You should check out your Windows Master Volume (or Volume Control) settings. For that go to Volume Control ->Options ->Properties->Recording and make sure that you've checked Microphone Select check-box
Also check out Volume Control ->Options ->Properties->Playback and choose the device you are going to use for listening to you background music, for example, CD Player or Line In. Click OK and make sure you haven't check Mute check-boxes.

You can set it as 44100 Hz as set by default - your output file will be CD-quality. Actually the higher frequency - the higher the quality and the larger size of your output file. The sampling frequency is essentially the number of times the sound event is quantized within a given time period. Sampling frequencies are specified in KiloHertz (KHz), a term meaning samples per second. For example, "CD-quality" sound requires 16-bit words sampled at 44.1 KHz. Essentially this means 44,100 16-bit words (705,600 bits) are used to digitally describe each second of sound on a compact disc. The highest pitch possible is 22.05 KHz (approximately the top of human hearing range), which is half of 44.1 KHz.

Uncompressed WAV PCM
PCM, Pulse Code Modulation (developed in 1939), is a standard method for digitizing analog audio signals. This format is an uncompressed, raw bit stream, linear (transmitted in a linear series meaning that the stream of the signal is sequential rather than random and the amplitude of both the of the input signal and output signal remain at a fixed ratio and a sinusoidal wave input signal will result in a sinusoidal wave output signal at the same frequency), signed two's complement, fixed point encoded data file. A PCM encoder (ADC) may sample analog sound from 8,000 times per second (8 KHz) and use 8-bits to represent each sample, is usually utilized at 44.1 KHz and 16-bit resolution to match CD Audio standards, and can encode at 96 KHz and 24-bit (approximate). The procedure uses only two alternating pulse values (1 and 0) duplicating binary code. This codec creates a raw (RAW) data file (no header or footer information describing sample rate, sample resolution, monaural or stereo) and gives us only 256 possible amplitude values (based on 8-bit binary numbering). When the codec is set to sample at the Audio CD level at 44.1 KHZ sample rate (44,100 samples per second), with 16-bit resolution per sample (65,536 possible amplitude levels), this results in a file that requires 1,411 Kbps (kilobits per second, or 1.4MB) bit rate for representation / playback of one second of stereo music.

Compressed WAV
ADPCM, Adaptive Differential Pulse Code Modulation developed by Microsoft and IBM, produces a high-quality sound than found in the WAV format and a compression ratio of 4 to 1. This is a conversion of a PCM bit format that, similar to DPCM, attempts to predict the value of each successive sample during the encoding process. It also utilizes a variable bit rate procedure to reduce the difference between sample amplitude levels. Thus, it very efficient and reduces file size by encoding the difference between successive samples rather than expending all encoding bit resources on reproducing the sample. There is both a Microsoft ADPCM and an IMA (Interactive Multimedia Association) ADPCM (used in Sony MiniDisc in tandem with ATRAC). One can use IMA ADPCM in the WAV, AIFF and SND formats.

U-law, A-law Compression is a common lossy compression scheme, similar to ADPCM, which can be used on AU, AIFF and WAV files. It is an international standard for compressing voice quality audio. It has a compression ration of 2:1. The G.721, G.723-24 and G.723-40 ADPCM formats are CCITT standards for compression of 8000-Hz 14-bit samples into a 32-, 24- or 40-kbps data stream. These compressed formats have extremely slow decompression rates. Because it is optimized for speech, in the United States it is a standard compression technique for telephone systems (in Europe, a-law is used). On the Internet it is used for ".au" file formats, alternately know as "Sun audio" or "NeXT" format.

GSM is the international standard digital mobile telephony encoding format. It uses linear predictive coding to substantially compress the data by predicting the likely shape of the sound wave and recording the differences between the actual sound and the prediction. Compression and decompression are slow and the quality is not great, but the algorithm is freely available resulting in widespread use in products.

MPEG 2 Layer2 (MP2), MPEG 2 Layer3 (MP3)
MP3 was introduced as a part of the official MPEG-1 standard in 1992 and until today it is the most successful audio-standard since WAV. The German Fraunhofer Gesellschaft (FhG), which has developed this audio-compression still holds the key patents the MP3-techology inherits. The development started back in 1987 at the Fraunhofer Institut Integrierte Schaltungen as project EUREKA EU147. The final compression algorithm became later known as MP3. In April 1989 Fraunhofer applied patent on MP3 in Germany and it became part of the MPEG-1 standard in 1992. It was in January 1995 when Fraunhofer applied patent on MP3 in America as well and it was granted in November 1996. Using MP3-compression PC-users were able to compress an ordinary music-CD to one tenth of it's original size - thus 12 hours of music could be stored on a recordable CD that on the other hand could be played by a MP3-CD-player or an ordinary PC. What made MP3 that popular in the end was the online peer-to-peer program named Napster. Millions of songs were exchanged every day via the popular program. That was solely possible by MP3, because conventional formats such as WAV or AU were way to big in size with similar quality. MP3 also offered like WMA later the big advantage of being streamable (not all of the file has to be downloaded to listen to it).

WMA
Microsoft's respond to MP3, the Windows Media Audio-standard. As it is part of the Windows Media package, Windows Media Audio 8 was presented in early December 2000 and it is until now the best Windows Media product.
Windows Media Audio among other things is firmly integrated in Microsoft's Windows Media Player.
Microsoft promises with this version almost CD-quality with just a third of the source-file's size.
Above all WMA offers the advantage that copyright-protected songs cannot be published any further (Digital Rights Management). That's not the only reason why many music- and movie-corporations meanwhile decided in favour of WMA instead of MP3. Like MP3 WMA is almost predestined for the internet by offering streaming capabilities (see MP3 for details) both with WMA and WMV (Windows Media Video).

Ogg Vorbis
The development of the OGG standard began in 1993, then known as "Squish". OGG was right from the start an open source project and hence is free of any patents. It was designed as a substitute for MP3 and WMA and by now it is almost as popular and well known as MP3. Above all, the algorithm is still being developed what is mainly due to its flexibility. Although the sound-quality gets better with every further development the files are backwards compatible and can be played with older players as well. Like MP3 OGG offers encoding at variable bitrates. Using this compression parts of the song are encoded with a higher compression than others what depends on the source. Most times, this compression goes along with squishy noises or even small interruptions. OGG is also one of the very few formats that support multi-channel compression. Surround-files could theoretically be compressed with more than two channels. OGG is, like it's predecessors, streamable and although the used player has to support this feature, it's one of many good reasons for OGG.

VOX (Dialogic PCM)
Dialogic ADPCM format. The Dialogic ADPCM format is commonly found in telephony applications, and has been optimized for low sample rate voice. It will only save mono 16-bit audio, and like other ADPCM formats, it compresses to 4-bits/sample (for a 4:1 ratio). This format has no header, so any file format with the extension .VOX will be assumed to be in this format.

Sampling frequency also impacts fidelity. The sampling frequency is essentially the number of times the sound event is quantized within a given time period. Sampling frequencies are specified in KiloHertz (KHz), a term meaning samples per second. The key is understanding how sampling frequency affects fidelity is the Nyquist sampling theorem. Basically, when applied to audio signals the Nyquist theorem states that the highest possible pitch in the sound is one-half that of the sampling frequency.
For example, "CD-quality" sound requires 16-bit words sampled at 44.1 KHz. Essentially this means 44,100 16-bit words (705,600 bits) are used to digitally describe each second of sound on a compact disc. The highest pitch possible is 22.05 KHz (approximately the top of human hearing range), which is half of 44.1 KHz.

To change a skin for any of the AVS Audio Tools:

Step 1. Run the program you want to change the skin for (Open the AVS Audio Tools Manager and select an action from the list, or use Start menu).
Step 2. In the upper right-hand corner of the program window click on a small arrow

 or the letter S in some skins   and select a skin from a fall-out menu.

Please note, that if you change the skin for 1 tool, it will not change the skin for other tools, thus you can have different skins for different tools. NOTE! Skins do not apply to Music Mix tool. Skin variants:

Flora

Sandy

Sunlight