I have a heart-rate monitor, and a short wav file which sounds like a heart beating. This wav file has a duration of 1 second.
I want to play it in time with the user's heart rate. So if their heart rate is 100bpm, the sound should play for 0.6 seconds. If their heart rate is 60bpm, the sound should play for 1.0 seconds. And if their heart rate is 30bpm they are probably about to die.
Is it possible to alter the duration of time it takes to play a sound file? And what framework should I use? AVPlayer?
check this one: How can I use AVAudioPlayer to play audio faster *and* higher pitched?
It says you can change the .rate property of an AVAudioPlayer to play it faster.
ps: it won't change the pitch of the sound. If you also need this, check the question I mentioned.
Related
In my app users record themselves singing over a backing track, and then later playback the recorded audio and this backing track at the same time. I use expo-av for my audio system. The problem is that at the playback stage the audio is often out of sync because expo only really supports asynchronous audio. Does anyone have any advice on how to approach this problem at a high level?
A few of my ideas:
Mix the two audio files into a single file for playback. This almost works except for the fact that the recording and backing track are also out of sync. If I knew exactly how much they were offset, I could just add that amount of silence to one of the files when mixing. However, I haven't found a way to accurately calculate this offset.
Reduce time it takes for recording and playback to start, so that the latency is not noticeable. Some things I've found that help here are recording at lower quality and using smaller audio files. Any other tips here would be appreciated.
Use a different audio library than expo-av. Is there one that comes to mind that better supports synchronous audio? Ideally it would also be supported by Expo or at least React Native.
I'm not sure if it's allowed ask these questions here, but looks important for us webdevelopers (even bad dev like me :p ).
The question is about export setting videons on Premiere. I'm looking for a background video 30s like airbnb or paypal. Yesterday, I check paypal size and it's only 10/15 Mb for more than a minute. How did they do?
Obviously you want a low average bit rate. Things that can help with that are: keep the resolution low (you can scale it up a bit on the client); use H.264 High Profile (for the H.264 version); use 2-pass encoding; use variable bit rate. You can try increasing the GOP length too.
I assume there's no audio, so that shouldn't be an issue. (Can't remember if Adobe has an option for no sound track, but you can set the audio to a very low bit rate, or post-process it with ffmpeg or something to remove the audio track.)
If you have any control over the video content, you can try to keep it compressible. For example, avoid video with lots of detail or rapid motion. You might be able to selectively blur parts in a way that doesn't look bad. If it doesn't move too fast, you might be able to decrease the frame rate.
If you really want to optimize, you'll probably need to experiment a lot.
I'm wondering whether it's possible to slow down a sound in xcode. I mean I'll add some .mp3 file in my supporting files in xcode and I'll create app which will be able to speed it up or slow down. For example with slider. Is it even possible? If yes, could anyone help me with some idea? Thanks
AVAudioPlayer has a rate property which should be able to help you accomplish your goal.
http://developer.apple.com/library/IOS/#documentation/AVFoundation/Reference/AVAudioPlayerClassReference/Reference/Reference.html
The audio player’s playback rate. #property float rate
Discussion
This property’s default value of 1.0 provides normal playback rate.
The available range is from 0.5 for half-speed playback through 2.0
for double-speed playback.
To set an audio player’s playback rate, you must first enable rate
adjustment as described in the enableRate property description.
I also found a good SO post on the AVAudioPlayer's rate:
AVAudioPlayer rate
Seems like as you'd mentioned you could set a slider with values from 0.5 to 2.0 and on valuechanged modify the audio players rate by using
- (IBAction)changeValue:(UISlider *)sender
{
//made up assumed ivar names
if ([_audioPlayer respondsToSelector:#selector(setEnableRate:)])
_audioPlayer.enableRate = YES;
if ([_audioPlayer respondsToSelector:#selector(setRate:)])
_audioPlayer.rate = [NSNumber numberWithFloat:slideValue];
}
Playing PCM audio at a faster or slower rate than its sample rate changes its pitch and also introduces considerable artefacts. If you're OK with this, the approach you would use is to decode the MP3 into PCM audio and then use Direct Digital Synthesis Oscillator to control the playback rate.
If you want to maintain pitch but change speed, you need a audio time-stretching algorithm.
Dirac3 from DSP Dimension is a commercial product that can do it, and which is available for licensing for use in iOS application. Other commercial solutions exist.
DSP Dimension's blog provides a helpful tutorial on the basics of how to implement pitch-shifting
using a FFT. Time stretching is essentially the same process. However, there's a fair bit of secret sauce in the DIRAC plug-in they don't tell you about.
Be warned that unless you're a Electronics engineering, physics or maths graduate, you'll probably it tough going to fill in the blanks.
I have an array of floats which represent time values based on when events were triggered over a period of time.
0: time stored: 1.68
1: time stored: 2.33
2: time stored: 2.47
3: time stored: 2.57
4: time stored: 2.68
5: time stored: 2.73
6: time stored: 2.83
7: time stored: 2.92
8: time stored: 2.98
9: time stored: 3.05
I would now like to start a timer and when the timer hits 1 second 687 milliseconds - the first position in the array - for an event to be triggered/method execution.
and when the timer hits 2 seconds and 337 milliseconds for a second method execution to be triggered right till the last element in the array at 3 seconds and 56 milliseconds for the last event to be triggered.
How can i mimick something like this? i need something with high accuracy
I guess what im essentially asking is how to create a metronome with high precision method calls to play the sound back on time?
…how to create a metronome with high precision method calls to play the sound back on time?
You would use the audio clock, which has all the accuracy you would typically want (the sample rate for audio playback -- e.g. 44.1kHz) - not an NSTimer.
Specifically, you can use a sampler (e.g. AudioUnit) and schedule MIDI events, or you can fill buffers with your (preloaded) click sounds' sample data in your audio streaming callback at the sample positions determined by the tempo.
To maintain 1ms or better, you will need to always base your timing off the audio clock. This is really very easy because your tempo shall dictate an interval of frames.
The tough part (for most people) is getting used to working in realtime contexts and using the audio frameworks, if you have not worked in that domain previously.
Look into dispatch_after(). You'd create a target time for it using something like dispatch_time(DISPATCH_TIME_NOW, 1.687000 * NSEC_PER_SEC).
Update: if you only want to play sounds at specific times, rather than do arbitrary work, then you should use an audio API that allows you to schedule playback at specific times. I'm most familiar with the Audio Queue API. You would create a queue and create 2 or 3 buffers. (2 if the audio is always the same. 3 if you dynamically load or compute it.) Then, you'd use AudioQueueEnqueueBufferWithParameters() to queue each buffer with a specific start time. The audio queue will then take care of playing as close as possible to that requested start time. I doubt you're going to beat the precision of that by manually coding an alternative. As the queue returns processed buffers to you, you refill it if necessary and queue it at the next time.
I'm sure that AVFoundation must have a similar facility for scheduling playback at specific time, but I'm not familiar with it.
To get high precision timing you'd have to jump down a programming level or two and utilise something like the Core Audio Unit framework, which offers sample-accurate timing (at 44100kHz, samples should occur around every 0.02ms).
The drawback to this approach is that to get such timing performance, Core Audio Unit programming eschews Objective-C for a C/C++ approach, which is (in my opinion) tougher to code than Objective-C. The way Core Audio Units work is also quite confusing on top of that, especially if you don't have a background in audio DSP.
Staying in Objective-C, you probably know that NSTimers are not an option here. Why not check out the AVFoundation framework? It can be used for precise media sequencing, and with a bit of creative sideways thinking, and the AVURLAssetPreferPreciseDurationAndTimingKey option of AVURLAsset, you might be able to achieve what you want without using Core Audio Units.
Just to fill out more about AVFoundation, you can place instances of AVAsset into an AVMutableComposition (via AVMutableCompositionTrack objects), and then use AVPlayerItem objects with an AVPlayer instance to control the result. The AVPlayerItem notification AVPlayerItemDidPlayToEndTimeNotification (docs) can be used to determine when individual assets finish, and the AVPlayer methods addBoundaryTimeObserverForTimes:queue:usingBlock: and addPeriodicTimeObserverForInterval:queue:usingBlock: can provide notifications at arbitrary times.
With iOS, if your app will be playing audio, you can also get this all to run on the background thread, meaning you can keep time whilst your app is in the background (though a warning, if it does not play audio, Apple might not accept your app using this background mode). Check out UIBackgroundModes docs for more info.
I am running a countdown timer which updates time on a label.i want to play a tick sound every second.i have the sound file but it is not sync perfectly with time.how do i sync time with audio?also if i use uiimagepicker the sound stops how to manage this? if someone have tick sound like a clock has it would be great.
The best way to sync up your sound and time would be to actually play short - less than a second long - sound files (tick sound) once per second as TSTimer fires. It won't sound as nice as a real clock or chronometer ticking, but it would be easy to do. And if the sounds are that small, then you don't have to worry too much about latency. I think to be realistic you need to play two ticks per second, with the first and second ticks about 0.3 seconds apart, and the next one starting at the next second, with the fourth, again only about 0.3 seconds later. And so on.
For even tighter integration of sounds and GUI, you should read up on Audio Toolbox:
Use the Audio Toolbox framework to play audio with synchronization capabilities, access packets of incoming audio, parse audio streams, convert audio formats, and record audio with access to individual packets. For details, see Audio Toolbox Framework Reference and the SpeakHere sample code project.