I am making a rhythm game. I need to play a sound with different tempo. In other words e.g. if I have [AVAudioPlayer play] 8 times in 2 seconds.
Check out the
enableRate
and
rate
property on the AVAudioPlayer Class. After you create the audioplayer, but before you play, set
audioPlayer.enableRate=YES;
then after you play, set rate to a number above or below 1.0 to speed up or slow down the track. For music, less than 0.8 or more than 1.2 starts to sound bad, but for a few BMP up or down, it will easily do the trick.
Note that play sets the rate to 1 and stop sets the rate to 0, so be sure to set the desired rate after playing.
I've used Pitch Shifting using the Fourier Transform – Source Code
http://www.dspdimension.com/download/
Related
I'm using OpenAL in my app to play sounds based on *.caf audio files.
There's a tutorial which describes how to generate white noise in OpenAL:
amplitude - rand(2*amplitude)
But they're creating a buffer with 1000 samples and then just loop that buffer with
alSourcei(source, AL_LOOPING, AL_TRUE);
The problem with this approach: Looping white noise just doesn't work like this because of DC offset. There will be a noticeable wobble in the sound. I know because I tried looping dozens of white noise regions generated in different applications and all of them had the same problem. Even after trying to crossfade and making sure the regions are cut to zero crossings.
Since (from my understanding) OpenAL is more low-level than Audio Units or Audio Queues, there must be a way to generate white noise on the fly in a continuous manner such that no looping is required.
Maybe someone can point out some helpful resources on that topic.
The solution with the least change might just be to create a much longer OpenAL noise buffer (several seconds) such that the wobble is at too low rate to easily hear. Any waveform hidden in a 44Hz repeat (1000 samples at 44.1k sample rate) is within normal human hearing range.
I've recently been working on porting my game to be cross-platform, and decided to go with OpenAL for my cross-platform audio engine.
I have 16 "channels" (OpenAL sources) for playing up to 16 sounds concurrently. To play a sound, I switch which buffer is linked to a given source, and also set the gain, source position, and so on in order to play a sound in a given "channel" (source).
The problem is, I've noticed that my "gain" settings do not seem to have immediate effect. For instance, if a loud "lightning" sound plays in a given source at 0.5 gain, then when I have a button click sound play at 0.15 gain later, this click starts off FAR too loud. Then, each subsequent time it is played, the volume decreases until around the 3rd or 4th click it sounds like it's around the proper 0.15 gain.
The first button click is barely audible, and it seems to ramp up in volume until it reaches the 0.15 gain.
So in short, a "source" seems to be remembering the former gain settings, even though I am resetting those before playing a new sound in the same source! Is this a bug? Or something I don't understand about OpenAL? How can I get it to "instantly" change to the new gain/position settings?
Relevant code to play a sound:
[Channel is a value between 0 and 15, soundID is a valid index into the gBuffer array, stereoPosition is an integer between -255 and 255, and volume is between 0 and 255. This is from a function that's a wrapper for my game that used to use values between 0-255, so it converts the values to proper OpenAL values.]
// Stop any sound currently playing in this channel ("source")
alSourceStop( gSource[channel] );
// What sound effect ("buffer") is currently linked with this channel? (Even if not currently playing)
alGetSourcei( gSource[channel], AL_BUFFER, &curBuffer );
// attach a different buffer (sound effect) to the source (channel) only if it's different than the previously-attached one.
// (Avoid error code by changing it only if it's different)
if (curBuffer != gBuffer[soundID])
alSourcei( gSource[channel], AL_BUFFER, gBuffer[soundID] );
// Loop the sound?
alSourcei( gSource[channel], AL_LOOPING, (loopType == kLoopForever) );
// For OpenAL, we do this BEFORE starting the sound effect, to avoid sudden changes a split second after it starts!
volume = (volume / 2) + 1; // Convert from 0-255 to 0-128
{
float sourcePos[3] = {(float)stereoPosition / 50, 0.0, 2.0};
// Set Source Position
alSourcefv( gSource[channelNum], AL_POSITION, sourcePos ); // fv = float vector
// Set source volume
alSourcef( gSource[channelNum], AL_GAIN, (float)newVolume / 255 );
}
// Start playing the sound!
alSourcePlay( gSource[channel] );
I can post setup code too if desired, but nothing fancy there. Just calling
alSourcef( gSource[n], AL_REFERENCE_DISTANCE, 5.0f );
for each source.
We just faced the same problem when setting the ByteOffset for seeking to a position in a sample and found now the solution to get it working:
Just always delete and recreate the source before setting any parameter on it.
So if you want to change the gain and/or other parameters:
OpenAL.AL.DeleteSource(oldSourceID);
newSourceId = OpenAL.AL.GenSource();
OpenAL.AL.Source(newSourceId , OpenTK.Audio.OpenAL.ALSourcef.Gain, yourVolume);
OpenAL.AL.Source(newSourceId , OpenTK.Audio.OpenAL.ALSourcef.xxx, yourOtherParameter);
Hope it works for you and finally have a workaround after 10 years :-)
In my game one audio is paying using
[[SimpleAudioEngine sharedEngine]playeffect:#"audio.aac"];
function .
When I touch one sprite I played another audio which 1 sec.
My problem is that my first "audio.aac" stops when I touch sprite continuously for 8 - 10 times, any solution for it.
Help will be appreciated.
You have to use .wav 44100 Hz 16 bit stereo of .caff formats to play multiply sound effects at once.
You could use
[[SimpleAudioEngine]sharedEngine]playBackroundMusic:#"backgroundmusic.aac"];
to have the 2 minute clip loop while your playEffect
runs separately.
I've recently been trying to incorporate an intensive sound management class, where sound playback precision is a must.
What I'm looking for is the option to load a sound, set the playback starting position (or playhead), play for a certain time, pause the sound, set the 'playhead' position to a new interval and resume playback again. (with dynamic intervals).
I've tried using AVAudioPlayer for that matter - but it seems it's just too slow. The performance is just not what you expect, it lags when calling pause and setCurrentTime:.
It's the easiest library to use and the only one with stated setCurrentTime: function.
I come here asking for your help, a recommendation for a decent open-source SoundEngine that can handle interval setting (playhead movement) with low latency, or reference to where it is stated that OpenAL or AudioUnit tools can handle playback position setting.
Thank you in advance,
~ Natanavra.
It would be worth your time to check out the openAL programmer's guide that comes with the SDK. Its got all sorts of goodies!
From that:
Under source: Each source generated by alGenSources has properties which can be set or retrieved.
The alSource[f, 3f, fv, i] and alGetSource[f, 3f, fv, i] families of functions can be used to set or retrieve the following source properties:
...
AL_SEC_OFFSET f, fv, i, iv the playback position, expressed in seconds
AL_SAMPLE_OFFSET f, fv, i, iv the playback position, expressed in samples
AL_BYTE_OFFSET f, fv, i, iv the playback position, expressed in bytes
So you can get the playback position in seconds and divide by 60 to get your normalized time.
float pos = 0; alGetSourcef( sourceID, AL_SEC_OFFSET, &pos );
float normalizedPos = pos / 60.0f;
OpenAL definitely has the capabilities to playback sound pause whatever you like. remember OpenAL is often used in games as it delivers sound playback with low latency and on demand playback. You have a lot of control over the sound. compared to the AVAudioPlayer class.
Hope this helps
Do reply
Pk
I have a MPMoviePlayerController in my project.
Documentation says that next call:
moviePlayer.initialPlaybackTime = time;
starts at the closest key frame prior to the provided time.
Is it possible to start playing video from the specified time (not from the nearest key frame)?
No, it really isn't. Temporally compressed video streams can only generally start playback on a keyframe, as inter-frames depend on the keyframe for rendering. If seekability is important to you, consider making files with smaller keyframe intervals.