In my Codename One app, I need to limit the duration of an audio capture to few seconds. I call Capture.captureAUdio() for that and there does not seem to be an equivalent to Capture.capturePhoto(w, h) that could bound the captured audio to a given duration.
What I've done so far is measure the audio capture duration and if it is too long then ask the user to start again if they will. But I find it a bit cumbersome.
Is there anything better that can be done ?
Any hint appreciated,
No.
For audio though you can use use the media recording feature from MediaManager which isn't as intuitive as Capture but allows more "fine grained" control.
Related
I've got a little side project going on using SDL2/SDL_mixer and a couple other sound libraries. I've been trying for a while now to synchronize my audio and video but haven't been able to get it anywhere near successfully. All new to this stuff so forgive the poorman's logic and coding. At first I thought to set the delay to SDL_Delay(30) after every frame, and then a few other numbers in that range. Not quite right. Then I tried doing it by getting Ticks. Where I would get the difference between current_ticks and last_ticks and set a delay if the delta between ticks was <=30 and set the delay to 30-delta. Still not quite right (by far). Hoping someone on here with more experience might guide me in the right direction. In regards to the video, it's a visualizer of course, seems like a popular beginners project.
The basic way you synchronize audio and video is that you choose one to use as a timer source and present the other according to that timer. The easiest is generally audio, but because it's generally buffered ahead, you need some method of measuring what time in the audio stream is actually coming out of the speakers. Once you get that, it's just a matter of waiting until the audio reaches the right time for the next video frame and displaying it.
I'm trying to make a guitar practice website, and a critical functionality is to loop over very short mp3 files (a few seconds long), with absolutely zero gap in between. For example, it could a 4-measures-long chord progression, and I want to allow the user to loop over it seamlessly.
I tried using the HTML5 <audio> tag with the loop attribute. Google Chrome gives a small gap between the loops, but big enough to be totally unacceptable for my purpose. I haven't tested the other browsers, but I believe it won't work.
A possible workaround is to use ffmpeg to stream repetitions the same audio as an mp3. However, this costs a lot of bandwidth.
For myself I use Audacity to loop without gaps, but unfortunately Audacity doesn't have a web version.
So, do you have any ideas how I may loop over an mp3 in a browser with zero gap? I prefer non-Flash solutions, but if nothing else works I'll use Flash.
Edit:
Thank you for all your suggestions. Flash turns out to work decently. I've made a toy demo at http://vmlucid.lcm.hk/~netvope/audio/flash.html. To my surprise (I use to associate Flash with resource hog and browser crashes only), Flash and ActionScript are rather well designed and easy to use. It took me only 3 hours on my first Flash project :)
Have a look at this page. Listening for a while using Google Chrome 7, I found Method 1 works decently, while Method 3 gives the best results, though it's a bit of a hack. Ultimately, all browsers work differently, especially since HTML5 isn't finalized yet. If possible, you should opt for a Flash version, which I would think would give you the best loop.
in flash AS3 you can extract sound data with computeSpectrum() and give it to your Sound object exactly when it's needed (SampleDataEvent is fired).
I am not sure how well this will work, but if you knew your loop lasted 800 milliseconds - you could have the browser call the play method every 800ms... it still wouldn't be guaranteed to be perfect though. I don't think the browser is natively capable of delivering reliable audio looping at this point.
setInterval(function(){
document.getElementById("loop").play();
}, 800);
Rumor has it the best way to do pull this off in the most gapless fashion to use to multiple audio tags and alternate between them.
Or check out this utility: http://www.compuphase.com/mp3/mp3loops.htm I used it successfully for my flash projects when music had to be looped without gaps, and 99% of the time it worked. It takes WAV as an input.
Basically it is a kind of front-end for LAME mp3 encoder, which uses such settings as to prevent the gaps appearing. It won't work on very short sound effects (less than 0.5 second I believe).
Afterward all you have to do is use:
var sound:Sound = new MySoundEffect();
sound.play(0, 1000);
and it will loop one thousand times.
I am trying to record a stream from a webcam using Expression Encoder 4 SDK in WPF I can capture the video & audio streams and record these to disk however they are only recording at a base resolution of 320x240 the webcam is capable of capturing at 720p, how can I record at this resolution. Any help would be appreciated, I have been pulling my hair out trying to solve this all week.
Know this is a bit late but all questions need answers:
These might be a possible solution:
Check to see if your camera has it's own settings on the camera or comes with an installation disk.
for the expression encoder 4 put the video profile quality to max.
Good luck. If you are still around tell me, how it goes.
to change the "size" you can use the following line :
LiveJob.OutputFormat.VideoProfile.Streams[0].Size = new Size(1280,1080)
Or whatever you want it to be.
Encoder also offers a setting page that you can use.
That's what I did and then after setting the outputsize you can do that :
currentJob.OutputFormat.VideoProfile.Streams[0].Size = ((LiveSource)LiveDeviceSource).CropRect.Size;
Only 1 small limitation, you can't change the size while it's recording if you are publishing the source.
I'm looking for the most realistic way of playing sound of a rolling ball. Currently I'm using a Wav sample that I play over and over as long as the ball is moving - which just doesn't feel right.
I've been thinking about completely synthesizing the sound, which I know very little about (almost nothing), I'd be grateful for any tutorials/research materials/samples concerning synthesis of sound of a ball made of particular material rolling on surface made of another material. Also if this idea is completely wrong, please suggest another way of doing this.
Thanks!
I would guess that you'll get the biggest bang for your buck by doing a dynamic frequency adjustment on the sound that makes the playback frequency proportional to the velocity of the ball. I don't know what type of sound library you use, but most will support some variant of this.
For example, in FMOD you could use the Channel::setFrequency method. Ideally, you would compute your desired playback frequency based on your WAV's original sample frequency (Fo), the ball's current velocity (Vc), and the ball's 'ideal' velocity at which the default WAV sounds right (Vi). Something generally like:
F = Fo * ( Vc / Vi )
This will tend to break down as the ball gets farther away from the 'ideal' velocity. You might want to have several different WAVs that are appropriate for different speed ranges that you switch to at certain threshold velocities. Within each WAV's bracket, you'd do the same kind of frequency adjustment.
Another note: this is probably not something that is worth doing every frame. I'd guess that doing this more than 20 times per second would be a waste of time.
ADDENDUM: Playback frequency scaling like this can also be used for simulating the Doppler effect as well. Once you have your adjusted playback frequency, you'd perform another scale of the Frequency based on the velocity of the ball relative to the 'listener' (the camera).
Have you tried playing the sound forward, then playing it backward, and looping that? I use this trick graphically to creating repeating patterns. I don't know much about sound but it might work?
One approach might be to analyze the sound of a rolling ball, and decompose it into its component waveforms. Then you'd be able to generate your own wav file with synthesized waves.
You should be able to do this using an FFT on a sample of the sound.
One drawback is that the sound will likely sound synthesized - you'll have to add noise and such to make it sound more realistic. Getting it to sound real enough may be the hardest part.
I don't think you need the trouble to synthesize that. It would be way too hard to even sound convincing.
Depending on how your scene is, you could loop the sound foward/backwards and simulate a doppler effect applying a low pass filter and/or changing it's pitch.
By the way, freesoung.org is a great place for free samples. They are not professionally recorded but are a good starting point for manipulation. On the other hand, sound ideas has some great sample cds (they're actually industry standard) if you can find them on the cheap. You just have to search for which one has rolling ball sounds.
I really like the approach described in this SIGGRAPH paper:
http://www.cs.ubc.ca/~kvdoel/publications/foleyautomatic.pdf
It describes synthesizing the sound of a rock rolling in a wok (no, really :). The idea is to use modal synthesis (i.e. convolved impulse responses), and the results can be very convincing.
Here's a link to the video demo that goes with the paper:
http://www.cs.ubc.ca/~kvdoel/publications/foleyautomatic.mpeg
And here's a link to the JASS library (written by one of the authors), which was used to create the sound for the video:
http://www.cs.ubc.ca/~kvdoel/jass/jass.html
I'm not sure if you could make it run on a smart phone, but with an efficient enough convolution routine/approximation you might be able to do something interesting...
My question is 'why?' - do you see some benefit in this, or is it just for fun? Your question implies that you aren't happy with the wav you are using but I strongly believe that synthesising your own is going to sound far inferior.
If your wav sample doesn't sound right, I'd suggest try to find another sample. Synthesising a sound is not easy and is never going to sound as realistic as your sample.
Real time synthesis may require more resources for processing and computation. You may very well end up prerendering your synthesised sound into a wav file and performing a playback.
If you want to simulate the sound of different materials then you can use some DSP, or even simple tricks like slowing or speeding the wav playback. The simplest way is the prerender these in another application and store one copy of the file for each use.
I'm attempting to use a large number of short sound samples in a game I'm creating in Silverlight 2. The samples are less than 2 seconds long.
I would prefer to load all the audio samples onto the canvas during the initualization. I have been adding the media element to the canvas and a generic list to manage it. So far, it appears to work.
When I play the sample the first time, it plays perfectly. If it has finished playing and I want to re-use the same element, it cuts off the first part of the sound. To play the sample again, I stop and play the media element.
Is there another method I should use the samples so that the audio is not clipped and good performance is obtained?
Also, it's probably a good idea to make sure that all of your audio samples are brought down to the client side initially. Depending on how you set it up, it's possible that the MediaElements are using their progressive download functionality to get the media files from the server. While there's nothing wrong with this per se (browser caching should be helping you out after the initial download), it does mean that you have to deal with the browser cache, and there are some potential issues there.
Possible steps to try:
Mark your audio files as "Content". This will get them balled up in the .xap.
Load your audio files into MemoryStreams (see Application.GetResourceStream method) and call MediaElement.SetSource().
HTH,
Erik
Some comments:
From MSDN:
Try to limit the number of MediaElement objects you have in your application at once. If you have over one hundred MediaElement objects in your application tree, regardless of whether they are playing concurrently or not, MediaFailed events may be raised. The way to work around this is to add MediaElement objects to the tree as they are needed and remove them when they are not.
You could try to seek to the start of the sample to reset the point currently being played before re-using it with:
mediaelement.Position = new TimeSpan();
See also MSDNs MediaElement.Position.
One techique you can use, although I'm not sure how well it will work in Silverlight, is create one large file with all of your samples joined together (probably with a half-second or so of silence between each). Figure out the timecode for each sample and seek the media element to that position and play. You'll only need as many media elements as simultaneous sounds you want to play.