Video compression methods
The rate of the frames representation, in frames per second.
By Default value is: 23.97 frame rate
PCM - Standard Windows WAV format for non-compressed audio files. Pulse Code Modulation (PCM) is the standard method of digitally encoding audio. It is the basic uncompressed data format used in file types such as Windows .wav.
ALAW - Compressed WAV format. A-Law (or CCITT standard G.711) is an audio compression scheme common in telephony applications. It is a slight variation of the u-Law compression format, and is found in European systems. This encoding format compresses original 16-bit audio down to 8 bits (for a 2:1 compression ratio) with a dynamic range of about 13-bits. Thus, a-law encoded waveforms have a higher s/n ratio than 8-bit PCM, but at the price of a bit more distortion than the original 16-bit audio. The quality is higher than you would get with 4-bit ADPCM formats. Encoding and decoding is rather fast and generally, widely supported.
MP3 - MPEG Layer-3 format. Very popular format for keeping of music.
The mp3 algorithm development started in 1987, with a joint cooperation of Fraunhofer iis-a and the university of erlangen. it is standardized as iso-mpeg audio layer 3. it soon became the de facto standard for lossy audio encoding, due to the high compression rates (1/12 of the original size, still remaining considerable quality), the high availability of decoders and the low cpu requirements for playback. (486 dx2-66 is enough for real-time decoding). it supports multichannel files (although there's no implementation yet), sampling frequencies from 16khz to 24khz (mpeg2 layer 3) and 32khz to 48khz (mpeg1 layer 3). formal and informal listening tests have shown that mp3 at the 192-256 kbps range provide encoded results undistinguishable from the original materials in most of the cases.
mp3 uses the following for compression:
- huffman coding;
- m/s matrixing;
- intensity stereo;
- channel coupling;
- modified discrete cosine transform (mdct);
- polyphase filter bank.
Compression ratio is 1:10...1:12 corresponds to 128..112 kbps for a stereo signal.
MPEG Version 2.5 was added lately to the MPEG 2 standard. It is an extension used for very low bitrate files, allowing the use of lower sampling frequencies. If your decoder does not support this extension, it is recommended for you to use 12 bits for synchronization instead of 11 bits.
ULAW - Compressed WAV format. u-Law (or CCITT standard G.711) is an audio compression scheme and international standard in telephony applications. u-Law is very similar to A-Law, a variation of u-Law found in European systems. This encoding format compresses original 16-bit audio down to 8 bits (for a 2:1 compression ratio) with a dynamic range of about 13-bits. Thus, u-Law encoded waveforms have a higher s/n ratio than 8-bit PCM, but at the price of a bit more distortion than the original 16-bit audio. The quality is higher than you would get with 4-bit ADPCM formats. Encoding and decoding is rather fast and generally, widely supported.
ADPCM - Compressed WAV format. ADPCM (Adaptive Differential Pulse Code Modulation) is an audio compression scheme which compresses from 16-bit to 4-bit for a 4:1 compression ratio.
ADPCM stands for Adaptive Differential Pulse Code Modulation. ADPCM is a lossy compression mechanism. There are various flavors of ADPCM. This particular algorithm was suggested by Microsoft; its quality is similar to IMA (Interactive Multimedia Association) ADPCM. MS ADPCM compresses data recorded at various sampling rates. Sound is encoded as a succession of 4-bit nibbles. Each nibble represents the difference between the current sampled signal value and the previous value. The compression ratio obtained is relatively modest: 16-bit data samples encoded as 4-bit differences result in 4:1 compression format.
Microsoft ADPCM is directly supported on most Windows implementations as a native format. Although the quality of IMA ADPCM voice files is not great, the files are portable. There is a real advantage in having compact files that can be played on most Windows PCs.
GSM - Compressed WAV format. Good for keeping of human speech. It is lossy speech compression that allow to get telephone quality speech with 13 kbit/s. It is a standard used for telephone sound compression in European countries and its gaining popularity because of its quality.
GSM 06.10 stands for Global System for Mobile Communications and is a variant of LPC called RPE-LPC (Regular Pulse Excited - Linear Predictive Coder) and is a European standard originally for use in encoding speech for satellite distribution to mobile phones. It can be found in use in various telephony products such as voice mail applications.
It compresses 160 13-bit samples into 260 bits (or 33 bytes), i.e. 1650 bytes/sec (at 8000 samples/sec). It results in very good compression with good quality output but is very costly in terms of performance.
Mono - Mono data uses one channel.
Stereo - Stereo data uses two channels
Frequency - Number of sampling per second