# Tag Info

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A sample is the smallest usable quantum of digital audio. The term frame isn't formally defined in pure audio terms, but is often used in relation to video that may accompany an audio track. In that context a frame is the quantity of audio samples taken during a video frame interval. If there are 30 video frames per second and 48000 audio samples per ...

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This comes down to the question of audibiltiy of phase in a signal. To test this, I created two signals (both with f0 = 220.5 Hz, Fs = 44100 Hz) in Matlab, a saw wave with an upwards facing ramp and one with a downwards facing ramp: ramp_up = repmat(linspace(-1, 1, 200), 1, 440)'; ramp_dn = repmat(linspace(1, -1, 200), 1, 440)'; These are (excerpts of) the ...

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I am an engineering student my self and I use Audacity to generate Sine, Sawtooth, Square waves. It's easy as pie. Download Audacity from here. Open Audacity and click on Generate from the Toolbar on top. Select the waveform you need, the frequency, the amplitude and duration you want it. Play the track.

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The term 'frame' has several different meanings, dependent on the context. Beside the obvious meaning in video contexts, and the usage in audio codecs like MP3, a frame may also be used as a name for all the samples in a bundle with several channels, which belong to the same point in time. Yet another usage of the word is with communication protocols, where ...

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The highly simplified answer is that sound in the real world is not sine waves with fixed frequencies. We learn about sound initially as frequencies or notes and simplify the concept down to thinking about various sine waves to make it easier to understand the fundamentals of what is going on. The reality, however, is that all sound propagates through the ...

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You may want to check out SPEAR. From the website: SPEAR is an application for audio analysis, editing and synthesis. The analysis procedure (which is based on the traditional McAulay-Quatieri technique) attempts to represent a sound with many individual sinusoidal tracks (partials), each corresponding to a single sinusoidal wave with time varying frequency ...

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Any DAW (digital audio workstation) or even most video editing software will display a waveform for the audio in a particular file. You can not, however, make a "sine wave of whatever audio file" because the file doesn't contain a sine wave. A sine wave is a very particular type of fixed frequency oscillation. You would set a signal generator to a ...

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The effect of changing the phase of a harmonic, might simply equate to (acoustically) that particular harmonic coming (to the ears) from a slightly different distance than the fundamental. Thinking about it like this, I would have to say that the probability of it 'changing' the sound is quite limited. You are more likely to get a change in timbre of the ...

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A waveform is a period of superimposition of all the frequencies involved, each of which can be deconstructed into sine waves. Therefore a waveform will always represent those frequencies, theoretically. If the waveform is cut or sampled too short, some lower frequencies might be excluded. It is possible however to have different waveforms represent near-...

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You should check out Sonic Visualiser. It's free, has image export built in and is generally designed for this sort of thing.

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From my experience as a vinyl cutting engineer, the phase response makes it look more triangular in shape, but it sounds very similar.

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You can use Sony Sound Forge, there's a waveform generator inside of it. Best for you could be Audacity which is free see the doc. You could also use a synth in a DAW to generate sines even squares etc...

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Essentia is an Open-source library and tools for audio and music analysis, description and synthesis It includes some pre-built extractors that you can use.

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Ok, let's start from the basics. Audio is usually recorded (sampled - converted from a continuous analogue signal to a digital record) and stored as a series of numbers representing the momentary voltage of the signal at regular intervals. Sampling rate is the rate at which we check and record the momentary value of the analogue signal. Common sampling ...

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Assuming that your app is working at all, the most likely explanation is that the microphone circuit in your tablet contains a high pass filter (a filter which removes low frequencies). This means that the sounds that are bothering you, which are low in frequency, will be recorded as much lower in level than they actually are in reality. You may be able to ...

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One option is to apply a brickwall limiter with a threshold around -6 dBFS to the track. That should lower the higher audio peaks. You would then have room to apply a higher gain to the whole track. Be aware that a limiter might produce undesirable artifacts, you should therefore test the limiter settings on a few peaks to fine tune the threshold before ...

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There is a program called PhonoPaper made by a Russian group. It is essentially a graphical representation of sound that can be read by a camera and specific software. Quite cool, if somewhat novel. http://warmplace.ru/soft/phonopaper/ There is possibly some way of reading a sonograph image back into sound. Perhaps look into SSTV technology too which is ...

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They will sound exactly the same. The human ear responds to the frequency of the waveform - not the direction of the waveform itself.

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A way to understand timbre and differences between instruments is to interpret harmonics by casting an eye on their spectrograms: The flute has some tremolo visible on the high-end sustained harmonics that look consistent throughout the sample. Piano, on the other hand, has gradient like harmonics fading towards the higher ones. Also the wavefile has a ...

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ableton has a drag n drop feature that allows you to place a wave or mp3 in a midi track and get kind've interpretation of what the melody and harmony is, though you may have to use your sensibilies to extract the parts you want

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This sounds like an audio to midi converter. By analysing the audio file, you can retrieve pitch and duration information which coincidentally is what you require to generate midi note-on and note-off messages. Something like this?

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Generally, the trick is to cut at the quietest point, just before the beat, where the waveform crosses the zero line. If your beat is a kick drum, that's usually the simplest to find, it will form the biggest wave shape at the lowest obvious frequency. Back in the analog tape days, you'd shuffle the tape back & forth until you could hear that slow ...

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I don't think you'll find a method that completely removes the backing, but you may be able to reduce the volume enough for an acceptable result: You simply put all the all the different versions on a track each in a multitrack program like Reaper, Cubase or Pro Tools. Make sure you precisely align the songs so the wave curves follow each other at sample ...

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Thing is that such a signal only exist in theory - practically this signal will be made up of a number of sine waves with varying frequency and amplitude. The generator or the sample rate format determines the upper frequency limit (i.e. how "sharp" that seemingly discontinuous signal edge will be a the corner/peaks). This alone will transform your step ...

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Guys who master vinyl will tell you something like this: - A pure step signal has an infinite spectrum and, considering you can't hear above 20kHz (a lot less as you get older) there's no point doing what you suggest. You may then say "oh please do it" and the vinyl mastering guy might say "OK we'll give it a go" and then completely ignore your request ...

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Practicing proper editing/cutting of audio samples improves heavily the skill you are looking for. Try for example to cut a speech (high quality, speaking your language) into its smallest parts. Try to separate the consonants from the vowels. This way you have to zoom into the waveform very deeply (see the describing shape of a particular sounding), until ...

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Vertical asymmetry's not a problem; likely just a product of the phase relationships between the superimposed sounds. There are ways of rectifying it (google phase rotator plugins) but it's no harm. It's not DC offset - the entire waveform, including the silence at the beginning/end, would be shifted vertically one way or the other away from the infinity ...

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