WAV

Description

WAV files on the PC platform is very common, the most classic of multimedia audio files, as early as August 1991 appeared in Windows3. 1 on the operating system, file extension WAV, waveForm is shorthand, also known as wave files, sound waveform may be stored directly, reducing the wave curve is very realistic. WAV file format referred to WAV format is a format for storing digital audio sound waveform by Microsoft and IBM jointly designed, after several revisions, can be used for Windows, Macintosh, Linux and other operating systems. WAV supports a variety of digital audio sampling frequency and channel, as standard WAV file format and CD format, sampling frequency is 44.1kHz, 16-bit quantization digital, so the sound quality and file CD is almost the same. WAV following features: recording real natural sound waveform, substantially no data compression, data volume.

In general, the WAV file obtained by reducing the sound quality of the sound card sampling depends on the size of the sample, the higher the sampling rate, the better the sound quality, but the greater the cost, WAV files the larger,

storage

sound waves from the sound source by a microphone is converted into an electrical signal continuously changes, after amplification, the anti-aliasing filter, sampled at a fixed frequency , signal amplitude from the value of each sample is detected in a sampling period; integrated value of the next binary number which is represented by an analog electrical signal by the quantization; and stored as the last encoded audio stream data. Some applications in order to save storage space, before storage, but also on the sampling data to be compressed.

File Structure

In the Windows environment, the majority of multimedia files are in accordance with the resource Interchange File Format (ResourceslnterchangeFileFormat) to store information, referred to RIFF format. RIFF file constituting the basic unit is called a block (chunk). Each RIFF file is made up of several blocks. Each block (the chunk) consists of three parts identification block, the data block length and composed.

wherein the block is the block name stored identification of four ASCII characters thereof. Dissatisfied with four characters are filled with spaces to the right. Block length field,

4 bytes, stores lengths of the current data block, the block does not include identification and block length field. Therefore, the actual length value within a block is the block length field plus 8. RIFF predetermined format, and only the RIFF LIST block may contain sub-blocks, other blocks may not contain sub-blocks. RIFF format a document is itself a block. Which is the first 4 bytes of identification documents "RIFF", also RIFF block identifier, indicating the document is valid a RIFF file; the second part length of the file data, 4 bytes, a value of the file length -8; RIFF third part block data, wherein, the first four bytes of the file format type identifier, such as: "WAVE", "AVI", etc., other RIFF later blocks into sub-blocks.

WAV format RIFF file structure is employed. It is composed of at least three blocks, namely, RIFF, fmt and Data. All WAV files based compression encoding block must contain fact. In addition, all other blocks are optional. Block mt, Data and fact are sub-blocks RIFF block. Type of file format WAV file identifier "WAVE". The basic structure as shown in Table 1.

encoding

coding include two aspects, first, according to the data stored in a certain format, the second is the use of certain data compression algorithm. WAV format encoded audio stream no hard and fast rules, supports uncompressed PCM (PulsCodeModulation) pulse code modulation format, also supports the compression type of Microsoft Adaptive Differential Pulse Code Modulation MicrosoftADPCM (AdaptiveDifferentialPulsCodeModulation), the International Telegraph Union (InternationalTelegraphUnion) to develop speech compression standard ITUG.711a-law, ITUG.711-law, IMAADPCM, ITUG.723ADPCM (Yamaha), GSM6.10, ITUG.721ADPCM coding and other compression algorithms. MP3 encoding can also be used in WAV, simply install the appropriate Decode, you can play MP3 music in WAV.

PCM coding is directly stored acoustic sample data quantized uncompressed generated, it is considered pure lossless encoding format, the advantage of high-quality audio signal is obtained. WAV format PCM based WAV format is the most basic, the sound card is directly supported, capable of storing the sampled audio data directly, the stored data directly playing through the sound card, the reduction wave curve very close to the original speech waveform, sound playback quality is the first class, in the Windows platform is the best supported, is often used as an intermediate between other files encoded file conversion. PCM disadvantage is that the file size is too large, not suitable for long-time recording. Because of this, there was more developed and improved encoding formats, such as PCM encoded on the basis of: DPCM, ADPCM coding.

factors

sampling frequency : number of times per second to capture audio data. The higher the sampling frequency, the higher audio fidelity. 16-bit wide computer sound card configuration, generally includes the use of a sampling frequency 11025Hz, 22050Hz, 44100Hz and 48000Hz four kinds, wherein the sample employed 11025Hz sound effects corresponding to effect sound telephone; 22050HZ using sampled sound effects corresponding to FM radio effect; the use of 44100HZ sampled sound effects equivalent to CD sound effect.

sample bits (amplitude sampling precision): the sampling values ​​or sampled value is used to measure a parameter fluctuating sound, the sound card is the resolution. The greater its value, the higher the resolution, the ability of sound. Currently 16-bit sound samples computer configuration bits comprises two kinds of 8-bit and 16-bit.

Number of channels : there are mono and stereo points, using only monophonic sound of a speaker sound (sound card also some information into a monophonic two speakers simultaneously output), allows two stereo speakers are WAV sound (left and right channels have generally division), so that the space can feel the effect of audio information. Obviously, the two-channel data reduction characteristics closer to the people listening habits, but the amount of data collected will get doubled.

In general, a sample scale samples (information amount), the higher the sampling rate, the better the sound quality, but the greater the cost of the audio waveform. Since speech typically to eight samples can better 11.025KHz reduction, therefore, the audio waveform is generally applicable to the following situations:

1, the speech sound playback voice, sound effects on the quality of the sound less demanding applications.

2, the sound and other data to be loaded from the CD-ROM disc drive while transmitting audio data where the processing time is not exclusive.

3, need to be stored in the PC hard drive in an audio data points or less, and when a sufficient available storage space.