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Sound Normalizer 3.3 Final



Document processing: The following are the six major processing steps undertaken by a synthesis processor to convert marked-up text input into automatically generated voice output. The markup language is designed to be sufficiently rich so as to allow control over each of the steps described below so that the document author (human or machine) can control the final voice output. Although each step below is divided into "markup support" and "non-markup behavior", actual behavior is usually a mix of the two and varies depending on the tag. The processor has the ultimate authority to ensure that what it produces is pronounceable (and ideally intelligible). In general the markup provides a way for the author to make prosodic and other information available to the processor, typically information the processor would be unable to acquire on its own. It is then up to the processor to determine whether and in what way to use the information.




Sound Normalizer 3.3 Final



Text-to-phoneme conversion: Once the synthesis processor has determined the set of tokens to be spoken, it must derive pronunciations for each token. Pronunciations may be conveniently described as sequences of phonemes, which are units of sound in a language that serve to distinguish one word from another. Each language (and sometimes each national or dialect variant of a language) has a specific phoneme set: e.g., most US English dialects have around 45 phonemes, Hawai'ian has between 12 and 18 (depending on who you ask), and some languages have more than 100! This conversion is made complex by a number of issues. One issue is that there are differences between written and spoken forms of a language, and these differences can lead to indeterminacy or ambiguity in the pronunciation of written words. For example, compared with their spoken form, words in Hebrew and Arabic are usually written with no vowels, or only a few vowels specified. In many languages the same written word may have many spoken forms. For example, in English, "read" may be spoken as "reed" (I will read the book) or "red" (I have read the book). Both human speakers and synthesis processors can pronounce these words correctly in context but may have difficulty without context (see "Non-markup behavior" below). Another issue is the handling of words with non-standard spellings or pronunciations. For example, an English synthesis processor will often have trouble determining how to speak some non-English-origin names, e.g. "Caius College" (pronounced "keys college") and President Tito (pronounced "sutto"), the president of the Republic of Kiribati (pronounced "kiribass").


Prosody analysis: Prosody is the set of features of speech output that includes the pitch (also called intonation or melody), the timing (or rhythm), the pausing, the speaking rate, the emphasis on words and many other features. Producing human-like prosody is important for making speech sound natural and for correctly conveying the meaning of spoken language.


Markup support: The voice element allows the document creator to request a particular voice or specific voice qualities (e.g. a young male voice). The audio element allows for insertion of recorded audio data into the output stream, with optional control over the duration, sound level and playback speed of the recording. Rendering can be restricted to a subset of the document by using the trimming attributes on the speak element.


Non-markup behavior: The default volume/sound level, speed, and pitch/frequency of both voices and recorded audio in the document are that of the unmodified waveforms, whether they be voices or recordings.


This element is designed strictly for phonemic and phonetic notations and is intended to be used to provide pronunciations for words or very short phrases. The phonemic/phonetic string does not undergo text normalization and is not treated as a token for lookup in the lexicon (see Section 3.1.5), while values in say-as and sub may undergo both. Briefly, phonemic strings consist of phonemes, language-dependent speech units that characterize linguistically significant differences in the language; loosely, phonemes represent all the sounds needed to distinguish one word from another in a given language. On the other hand, phonetic strings consist of phones, speech units that characterize the manner (puff of air, click, vocalized, etc.) and place (front, middle, back, etc.) of articulation within the human vocal tract and are thus independent of language; phones represent realized distinctions in human speech production.


The alphabet attribute is an OPTIONAL attribute that specifies the phonemic/phonetic pronunciation alphabet. A pronunciation alphabet in this context refers to a collection of symbols to represent the sounds of one or more human languages. The only valid values for this attribute are "ipa" (see the next paragraph), values defined in the Pronunciation Alphabet Registry and vendor-defined strings of the form "x-organization" or "x-organization-alphabet". For example, the Japan Electronics and Information Technology Industries Association [JEITA] might wish to encourage the use of an alphabet such as "x-JEITA" or "x-JEITA-IT-4002" for their phoneme alphabet [JEIDAALPHABET].


Note that the behavior of this attribute for label values may differ from that of numerical values. Use of a numerical value causes direct modification of the waveform, while use of a label value may result in prosodic modifications that more accurately reflect how a human being would increase or decrease the perceived loudness of his speech, e.g., adjusting frequency and power differently for different sound units.


The soundLevel attribute specifies the relative volume of the referenced audio. It is inspired by the similarly-named attribute in SMIL [SMIL3]. Synthesis processor support for this attribute is REQUIRED in the Extended profile.


When used on a complete file, then the result is exact. But the analysis of a whole file can take some time. Therefore, when used with the option dynamic volume adjustment, then the first 6 seconds of a sound file are analyzed on playback to estimate the initial adjustment factor. This takes only a few milliseconds. The analysis continues while the file is being played back and the initial estimation becomes more and more precise and as such the volume adjustment improves.


It can be of help if you're using voice activation instead of push-to-talk in your voice applications. Sometimes sounds may start off a little too quietly, so that the voice activation threshold is met too late and the beginning of the sound is not transmitted to your interlocutors.


This option plays another very short, but properly loud sound before playing the actual sound and triggers the voice activation beforehand. You can also exchange the provided sound with one of your own sounds.


The Input Levels is a normal window that shows the current levels at the sound input, and lets you control the inputgain (amplification). Not all audio hardware supports input gain. You can show and hide the Input Levels by using theWindow menu. The lower left-hand corner of document window also shows the input and output level meters.


The sample size will affect the sound quality and file size of the file on disk. A smaller sample size (8-bit beingthe smallest) will result in a smaller file size, but also lower sound quality. The 16-bit setting will produce a filetwice the size of an 8-bit file, but with significantly better sound quality, while the 24-bit setting will produce afile three times the size of an 8-bit file. The 16-bit settings is usually the best compromise between file size andsound quality.


Expanding is useful when you want to increase the dynamic range of the audio. It is also useful if you have a noisyrecording and want to reduce the volume of the quieter passages so you don't notice the noise as much. It does have theside effect of changing the way sounds decay and can end up silencing some parts that are quieter.


If you consider waves that go above the centerline of the display to be positive and waves that go below to benegative, the Invert filter just makes the positive parts negative and vice-versa. This filter is useful when you have astereo file and one of its channels is inverted relative to the other. The audio will sound like it's coming from thesides when you listen to it in stereo, with no audio coming from the center.


Note that the effect will end abruptly at the end of your selection, or the end of the file. If you want the delay orecho to decay naturally, you will need to select a few seconds beyond the sound, first adding some silence to the end ofthe file if necessary.


This effect adds a natural reverberation effect to the selected audio. When you are in a room, a hall, anauditorium, a stadium, or any other kind of enclosed chamber, the sounds you hear have some kind of reverberationbecause of the sound waves bouncing back and forth between the walls, the floor, and the ceiling. This effect ismost noticeable in a large enclosed stadium, where the announcer's voice echoes through the stadium. You first hearthe announcer's voices, and then you hear several, less distinct echoes of the announcer's voice. Usually you don'tnotice reverb because your ears are used to hearing it, but without it, the audio sounds flat, dry, and lacking incharacter. Our ears use reverb to define the size and shape of the room we're in.


Audio signals in the computer are often recorded without any reverb. If you record an instrument directly, or if youuse a unidirectional microphone or one close to the sound source, you will get little or no reverb in the signal. Tomake the audio sound grander, we add reverb.


The "Decay Length" controls how long the reverberations can be heard bouncing between the walls. A short decay meansthat the reverberations die away quickly, while a long delay means that they can be heard longer. Generally, a bareroom with hard surfaces like tile and stone reflect sound well, and will allow the reverberations to keep bouncingaround longer. A room with carpets, drapes, and lots of soft furniture will cause the reverb to die away veryquickly because all those soft surfaces absorb the sound. 350c69d7ab


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