth: 128 ~ 112kbps (compression 10 ~ 12 times)
features: complex encoding, used for high-quality sound transmission on the Internet, such as MP3 music Compressed 10 times, 2 channels. MP3 is a hybrid compression technology based on the advantages of MUSICAM and ASPEC. Under the technical conditions at that time, the complexity of MP3 was relatively high, and the encoding was not conducive to real-time. However, due to the high level of MP3 under the condition of low bit rate Excellent sound quality, making it the darling of soft decompression and Internet broadcasting.
Advantages: High compression ratio, suitable for transmission on the Internet
Disadvantages: When MP3 is 128KBitrate and below, there will be obvious high-frequency loss
Application field: voip
royalty method: Free
Remarks: Same as MPEG-1 audio layer 1
MPEG-2 audio layer
type: Audio
Maker: MPEG
Required bandwidth: Same as MPEG-1 layer 1, layer 2, layer 3
Features: MPEG-2 sound compression encoding uses the same codec as MPEG-1 sound , Layer 1, Layer 2 and Layer 3 have the same structure, but it can support 5.1-channel and 7.1-channel surround sound.
Advantages: Support 5.1-channel and 7.1-channel surround sound
Disadvantages:
Application field: voip
Royalty method: Charge by individual Remarks
: MPEG-2 sound compression encoding uses the same codec as MPEG-1 sound, layer 1, layer The structure of layer 2 and layer 3 is also the same, but it can support 5.1-channel and 7.1-channel surround sound.
AAC (Advanced Audio Coding, advanced audio coding)
type: Audio
makers: MPEG
Required bandwidth: 96-128 kbps
Features: AAC can support any number of audio channel combinations between 1 and 48, including 15 low-frequency effects audio channels, dubbing/multi-voice audio channels, and 15 channels of data. It can transmit 16 sets of programs at the same time, and the audio and data structure of each set of programs can be specified arbitrarily.
The main possible application range of AAC is concentrated on Internet network transmission, digital audio broadcasting, including satellite live broadcast and digital AM, and digital TV and theater systems. AAC uses a very flexible entropy coding core to transmit coded spectral data. With 48 main audio channels, 16 low-frequency enhancement channels, 16 integrated data streams, 16 dubbing, 16 arrangements.
Advantages: Support multiple audio channel combinations and provide high-quality sound quality
Disadvantages:
Application field: voip
royalty method: one-time fee
Remarks: AAC formed the international standard ISO 13818-7 in 1997. Advanced Audio Coding (AAC) was successfully developed and became a new generation of audio compression standard following the MPEG-2 audio standard (ISO/IEC13818-3).
In the early days of MPEG-2 formulation, it was originally intended to keep its audio coding part compatible with MPEG-1. However, it was later defined as a multi-channel audio standard for higher quality in order to meet the requirements of broadcast television. Naturally, this standard is not compatible with MPEG-1, hence the name MPEG-2 AAC. In other words, on the face of it, both making and playing AAC would require completely different tools than MP3.
Dolby AC-3
Type: Audio
Maker: American Dolby Company
Required Bandwidth: 64kbps
Features: The provided surround sound system is composed of 5 full-band channels plus a subwoofer channel, and the information of 6 channels is produced All digitized in the process of restoration and restoration, little loss of information, rich in details, with true stereo effect, widely used in digital TV, DVD and home theater.
Advantages: surround sound, little loss of information, rich in details, with real stereo effect
Disadvantages:
application field: voip
royalty method: charged by each
Remarks: Dolby Digital AC-3 (Dolby Digital AC-3): American Dolby Company The developed multi-channel full-band sound coding system provides a surround sound system consisting of 5 full-band channels plus a subwoofer channel. The information of the 6 channels is all digitized during the production and restoration process, and the information loss is very small. Less, rich in detail, with a true stereo effect, widely used in digital TV, DVD and home theater.
ASPEC (Audio Spectral Perceptual Entropy Coding)
Type: Audio
Formulator: AT&T
Required Bandwidth: 64kps
Features: The audio quality has been significantly improved, but the computational complexity has also been greatly increased, and the sound quality is seriously degraded when reverberation and low bit rate.
Pros: Dramatically improved audio quality
Cons: Increased computational complexity. Block boundary effects, increased precomputation complexity. The sound quality is seriously degraded when reverberation and low bit rate are
applied. Application field: voip
royalty method: charge by individual
Remarks: transformation compression technology
PAC (Perceptual Audio Coder)
Type: Audio
Maker: AT&T
Required Bandwidth: 64kps
Features: The audio quality has been significantly improved, but the sound quality is seriously degraded when reverberation and low bit rate.
Advantages: Significantly improved audio quality
Disadvantages: Block boundary impact, pre-echo, and low bit rate sound quality is severely degraded
Application field: voip
royalty method: Charge by individual
Remarks: Transform compression technology
HR
Type: Audio
Formulator: Philips
Required Bandwidth: 8Kbps
Features: The purpose is to increase the capacity of GSM network, but it will damage the voice quality; due to the current shortage of network frequencies, some large operators have opened this method in densely populated areas of large cities to increase capacity.
Advantages: Large system capacity
Disadvantages: Poor voice quality
Application field: GSM
royalty method: Charge by each
Remark: HF half-rate, is a GSM voice coding method.
FR
type: Audio
Maker: Philips
Required bandwidth: 13Kbps
Features: It is the communication coding method of general GSM mobile phones, and can obtain the voice communication quality of about 4.1 Qos (ITU stipulates that the voice communication quality Qos full score is 5)
Advantages : The voice quality has been improved.
Disadvantage: The system capacity is reduced.
Application field: GSM
royalty method: Charge by one
Remark: FR full rate is a GSM voice coding method.
EFR
Type: Audio
Maker: Philips
Required Bandwidth: 13Kbps
Features: For GSM mobile phones based on full-rate 13Kbps voice coding and transmission, can get better and clearer voice quality (close to Qos4.7), need network service provider Only when this network function is enabled, the mobile phone can cooperate with it.
Advantages: Good sound quality
Disadvantages: The network service provider needs to open this network function, and the system capacity is reduced
Application field: GSM
royalty method: Charge by each
Remark: EFR enhanced full rate, a GSM network voice encoding method.
GSM-AMR (Adaptive Multi-Rate)
type: Audio
Developer: Philips
Required bandwidth: 8Kbps (4.75 Kbps~12.2 Kbps)
Features: Can replace and silence voice, smooth noise, support intermittent transmission, voice Dynamic scouting. It can provide high-quality voice effects under various network conditions.
Advantages: Excellent sound quality
Disadvantages:
Application field: GSM
Royalty method: Fee per unit
Remarks: GSM-ASM is an audio standard widely used in GPRS and W-CDMA networks. GSM-AMR is defined in the specification ETSI GSM06.90. AMR voice coding is the default coding standard of GSM 2+ and WCDMA, and it is the voice coding standard of the third generation wireless communication system. The GSM-AMR standard is based on ACELP (Algebraically Excited Linear Prediction) coding. It can provide high-quality voice effects under a wide range of transmission conditions.
EVRC (Enhanced Variable Rate Coder, Enhanced Variable Rate Coder)
type: Audio
Maker: Qualcomm Communications Corporation (Qualcomm) in the United States
Required bandwidth: 8Kbps or 13Kbps
Features: Support three bit rates (9.6 Kbps, 4.8 Kbps and 1.2 Kbps), noise suppression, mail filtering. It can provide high-quality voice effects under various network conditions.
Advantages: Excellent sound quality
Disadvantages:
Application field: CDMA
Royalty method: Charge by each
Remark: EVRC coding is widely used in CDMA network. The EVRC standard follows the content of the specification TIA IS-127. EVRC coding is based on the RCELP (Relaxed Code Excited Linear Prediction) standard. The encoding can operate with the capacity of Rate 1 (171bits/packet), Rate 1/2 (80bits/packet) or Rate 1/8 (16bits/packet). On request, it can also generate empty packets (0bits/packet).
QCELP (QualComm Code Excited Linear Predictive, stimulated linear predictive coding)
Type: Audio
Formulator: Qualcomm Communications Corporation (Qualcomm) in the United States
Required bandwidth: 8k speech coding algorithm (can work at 4/4.8/8/9.6Kbps Wait for a fixed rate, and work at a variable rate between 800Kbps and 9600Kbps)
Features: Use an appropriate threshold to determine the required rate. QCELP is an 8k voice coding algorithm (it can provide voice compression quality close to 13k at a rate of 8k). This is a variable rate speech coding, according to the characteristics of human speech (you should be able to understand that our daily communication and communication do not always maintain a constant way of speaking, there are interruptions, different sound frequencies, etc. are human An optimization technique adopted by the natural expression of .
Advantages: clear voice, low background noise, large system capacity
Disadvantages: not free
Application field: CDMA
royalty method: pay a fee for the right to use every year
Note: QCELP stands for QualComm Code Excited Linear Predictive (QualComm Excited Linear Predictive Coding). The patented voice coding algorithm of Qualcomm Communications Corporation of the United States is the voice coding standard (IS95) of the second generation digital mobile phone (CDMA) in North America. This algorithm can not only work at fixed rates such as 4/4.8/8/9.6kbit/s, but also work at variable rates between 800bit/s and 9600bit/s. The QCELP algorithm is considered to be the most efficient algorithm so far. One of its main features is to use an appropriate threshold to determine the required rate. The I'1 limit value varies with the noise level of the scene, which suppresses the background noise, so that even in a noisy environment, good voice quality can be obtained. The voice of CDMA8Kbit/s is similar to that of GSM 13Mbit/s. CDMA adopts a series of technologies such as QCELP coding, and has the advantages of clear voice and low background noise. Its performance is obviously better than other wireless mobile communication systems, and its voice quality can be compared with that of wired telephones. Low wireless radiation.