I am using OpenAL to pitch shift a note. e.g.
alSourcef(source, AL_PITCH, aPitch);
I am noticing however an audible click when I do this. Other than that the pitch is perfect, correct pitch etc.
Any ideas what might be causing this?
I haven't used OpenAL, but in other sound libraries I have seen this "artifact". There is usually, when dealing with tone generator etc. a variable for the time it takes a tone to reach 100% volume level, I can for the life of me not remember what it is called :)
like this:
playTone(400 Hz, 40 dB, 50 ms, 3000 ms).
where 400 is the Hz, 40 dB the volume, 3000 milliseconds is the duration and 50 milliseconds is the time it takes from starting the tone at volume 0 (or +100dB) to it reaches 40 dB. I simply can't find the word right now.
Anyways, if you have the ability to set this variable, try doing that, just set it to something like 10 ms. You wont be able to hear it, but it has removed clicking sounds for me in both an open source sound library I used for the iPhone and in some Java/Processing libraries I used in the past.
Maybe it has to do with the way the underlying code is triggering some hardware connected to the speaker?
i have experience on this one, mostly it is because you shift the pitch too high or too low, shifting pitch is stretching or shrinking wave-data length, the case is if your data does not have enough sample to stretch it will sound "weird", in case of shortening the length (pitch-up) if your playback buffer does not have enough sample to feed in time, it will lag or jitter because conceptually the playing rate is increased to due shortened the length of audio, mostly clicking or popping is what you heard.
to prevent this, you should limit the shifting range, mostly 0.5 to 2.0 is the limit on most sound-card, and it is vary across soundcard, since shifting the pitch could be make better by using some advanced smoothing and processing in DSP, so it will depend on processing power of your DSP or CPU to do such processing. i've tried it using onboard intel HDA that the limit is mostly 0.5 to 2.0, but using X-Fi soundcard it is better, shifting to 0.1 .. 5.0 doesn't have a problem
Related
I am using a G0 with one ADC and 8 channels. Works fine. I use 4 channels. One is temperature that is measured constantly and I am interested in the value every 60s. Another one is almost the opposite: it is measuring sound waves for a couple a minutes per day and I need those samples at 10kHz.
I solved this by letting all 4 channels sample at 10kHz and have the four readings moved to memory by DMA (array of length 4 with 1 measurement each). Every 60s I take the temperature and when I need the audio, I retrieve the audio values.
If I had two ADC's, I would start the temperature ADC reading for 1 conversion every 60s. Non-stop. And I would only start the audio ADC for the the couple of minutes a day that it is needed. But with the one ADC solution, it seems simple to let all conversions run at this high speed continuously and that raised my question: Is there any true downside in having 40.000 conversions per second, 24 hours per day? If not, the code is simple. I just have the most recent values in memory all the time. But maybe I ruin the chip? I use too much energy I know. But there is plenty of it in this case.
You aren't going to "wear it out" by running it when you don't need to.
The main problems are wasting power and RAM.
If you have enough of these, then the lesser problems are:
The wasted power will become heat, this may upset your temperature measurements (this is a very small amount though).
Having the DMA running will increase your interrupt latency and maybe also slow down the processor slightly, if it encounters bus contention (this only matters if you are close to capacity in these regards).
Having it running all the time may also have the advantage of more stable readings, not being perturbed turning things on and off.
I am using jsn-srf04t ranging sensor (25cm to 5m range) I want to know when it is going out of range(when under 25cm)
The problem is when it goes under 25 cm, the sensor output sometimes goes to (90cm to 95cm or 100cm to 120cm) and this cause undetectability of that it is really out of range or not!
Is there any solution?
This question is not directly related, but I thought I post a suggestion/answer anyway.
SRF04's can detect the distances as small as 3 cms. Please measure the width of the output echo pulse using an oscilloscope. It can be from 100uS to 18mS and if there is no object within its range, the echo pulse is 36ms.
If the measured pulse width from the oscilloscope is agreeing with what you say, then presumably the SRF04 is faulty, or there is a problem with its mounting etc.
If the width of the pulse is measured in uS, then dividing by 58 will give you the distance in cm, or dividing by 148 will give the distance in inches.
The SRF sensors can be triggered as fast as every 50mS, or 20 times each second. You should wait 50ms before the next trigger to ensure the ultrasonic "beep" has faded away and will not cause a false echo on the next ranging.
Otherwise, check your timer configuration. Ensure that it can measure a pulse in the order of hundreds of microseconds with at least a resolution of tens of microseconds.
If you are using this, then perhaps you are at the lowest possible level.
I'm running a test to measure the basic latency of my iPhone app, and the result was disappointing: 50ms for a play-through test app. The app just picks up mic input and plays it out using the same render callback, no other audio units or processing involved. Therefore, the results seemed too bad for such a basic scenario. I need some pointers to see if the result makes sense or I had design flaws in my test.
The basic idea of the test was to have three roles:
My finger snap as the reference sound source.
A simple iOS play-thru app (using built-in mic) as the first
listener to #1.
A Mac (with a USB mic and Audacity) as the second listener to #1 and
the only listener to the iOS output (through a speaker connected via
iOS headphone jack).
Then, with Audacity in recording mode, the Mac would pick up both the sound from my fingers and its "clone" from the iOS speaker in close range. Finally I simply visually observe the waveform in Audacity's recorded track and measure the time interval between the peaks of the two recorded snaps.
This was by no means a super accurate measurement, but at least the innate latency of the Mac recording pipeline should have been cancelled out this way. So that the error should mainly come from the peak distance measurement, which I assume should be much smaller than the audio pipeline latency and can be ignored.
I was expecting 20ms or lower latency, but clearly the result gave me 50~60ms.
My ASBD uses kAudioFormatFlagsCanonical and kAudioFormatLinearPCM as format.
50 mS is about 4 mS more than the duration of 2 audio buffers (one output, one input) of size 1024 at a sample rate of 44.1 kHz.
17 mS is around 5 mS more than the duration of 2 buffers of length 256.
So it looks like the iOS audio latency is around 5 mS plus the duration of the two buffers (the audio output buffer duration plus the time it takes to fill the input buffer) ... on your particular iOS device.
A few iOS devices may support even shorter audio buffer sizes of 128 samples.
You can use core audio and set up the audio session to have a very low latency.
You can set the buffer size to be smaller using AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration,...
Using smaller buffers causes the audio callback to happen more often while grabbing smaller chunks of audio. Keep in mind that this is merely a suggestion to the audio system. iOS will use a callback time suitable value based on your sample rate and integer powers of 2.
Once you set the buffer duration, you can get the actual buffer duration that the system will use using AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration,...
I'll summarize Paul R's comments as the answer, which has solved my problem:
50 ms corresponds to a total buffer size of around 2048 at a 44.1 kHz sample rate, which doesn't seem unreasonable given that you have both a record and a playback path.
I don't know that the buffer size is 2048, and there may be more than one buffer in your record-playback loopback test, but it seems that the effective total buffer size in you test is probably of the order of 2048, which doesn't seem unreasonable. Of course if you're only interested in record latency, as the title of your question suggests, then you'll need to find a way to tease that out separately from playback latency.
I want to add a few bytes of data to a sound file (for example a song). The sound file will be transmitted via radio to a received who uses for example the iPhone microphone to pick up the sound, and an application will show the original bytes of data. Preferably it should not be hearable for humans.
What is such technology called? Are there any applications that can do this?
Libraries/apps that can be used on iPhone?
It's audio steganography. There are algorithms to do it. Refer to here.
I've done some research, and it seems the way to go is:
Use low audio frequencies.
Spread the "bits" around randomly - do not use a pattern as it will be picked up by the listener. "White noise" is a good clue. The random pattern is known by the sender and receiver.
Use Fourier transform to pick up frequency and amplitude
Clean up input data.
Use checksum/redundancy-algorithms to compensate for loss.
I'm writing a prototype and am having a bit difficulty in picking up the right frequency as if has a ~4 Hz offset (100 Hz becomes 96.x Hz when played and picked up by the microphone).
This is not the answer, but I hope it helps.
Hey all, I've got a method of recording that writes the notes that a user plays to an array in real time. The only problem is that there is a slight delay and each sequence is noticeably slowed down when playing back. I upped the speed of playback by about 6 miliseconds, and it sounds right, but I was wondering if the delay would vary on other devices?
I've tested on an ipod touch 2nd gen, how would that preform on 3rd, and 4th as well as iphones? do I need to test on all of them and find the optimal delay variation?
Any Ideas?
More Info:
I use two NSThreads instead of timers, and fill an array with blank spots where no notes should play (I use integers, -1 is a blank). Every 0.03 seconds it adds a blank when recording. Every time the user hits a note, the most recent blank is replaced by a number 0-7. When playing back, the second thread is used, (2 threads because the second one has a shorter time interval) that has a time of 0.024. The 6 millisecond difference compensates for the delay between the recording and playback.
I assume that either the recording or playing of notes takes longer than the other, and thus creates the delay.
What I want to know is if the delay will be different on other devices, and how I should compensate for it.
Exact Solution
I may not have explained it fully, that's why this solution wasn't provided, but for anyone with a similar problem...
I played each beat similar to a midi file like so:
while playing:
do stuff to play beat
new date xyz seconds from now
new date now
while now is not > date xyz seconds from now wait.
The obvious thing that I was missing was to create the two dates BEFORE playing the beat...
D'OH!
It seems more likely to me that the additional delay is caused by the playback of the note, or other compute overhead in the second thread. Grab the wallclock time in the second thread before playing each note, and check the time difference from the last one. You will need to reduce your following delay by any excess (likely 0.006 seconds!).
The delay will be different on different generations of the iphone, but by adapting to it dynamically like this, you will be safe as long as the processing overhead is less than 0.03 seconds.
You should do the same thing in the first thread as well.
Getting high resolution timestamps - there's a a discussion on apple forums here, or this stackoverflow question.