ACC is a technology provided by Dolby Laboratories for the music community and is a high compression ratio encoding algorithm. The actual experience is that the same bit rate, ACC's listening is better than MP3, and there is a lot of ACC audio on the apple. So now the audio transmission format of iphone is also acc format, the code rate is equivalent to SBC, but the sense of hearing is said to be better than SBC. Because I have not done a serious comparison, I think it is slightly better than SBC. AAC claims to be "a maximum of 48 channels of audio, with a sampling rate of 96 KHz, and provides the equivalent of ITU-R broadcast quality for 5.1-channel music programs at a data rate of 320 Kbps." Compared with MP3, its sound quality is better, and it can save about 30% of storage space and bandwidth. It is a technology developed in accordance with the specifications of MPEG-2. Panasonic's mp3 products use this encoding, of course, also compatible with mp3 format, it can be said that aac is a very easy to use audio format, 128kbps aac enough to compete with 224kbps mp3, space is almost half, but in The styles of aac and mp3 are not the same in space and structure. There are three main extensions to AAC encoding: .AAC - Uses MPEG-2 Audio Transport Stream (ADTS, see MPEG-2) container, which is different from MP4/M4A format using MPEG-4 container, belongs to traditional AAC encoding (FAAC default package, but FAAC can also output MPEG -4 packaged AAC) .MP4 - AAC encoding using the simplified version of MPEG-4 Part 14 (Part 14), 3GPP Media Release 6 Basic (3gp6, see 3GP) (Nero AAC encoder can only output AAC in MPEG-4 package) ; .M4A - Apple iTunes uses the ".M4A" name for pure audio MP4 files in order to distinguish between pure audio MP4 files and MP4 files containing video. The essence of M4A is the same as that of audio MP4, so the audio MP4 file can also be directly changed with the extension M4A. The earliest format should be SBC, which is the encoding format mandated by the A2DP (Advanced Audio DistribuTIon Profile) protocol. All Bluetooth will support this protocol, so all Bluetooth audio chips will also support this protocol. The specific parameters of the SBC code rate during transmission are not found. According to the information given by Sony's official website, it is: 328Kbps, 44.1KHZ. This bit rate is actually similar to the high quality MP3. However, because the Bluetooth transmission intermediate device needs to be transcoded, taking MP3 files as an example, the transcoding process is MP3-"PCM-"SBC-"PCM, each time transcoding will lose details, resulting in SBC hearing will be better than the original MP3 To be bad. The input signal is first divided into several sub-band signals in different frequency bands by a set of band pass filters, and then these signals are converted into baseband signals by frequency shifting, and then respectively sampled. The sampled signal is quantized, encoded, and combined into a total code stream for transmission to the receiving end. At the receiving end, the code stream is first divided into sub-band code streams corresponding to the original sub-band signals, and then decoded, the spectrum is moved to the original position, and finally the band-pass filtered and added to obtain the reconstructed signal. Advantages: It can utilize the characteristics of different sensitivity of human ear (or human eye) to different frequency signals, adopt coarser quantization in human auditory (or visual) insensitive parts, and adopt finer quantization in sensitive parts to Get a better subjective auditory (visual) effect. For example, the pitch and formants of speech are mainly concentrated in the low frequency band, so more bits can be allocated to represent their samples; and the high frequency band where fricative and similar friction noise appear can be allocated fewer bits, so that it can be fully compressed Voice data. The quantization noise of each sub-band is bound in the sub-band, so that the signal in the frequency band with less energy can be prevented from being covered by the quantization noise in other frequency bands. The specific implementation of the filter cannot be an ideal bandpass, and its amplitude effect inevitably has a limited roll-off. Therefore, when dividing the sub-bands, only the sub-bands may overlap or have a certain gap between the sub-bands. If the former is sampled at the Nessler frequency, aliasing distortion will occur, and the latter will cause the original partial frequency band to be filtered and lost, and the reconstructed signal will be distorted. Solutions to this problem are quadrature mirror filtering and time domain aliasing. APTX is CSR's patented encoding algorithm. After being acquired by Qualcomm, APTX has been widely promoted in Android phones. According to the official website, aptX is divided into three types: aptX, aptX HD and aptX Low Latency. According to the name, it can be considered as traditional aptX, high quality aptX (estimated to improve bit rate) and low time delay aptX (in watching video and playing Time delay is very important when CS is). The Qualcomm® aptXTM audio coding algorithm originated in the late 1980s at Queen's University Belfast. This innovaTIve work was focused on bit rate reducTIon and achieved significant bit rate efficiencies while preserving audio quality. aptX has been the best kept secret of the professional audio industry, used by major public broadcasters and film studios around the world. Now available on leading consumer devices, aptX enables music lovers to enjoy the rich listening experience that aptX delivers. Qualcomm probably means that aptX is my home, and then the audio coding algorithm that Queen's University began to develop in the 1980s. This algorithm has high bit rate efficiency while maintaining high audio quality. As a professional wireless audio transmission solution, aptX is used in public broadcasting and movie audio, and now it is finally placed in consumer products. Therefore, aptX is not estimated to have a high transmission rate, which may be similar to the previous two. However, thanks to efficient coding, the details of sound retention are more, and the actual listening is better than the previous two. The promotion of aptX is also called Achieve a CD level of listening. Apt-X is a digital audio compression algorithm based on subband ADPCM (SB-ADPCM) technology. The original algorithm was proposed by Dr. Stephen Smyth in the 1980s. Developed by Audio Processing Technology (now merged with CSR) and named apt-X. Originally used in the professional audio and broadcast field. In recent years, apt-x in the field of Bluetooth wireless audio transmission has replaced the SBC (Sub-band Coding) due to its low latency, good fault tolerance and high sound quality. Currently available in the apt-x family are aptX Bluetooth, aptX Enhanced, aptX Live (launched in 2007) and aptX Lossless (launched in 2009). apt-X has the following features: 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) The use of Apt-X is mainly focused on Bluetooth headsets and Bluetooth speakers, and both the terminal and the Bluetooth headset must support Apt-X to perform its functions. There is a delay in Bluetooth audio transmission. The biggest feeling is that the audio and video delay can be minimized. Now it is the turn of Dafa to appear. Dafa has simply and rudely improved the channel. On devices that support LDAC, the communication rate of Bluetooth is close to 1M. LDAC can transmit about 3 times the data of normal Bluetooth*1 (*2 at the maximum transmission speed of 990kbps), allowing you to listen to Hi-Res Audio when you enjoy Hi-Res Audio*3 music in wireless situations. Sound quality. At such high transmission speeds, it is possible to transmit lossless music. Of course, the transport format of this near-private protocol also leads to the compatibility of only a few devices. But there is no doubt that LDAC has greatly improved the transmission rate, making the transmitted audio quality higher, and the listening experience is naturally the best. The next issue will cover the current situation of various Bluetooth audio SOC vendors and support for each format. Then the development board of CSR8670 and CSRA64215 will be broadcast live and actual development. LDAC is a wireless audio coding technology developed by Sony. It was first unveiled at the 2015 CES Consumer Electronics Show. At the time, Sony said that LDAC technology was three times more efficient than standard Bluetooth encoding and compression systems. In this way, those high-resolution audio files will not be over-compressed when wirelessly transmitted, which will greatly degrade the sound quality. The reason why Sony's decision is quite interesting is that LDAC technology used to be almost "Sony-only" in the past, only on its own mobile phones, players, headphones, Bluetooth speakers. Because LDAC technology relies on the simultaneous support of audio and speaker systems, this makes the technology a unique highlight of Sony devices. Nowadays, as long as it is a mobile phone with Android O installed, you can use LDAC to let users enjoy wireless HD music on the playback system that supports this technology.
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ACC (Advanced Audio Coding)