Search results
Results from the WOW.Com Content Network
FIPS 137, originally issued as FED-STD-1015, is a secure telephony speech encoding standard for Linear Predictive Coding vocoder developed by the United States Department of Defense and finished on November 28, 1984. [1] It was based on the earlier STANAG 4198 [2] promulgated by NATO on February 13, 1984.
Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders (e.g., FS-1015).
Speech coding differs from other forms of audio coding in that speech is a simpler signal than other audio signals, and statistical information is available about the properties of speech. As a result, some auditory information that is relevant in general audio coding can be unnecessary in the speech coding context.
Linear predictive coding (LPC) is a method used mostly in audio signal processing and speech processing for representing the spectral envelope of a digital signal of speech in compressed form, using the information of a linear predictive model. [1] [2] LPC is the most widely used method in speech coding and speech synthesis.
In the process of encoding, the sender (i.e. encoder) uses verbal (e.g. words, signs, images, video) and non-verbal (e.g. body language, hand gestures, face expressions) symbols for which he or she believes the receiver (that is, the decoder) will understand. The symbols can be words and numbers, images, face expressions, signals and/or actions.
Deep learning speech synthesis refers to the application of deep learning models to generate natural-sounding human speech from written text (text-to-speech) or spectrum . Deep neural networks are trained using large amounts of recorded speech and, in the case of a text-to-speech system, the associated labels and/or input text.
The speech encoder accepts 13 bit linear PCM at an 8 kHz sample rate. This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table. In GSM, the encoded speech is passed to the channel encoder specified in GSM 05.03.
The Open Speech Repository [4] provides some freely usable, prerecorded WAV files of Harvard Sentences in American and British English, in male and female voices. Harvard lines are also used to observe how an actor's mouth can move when they are talking. This can be used when creating more realistic CGI models. [1]