Can Finch change the user's recorded voice into a robot's voice? By changing the pitch, rate, tempo etc?
I’m afraid not. It can change the sound pitch, but only while also changing the speed.
Related
I am having so much trouble doing something that should be SIMPLE. I do sales for a golf course and I have to read the same thing over and over again on every call and it gets so damn annoying. I want to be able to play a pre-recorded wave/mp3 file through the mic input of my headset so I can just play the recording at the right point in the sales cycle instead of repeating it 200X a day. I have succeeded in doing it with stereo mix BUT it will disable the voice aspect of the microphone so when the recording is finished, I have to jump into setting real fast and switch the mic input - which is not doable.
I know there is a way to do this. I see twitch streamers do this sort of thing all the time. I have tried SO MANY methods and nothing seems to work.
Recently, I discover that my tutorial videos could be seen at 1.5x playback speed without losses in quality (they are actually better to see, as I normally speak slowly). My problem is that if I change the speed of the video when using a video editor, like Kdenlive, the audio becomes distorted and turns into a mess (higher pitch, I believe).
How could I obtain the same quality as VLC "playback fast" and Youtube "playback speed 1.5" for the audio track? I'm a layman in audio/video editing, so I'm also satisfied with partial answers, like the identification of which terms I should search for in this case.
It might be better to take your audio track and use something like Sound Forge to automatically remove silence. Just be sure to add a pad to that (built into sound forge) otherwise the speech will sound way to chopped and fast.
Aside from that, you could also use Vegas to (then) chop the video to keep pace with your new speech rate. Vegas is a video editing program that is best for this kind of down and dirty editing.
I'm trying to use FMOD to develop an application that is expected to be able to play audio more slowly than normal so that the user could hear the audio more clearly. In my code, I called Channel::setFrequency like this:
float normal_frequency;
channel->getFrequency(&normal_frequency);
channel->setFrequency(normal_frequency * speedSelected);
If the value of speedSelected is lower than 1, for example 0.8, the audio will indeed be played more slowly than normal, but the voice sounds really odd. Playing slowly doesn't enable me to hear audio more clearly at all.
By contrast, Microsoft's Windows Media Player works perfectly when it plays audio more slowly than normal.
Is there a way to solve this problem?
If by "sounds really odd" you mean the pitch has been altered then this is the expected outcome. If you want to correct the pitch while adjusting the speed you will need to use the pitch shifter DSP.
I'm working on a simple music visualization. Probably not relevant, but I am doing the sound processing using the new WebKit Audio Data API and the dsp.js library.
I want to make a text vibrate (grow/shrink) to the rhythm of the music. What is the best way to do this?
What I've done so far is ran the signals through a FFT. I look at the bottom 10% of frequencies (bass notes?) and when the amplitude surpasses a certain threshold, I animate the text.
Does this sound right? Or am I completely off?
You say you've done it, and then you ask if you are way off? Well, you tell us: does it work for your application?
One potential problem is that the FFT is slow, both in that there may be a lag between your input and output and there will be a lot of CPU used. I don't expect this will matter for your application, but, in general, you are better off using a low-pass filter. When the output of the low-pass goes above some level, you can use that to trigger something for some short amount of time.
Another issue is simply that this is only a very basic beat detection algorithm. It might work for bass-heavy "four on the floor" music, but you'll need to figure out where the threshold goes and how to keep it moving when the bass stops or something. You may want to research beat detection algorithms. The open source aubio has some.
http://aubio.org/
I'm trying to develop an online application where the user writes some text and the software sings it back to the user.
I can currently generate the audio file with the words spoken by the computer using espeak, but I have no idea how to make it sound like a song, how to add rhythm to it.
I'm able to change the pitch and tempo using rubberband, but that's as far as I've gotten.
Does anyone have a clue how to make this happen?
If you want to use rubberband to change duration and pitch, then I think the hard part is going to be mapping from phonemes/syllables in the text to corresponding audio ranges in the speech systhesis output, for which I have no simple suggestion. (Ideally you'd get inside the speech synthesiser so that it would provide you with the mapping from phonemes to audio location.)
A simpler alternative might be to try Speech Synthesizer Markup Language - SSML. It has a "pitch" and "duration" elements that can absolutely specify pitch in Hz and duration in seconds. You can also specify volume, for controlling dynamics.
Given this, you could try to convert the text into a SSML document, and mark up words/syllables/phonemees with pitch/duration and volume attributes.
I've ended up using Festival's singing mode. It sounds reasonably well, except for the fact it only works with English voices.