Is there anyway I can get better playback controls? For example I'd like to be able to carefully scrub through playback, like if I were learning a guitar solo or something. I might like to slow down, frequency morph, etc. Is the audio playback locked down pretty tight or can I control how the audio hits my sound card?
Thanks,
Tony
Unfortunately, the API Terms of Service prevent you from doing this sort of audio manipulation to Spotify's audio.
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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 regularly testing smartphones for my blog and I'd like to measure their audio quality (file playback but also call quality).
I thought of connecting the jack port from the smartphone to the line-in input of my soundcard and play some sounds to measure the quality through a software.
I'd like to measure sound quality based on THD, SNR, Crosstalk, ...
Is there a software I could use to do this? Would you recommend another method to achieve the same?
Thanks
Laurent
I found found something that might do the trick. It's a program called RightMark Audio Analyzer, it can use a test signal on external devices to be re-analyzed through line-in.
I'm trying to make a video tutorial, so i decided to record the speeches using a TTS online service.
I use Audacity to capture the sound, and the sound was clear !
After dinning, i wanted to finish the last speeches, but the sound wasn't the same anymore, there is a background noise(parasite) which is disturbing, i removed it with Audacity, but despite this, the voice isn't the same ...
You can see here the difference between the soundtrack of the same speech before and after the occurrence of the problem.
The codec used by the stereo mix peripheral is "IDT High Definition Codec".
Thank you.
Perhaps some cable or plug got loose? Do check for this!
If you are using really cheap gear (built-in soundcard and the likes) it might very well also be a problem of electrical interference, anything from ...
Switching on some device emitting a electro magnetic field (e.g. another monitor close by)
Repositioning electrical devices on your desk
Changes in CPU load on your computer (yes i'm serious!)
... could very well cause some kinds of noises with low-fi sound hardware.
Generally, if you need help on audio sounding wrong make sure that you provide a way to LISTEN to the files, not just a visual representation.
Also in your posted waveform graphics i can see that the latter signal is more compressed, which may point to some kind of automated levelling going on somewhere in the audio chain.
Is there a difference between audio.play() and media.play() and which one is better?
The audio.* API calls use the OpenAL audio layer to play. They are considered a safer and better way to play audio in Corona SDK. You can have 32 different sounds playing at once. You can control the volume on each channel independently, pause and resume, fade in, fade out, etc. It is the preferred way to play sound.
The media.* API calls write directly to the hardware and you cannot control the volume, have multiple sounds going on. The media.* API Calls though are good for video, playing long clips, like podcasts since that audio can be backgrounded, but more importantly, on Android, Google has decided to poorly implement OpenAL and under 4.x there is a significant lag from the time you tell audio.play() to play a sound and it really happening. The lag isn't as bad under 2.2 and 2.3, but there still is a lag. The media.* api calls, if you're playing a short clip will play in a timely fashion.
media API:Only one sound can be playing using this sound API. Calling this API with a different sound file will stop the existing sound and play the new sound.
Having just witnessed Sound Load technology on the Nintendo DS game Bangai-O Spritis. I was curious as to how this technology works? Does anyone have any links, documentation or sample code on implementing such a feature, that would allow the state of an application to be saved and loaded via audio?
Its the same old thing used in ZX Spectrum era. You load programs/games from tape.Only the sound quality and the filters are probably better.
In my opinion something like Bluetooth or WiFi is better. You can also send files that can be put on some storage and then load them. I find these methods much easier than sound because if there is a lot of noise around you cannot do much.
It is just a conversion of data to audio and then back from audio to data.
Search for Zotyocopy and Copy86M on google - these are the utilities used for saving a game to tape after loading it into memory on zx spectrum.
If you want to pass data as audio through the air there are a few things you need to be aware of though, such as how the speaker and microphone interact for example. It is important that they don't distort or alter the sound too much as what you are sending are in fact the raw bytes.
Some audio software will let you open any file as audio so that you may listen to it. If you record audio as data do not use lossy compression such as mp3 on the audio file!