This article comes from:
Http://blog.csdn.net/jtlyr/article/details/5321884
Here are some other websites that introduce wav files, which are recorded below:
Https://ccrma.stanford.edu/courses/422/projects/WaveFormat/
Http://blog.csdn.net/zhihu008/article/details/7854529
There are many methods for compressing PCM audio files into ADPCM files (such as ms acm and SOx). This article mainly introduces the public algorithm (as shown below, if you need to
Transferred from: http://blog.csdn.net/nogodoss/article/details/10399243
Http://blog.csdn.net/kindyb/archive/2005/10/13/503024.aspx
First, overview:
This paper describes how to convert from IMA-ADPCM file to PCM file by IMA-ADPCM compression and decompression algorithm. The main contents include: The internal structure of PCM and IMA-ADPCM wave files, IMA-
ADPCM compression AlgorithmThe ADPCM (Adaptive differential Pulse Code modulation) is a lossy compression algorithm for 16bits (or 8bits or higher) sound waveform data, which saves 16bit of data per sample in the sound stream at 4bit Storage, so the compression ratio is 1:4. and the compression/decompression algorithm is very simple, so it is a low-space consumption, high-quality and high-efficiency sound t
ADPCM (adaptive Differential Pulse Code Modulation) is a Lossy Compression Algorithm for 16 bits (or 8bits or higher) Sound waveform data, it stores the 16-bit data sampled in the audio stream in 4 bits, so the compression ratio is. in addition, the compression/decompression algorithm is very simple, so it is a good way to obtain high-quality and efficient sound with low space consumption. The data file with the suffix ". Aud" for storing sound is mos
indicates that the difference signal requires a small number of digits. An improvement solution for this common method is to use the N samples to predict the current sample according to certain rules, and then quantify the error between the predicted value and the actual value, based on the error signal, the original signal is restored using the same prediction method as the sending end. This is DPCM.
APCM (Adaptive pulse code modulation -- adaptive pulse Coding Modulation)Adaptive pulse coding
I was so happy to read the materials. The conversion class supports Vox conversion in 8/16-bit, 6000/8000 sampling rate, ADPCM, mulaw, and alaw formats.
It is strange that the sound formats recorded by the Huawei ICD platform are slightly different. The speech of a platform can be converted using our program, however, after the recordings on another platform are converted, the waveforms above the 0 axis are lost, but the sound is basically the same. I
adpcm (CRI), a Japanese software company.It seems to be a file that actually contains hundreds or even thousands of files this technology is widely used in DC, ps2, X-BOX and other game machine typesAfsexplorer can change the audio image in the game or even edit an AFS tool.
Practice!
Sakura War 1 does not need to be played directly by any WAV player, but although the music file is WAV, the sound played is terrible and I don't know how to deal wi
Brew supports two types of audio formats: midi-based file formats and special encoding audio file formats. Midi-based file formats include. mid ,. DLS .. xmf and. the DLS file is defined by the device and cannot be played by itself. The xmf file combines Midi and DLS for playback. Similar to xmf, the PMD file is a file container that can contain Midi, DLS, and audio in other encoding formats, as well as images, text, led, or Vibration Synchronization information. Encoding audio formats include q
Like videos, audio transmission often requires compression. The following describes the use of the PCM audio bare stream File compression and decoding library provided by hith, comparing the compressed and decoded data of a file, we can clearly find that the audio compression process of PCM> ADPCM is lossy. For file comparison, I am using the BCompare software, which is definitely a must-have for programmers! If you do not have any children's shoes, w
encoder uses multi-pulse Maximum Likelihood quantization (MP-MLQ), and the low-rate encoder uses the digital generator to stimulate the linear prediction (ACELP) method, both encoder and decoder must support these two rates and can convert the two rates between frames;This system can also compress and decompress music and other audio signals, but it is optimal for voice signals. It adopts the mute compression for discontinuous transmission, this means that artificial noise is added to the bit s
streaming applications;Royalty: freeNote:
G.722.1 is a set of Low-bit-rate and low-complexity broadband speech coding algorithms proposed by Polycom. It mainly uses the Transform Domain encoding method, so it can ~ 4000Hz) and less than 7 kHz music encoding, sampling rate is 16 kHz, its 32 Kbps bit rate reconstruction speech quality is equivalent to 64 Kbps g.722 Sb-ADPCM. In practical applications, the audio quality of this algorithm is higher than
Summary
In this chapter, you learned the sound effects in xNa. Whether you like it or not, you must use xact. Xact enables you to play sound effects and music on the Xbox 360 and Windows platforms.
The following are the key points you have learned in this chapter:
Xact is the only way to use xNa to play a sound on Xbox 360.
The xact tool may be annoying at first, but once you get used to it, you can enjoy new and cool features. For exampleCodeTo enable fade-in and fade-out, it is easier
wait for the sound file to be fully downloaded before it can begin playback, and it will play continuously until there is a clear stop command. Streaming Sound (Stream sounds) starts playing as soon as the data is downloaded from a few frames, and the playback of the sound is synchronized with the timeline that is played on the site.After we import the audio file, you can double-click the sound file in the library to bring up the Properties dialog box.You can choose to set the output properties
Flash Animation | compression
Add sound in Flash can greatly enrich the performance of the animation, but if we work hard to edit the sound can not be very good with the animation, or the sound file too large impact on the speed of flash, the effect will be greatly reduced. So how do you get the best results out of flash sounds? Let's explore the problem together.
The compression of sound files in Flash
When we import a flash file into a Web page, we have to consider the size of the Flash anim
ITUAudio analytics performance metrics
Zookeeper
Algorithm
Set Rate
Bit Rate
Sample Data
Time Delay
CBR
VBR
Physical Environment
Multi-channel
G.711
PCM
8 kHz
64 kbit/s
8 bit
125 μs
Yes
No
No
No
G.711.1
Mdct, A-Law, U-Law
8 K or 16 kHz
64, 80, 96 kbit/s
16 Bit
11.875 MS
Yes
Yes
No
No
G.718
CELP, mdct
8 or 16 kHz
8, 12, 12.65, 16, 24, 32 kbit/s
16 Bit
42.875 MS-43.87
Compression
Add sound in Flash can greatly enrich the performance of the animation, but if we work hard to edit the sound can not be very good with the animation, or the sound file too large impact on the speed of flash, the effect will be greatly reduced. So how do you get the best results out of flash sounds? Let's explore the problem together.
The compression of sound files in Flash
When we import a flash file into a Web page, we have to consider the size of the Flash animation, especially
.
Figure 5: You can set the global compression when exporting the SWF file.
Keep in mind that the sound effects you make are likely to be played through tiny, single frequency speakers, so the sound quality is limited by the speakers. Ideally, you'd better use the actual device for testing. However, I found ADPCM, 11kHz, 4 bit, Mono such a set of global export settings can be a good result in most cases. For tiny speakers, it balances the sound quali
figure:
You can choose to set the output properties of the audio footage in the Exprotsettings column. There are three types of audio compression available: ADPCM (adaptive Audio pulse encoding), MP3, and Raw (uncompressed).The following parameter options can be set in detail for the output audio effect.
When selecting ADPCM:
Perprocessing: Select whether to merge into mono
Samplerate: Output sam
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