PCMU(G.711U)
Type: Audio
Maker: ITU-T
Required bandwidth: 64Kbps (90.4)
Features: PCMU and PCMA can provide better voice quality, but they occupy a higher bandwidth, requiring 64kbps.
Advantages: excellent voice quality
Disadvantages: higher bandwidth usage
Application field: voip
Royalty method: Free
Note: PCMU and PCMA can both 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 needs 8kbps). If the bandwidth is sufficient and better voice quality is needed, PCMU and PCMA can be used, or even the wideband encoding method G722 (64kbps) can be used, which can provide high-fidelity sound quality.
PCMA(G.711A)
Type: Audio
Maker: ITU-T
Required bandwidth: 64Kbps (90.4)
Features: PCMU and PCMA can provide better voice quality, but they occupy a higher bandwidth, requiring 64kbps.
Advantages: excellent voice quality
Disadvantages: higher bandwidth usage
Application field: voip
Royalty method: Free
Note: PCMU and PCMA can both 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 needs 8kbps). If the bandwidth is sufficient and better voice quality is needed, PCMU and PCMA can be used, or even wideband encoding method G722 (64kbps) can be used, which can provide high-fidelity sound quality.
ADPCM (Adaptive Differential PCM)
Type: Audio
Maker: 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 waveform code with better performance. Its core idea is:
① Use the adaptive idea to change the size of the quantization step, that is, use a small quantization step (step-size) to encode small differences, and use a large quantization step to encode large differences;
②Use past sample values to estimate the predicted value of the next input sample, so that the difference between the actual sample value and the predicted value is always the smallest.
Advantages: low algorithm complexity, low compression ratio (CD sound quality>400kbps), and the shortest codec delay (compared to other technologies)
Disadvantages: average sound quality
Application field: voip
Royalty method: Free
Note: 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 in 4bit, so The compression ratio is 1:4. The compression/decompression 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
Maker:
Required bandwidth: 2Kbps-4.8Kbps
Features: large compression ratio, large amount of calculation, low sound quality, low price
Advantages: large compression ratio, cheap
Disadvantages: large amount of calculation, poor voice quality, low naturalness
Application field: voip
Royalty method: Free
Remarks: Parameter coding is also called sound source coding, which extracts characteristic parameters from the source signal in the frequency domain or other orthogonal transform domains and transforms them into digital codes for transmission. Decoding is the reverse 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, the parameter coding is to extract and code the characteristic parameters of the speech signal to make the reconstructed speech signal as accurate as possible, but the waveform of the reconstructed signal may be quite different from the waveform of the original speech signal. Such as: linear predictive coding (LPC) and various other improved types are all parametric coding. The coding bit rate can be compressed to 2Kbit/s-4.8Kbit/s, or even lower, but the voice quality can only reach medium, especially low naturalness.
CELP (Code Excited Linear Prediction, Code Excited Linear Prediction)
Type: Audio
Maker: European Telecommunications Standards Institute (ETSI)
Required bandwidth: 4～16Kbps rate
Features: Improve the quality of voice:
① Perform perceptual weighting on the error signal, and use the masking characteristics of human hearing to improve the subjective quality of speech;
②Use score delay to improve pitch prediction to make voiced voice expression more accurate, especially to improve the quality of female voice;
③ Use the modified MSPE criterion to find the "best" delay, so that the shape of the pitch period delay is smoother;
④Adjust the size of the random excitation vector according to the efficiency of long-term prediction to improve the subjective quality of speech; ⑤ Using an adaptive smoother based on channel error rate estimation, it can also synthesize with a higher naturalness when the channel error rate is high. High voice.
in conclusion:
① The CELP algorithm can obtain satisfactory compression effects in a low-rate coding environment;
②Using a fast algorithm can effectively reduce the complexity of the CELP algorithm, making it fully implementable in real time;
③CELP can successfully encode various different types of speech signals. This adaptability is more important to the real environment, 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 voice coding standard Adaptive Multi-Rate Voice Encoder (AMR) based on Code Excited Linear Predictive Coding (CELP), with the lowest rate of 4.75kb/s , To achieve communication quality. CELP Code Excited Linear Prediction 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 channel parameters, uses a codebook containing many typical excitation vectors as excitation parameters, and searches for an optimal excitation vector in this codebook every time it is encoded. 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. The US Federal Standard FS1016 is the CELP coding method, which is mainly used for high-quality narrow-band voice secure 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 immunity to background noise. CELP is a speech compression coding scheme widely used at low and medium speeds.
G.711
Type: Audio
Maker: ITU-T
Required bandwidth: 64Kbps
Features: algorithm complexity is small, sound quality is average
Advantages: low algorithm complexity, low compression ratio (CD sound quality>400kbps), and the shortest codec delay (compared to other technologies)
Disadvantages: higher bandwidth usage
Application field: voip
Royalty method: Free
Remarks: G.711 64kb/s pulse code modulation PCM released by CCITT in the 1970s.
G.721
Type: Audio
Maker: 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 the mutual conversion between 64 kb/s A-law or μ-law PCM rate and 32 kb/s rate.
G.722
Type: Audio
Maker: 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: Subband ADPCM (SB-ADPCM) technology
G.723 (low bit rate speech coding algorithm)
Type: Audio
Maker: ITU-T
Required bandwidth: 5.3Kbps/6.3Kbps
Features: The voice quality is close to good, the bandwidth requirement is low, and it is implemented efficiently. It is convenient for multi-channel expansion. The C5402 on-chip 16kRAM can be used to implement 53coder. Reach the voice quality required by ITU-TG723, with stable performance. It can be used for IP phone voice source coding or high-efficiency voice compression storage.
Advantages: low bit rate and small bandwidth requirements. And achieve the voice quality required by ITU-TG723, and the performance is stable.
Disadvantages: average sound quality
Application field: voip
Royalty method: Free
Remarks: G.723 speech encoder is a dual-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 standards formulated by the International Telecommunication Union (ITU) and can be applied to systems such as IP telephones. Among them, the 5.3kbits/s code rate encoder uses multi-pulse maximum likelihood quantization technology (MP-MLQ), and the 6.3kbits/s code rate encoder uses algebraic code excitation linear prediction technology.
G.723.1 (dual 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 optimal for voice signals. G.723.1 uses silent compression that performs discontinuous transmission, which means that artificial noise is added to the bit stream during the silent period. In addition to reserved bandwidth, this technology allows the modem of the transmitter to keep working continuously and avoids the on-and-off of the carrier signal.
Advantages: low bit rate and small bandwidth requirements. It also reaches the voice quality required by ITU-TG723, with stable performance, and avoids the on-and-off of the carrier signal.
Disadvantages: average voice quality
Application field: voip
Royalty method: Free
Note: The G.723.1 algorithm is a compression algorithm recommended by ITU-T for voice or other audio signals in low-rate multimedia services. Its target application systems include multimedia communication systems such as H.323 and H.324. At present, this algorithm has become one of the required algorithms in the IP phone system.
G.728
Type: Audio
Maker: ITU-T
Required bandwidth: 16Kbps/8Kbps
Features: Used in many fields such as IP telephony, satellite communications, voice storage, etc. G.728 is a low-latency encoder, but it is more complicated than other encoders. This is because 50-order LPC analysis must be repeated in the encoder. 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 complicated 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, the ITU announced the G.728 8kb/s CS-ACELP algorithm, which can be used in many fields such as IP telephony, satellite communications, and voice storage. 16 kbps Gv.728 low-latency code excitation linear prediction.
G.728 is a hybrid of low-bit linear prediction synthesis analysis encoder (G.729 and G.723.1) and backward ADPCM encoder. G.728 is an LD-CELP encoder, it only processes 5 samples at a time. For low-rate (56~128 kbps) Integrated Services Digital Network (ISDN) videophones, G.728 is a recommended voice encoder. Due to its backward adaptive characteristics, G.728 is a low-latency encoder, but it is more complex than other encoders, because 50-order LPC analysis must be repeated in the encoder. G.728 also uses an adaptive post filter to improve its performance.
G.729
Type: Audio
Maker: ITU-T
Required bandwidth: 8Kbps
Features: To achieve long-distance quality under good channel conditions, and have good robustness in the case of random bit errors, frame loss, and multiple transfers. This voice compression algorithm can be used in a wide range of fields, including IP phones, wireless communications, digital satellite systems, and digital dedicated lines.
The G.729 algorithm uses the "Conjugate Structure Algebraic Codebook Excited Linear Prediction Coding Scheme" (CS-ACELP) algorithm. This algorithm combines the advantages of waveform coding and parameter coding, based on adaptive predictive coding technology, using vector quantization, synthesis analysis, and perceptual weighting techniques.
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 to the 5ms forward view, the point-to-point delay generated by G.729 is 25ms, the bit rate is 8 kbps.
Advantages: good voice quality, wide application areas, vector quantization, synthetic analysis and perceptual weighting are used, and a hidden processing mechanism for frame loss and packet loss is provided
Disadvantages: Poor performance in handling random bit errors.
Application field: voip
Royalty method: Free
Note: The International Telecommunication Union (ITU-T) officially adopted G.729 in November 1995. ITU-T recommendation G.729 is also called "Conjugate Structure Algebraic Code Book Excited Linear Prediction Coding Scheme" (CS-ACELP), which is currently a relatively new voice compression standard. G.729 was jointly developed by several well-known international telecommunications entities in the United States, France, Japan and Canada.
G.729A
Type: Audio
Maker: 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 provided a hidden processing mechanism for frame loss and packet loss
Disadvantages: performance is worse than G.729
Application field: voip
Royalty method: Free
Remarks: In 1996, ITU-T formulated G.729A, a simplified G.729 scheme, which mainly reduced the computational complexity for real-time implementation. Therefore, G.729A is currently used.
GIPS
Type: Audio
Maker: Global IP Sound, Sweden
Required bandwidth:
Features: GIPS technology can automatically adjust the encoding bit rate according to the bandwidth status, providing low bit rate and 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 voice call effect that is clearer than the phone.
Advantages: Solve the problem of voice delay and echo, bring perfect sound quality, and provide voice call effect that is clearer than the phone
Disadvantages: Not Free
Application field: voip
Royalties: pay an annual fee for the right to use
Remarks: GIPS audio technology is a voice compression engine system dedicated to the Internet 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 status, providing low bit rate and 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 clearer voice call effect than telephone.
Apt-X
Type: Audio
Maker: Audio Processing Technology Corporation
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 in the professional audio field to provide high-quality audio. Its characteristics are:
①Using 4:1:4 compression and amplification scheme;
②Low hardware complexity;
③ Very low coding delay;
④ Realized by a single chip;
⑤Mono or stereo coding and decoding;
⑥Two-channel stereo at 22.5kHz can be realized with a single device;
⑦ Up to 48kHz sampling frequency;
⑧Good fault tolerance;
⑨Complete AUTOSYNC™ codec synchronization scheme;
⑩Low power consumption
Advantages: high-quality audio, low hardware complexity, low equipment requirements
Disadvantage: Not Free
Application field: voip
Royalties: One-time payment
Remarks: Subband ADPCM (SB-ADPCM) technology
NICAM (Near Instantaneous Companded Audio Multiplex)
Type: Audio
Maker: BBC Broadcasting Corporation
Required bandwidth: 728Kbps
Features: The application range is extremely wide, and it can be used for stereo or bilingual broadcasting
Advantages: The application range is extremely wide, the signal-to-noise ratio is high, the dynamic range is wide, and the sound quality is comparable to CD, so it is called Nicam, so NICAM is also called Nicam
Disadvantages: not Free, high bandwidth requirements
Application field: voip
Royalties: One-time payment
Remarks: NICAM is also called Nicam, which is the abbreviation of Near-Instantaneously Companded Audio Multiplex in English. Its meaning is quasi-instantaneously companded audio multiplex, which was successfully developed and researched by the British BBC Broadcasting Corporation.
In layman's terms, NICAM technology is actually two-channel digital sound technology. Its application range is extremely wide. The most typical application is the addition of two-channel digital sound technology to TV broadcasting, which is used for stereo or bilingual broadcasting to make full use of TV The frequency spectrum resources of the channel. This can be achieved without a lot of additional investment on the basis of conventional television broadcasting. In stereo broadcasting, it improves the signal quality of audio to make it close to the quality of CD. Moreover, NICAM technology can be used for high-speed data broadcasting and other data transmission proliferation 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 audio cassettes, 2 channels, and 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. At the same time, the coding efficiency and sound quality are also greatly improved, and the coding delay is increased accordingly. Can achieve "completely transparent" sound quality (EBU sound quality standard)
Disadvantages: higher bandwidth requirements
Application field: voip
Royalty method: Free
Remarks: MPEG-1 audio compression coding is the first international standard for high-fidelity audio data compression. It is divided into three levels:
--Layer 1 (Layer 1): Simple coding, used for digital audio cassette tapes
--Layer 2 (Layer 2): The algorithm complexity is medium, used for digital audio broadcasting (DAB) and VCD, etc.
--Layer 3 (Layer 3): The coding is complex, used for the transmission of high-quality sound on the Internet, such as MP3 music compression 10 times
MUSICAM (MPEG-1 audio layer 2, namely MP2)
Type: Audio
Maker: MPEG
Required bandwidth: 256～192kbps (compressed 6～8 times)
Features: The algorithm complexity is medium, used for digital audio broadcasting (DAB) and VCD, etc., 2 channels, and MUSICAM due to its appropriate complexity and excellent sound quality, in the production of digital studios, DAB, DVB and other digital programs , Exchange, storage, and transmission are widely used.
Advantages: The compression method is much more complicated than the time-domain compression technology. At the same time, the coding efficiency and sound quality are also greatly improved, and the coding delay is increased accordingly. 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
Maker: MPEG
Required bandwidth: 128～112kbps (compressed 10～12 times)
Features: The coding is complex, used for the transmission of high-quality sound on the Internet, such as MP3 music compression 10 times, 2 channels. MP3 is a hybrid compression technology proposed on the basis of combining the advantages of MUSICAM and ASPEC. Under the current technical conditions, the complexity of MP3 is relatively high, and the encoding is not conducive to real-time, but because of the high level of MP3 under low bit rate conditions The sound quality makes it the darling of soft decompression and network broadcasting.
Advantages: high compression ratio, suitable for dissemination 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: the same as MPEG-1 layer 1, layer 2, and layer 3
Features: MPEG-2 sound compression coding uses the same codec as MPEG-1 sound, the structure of layer 1, layer 2 and layer 3 is also the same, 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 coding uses the same codec as MPEG-1 sound. The structure of layer 1, 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
Maker: MPEG
Required bandwidth: 96-128 kbps
Features: AAC can support any number of audio channel combinations from 1 to 48, including 15 low-frequency effect channels, dubbing/multi-voice channels, and 15 data. It can transmit 16 sets of programs at the same time, and the audio and data structure of each program can be arbitrarily specified.
The main possible applications of AAC are concentrated on Internet network transmission, digital audio broadcasting, including satellite live broadcast and digital AM, as well as digital TV and cinema systems. AAC uses a very flexible entropy coding core to transmit coded spectrum data. It has 48 main audio channels, 16 low-frequency enhancement channels, 16 integrated data streams, 16 dubbing, and 16 arrangements.
Advantages: supports multiple audio channel combinations, providing high-quality sound quality
Disadvantages:
Application field: voip
Royalty method: one-time fee
Note: AAC formed the international standard ISO 13818-7 in 1997. Advanced Audio Coding (AAC) was successfully developed and became a new generation 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. But later in order to meet the requirements of studio TV, it was defined as a multi-channel audio standard that can obtain higher quality. Of course, this standard is not compatible with MPEG-1, so it is called MPEG-2 AAC. In other words, on the surface, to make and play AAC, you need to use completely different tools from MP3.
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