I want to record audio from an iPhone microphone and write those samples to a file. Looking at the documentation, it's not clear to me if I can simply perform the write operation inside the render callback function of the Remote IO unit? Or if I instead need to attach a generic output AU and write the samples coming out of that unit(?) The latter implies more overhead in terms of setting up an AUGraph, AUNodes etc, so I'd prefer the former.
You can do it in your Input Callback, using the ExtAudioFile API (and ExtAudioFileWriteAsync in particular). The "async" bit is what makes it viable in the realtime input callback.
See this answer for more, as it's quite a similar setup.
ExtAudioFileWriteAsync docs are here.
Related
I want to generate different types of traffic for analyzing OFDMA transmission in NS3. How can I simulate video and audio calls?
The first three options that come to mind are:
If you want to be as close to reality as possible, try out the Direct Code Execution (DCE) Module. I've never used it, so I'm not sure how well it's supported.
Use the OnOffApplication. The OnOffApplication allows you to set onTime, offTime, and a DataRate (among other variables). You can determine the rate at which data is sent for your audio or video program, and then provide those rates to the OnOffApplication. You may find the OnOffHelper convenient to set various parameters of an OnOffApplication.
Create your own Application. This option may be of particular interest since you could simulate variable bitrate audio/video calls. If you choose this option, I highly suggest you checkout the ns-3 tutorial for the walkthrough of fifth.cc to learn more about how to create your own Application.
The second option is probably easiest to use, but may not be as accurate as the first, or as flexible as the third.
I'd like to know if it's possible to record an audio playback into a file in ios without using microphone. In another word, is it possible to capture the audio playback "raw data" into a file?
I tried the AVAudioRecorder, but there is no option to record without using the microphone. Any idea? Thanks
Yes!
Your question is too generic, and the above answers it.
But, more specifically, as with most things there are several approaches possible, the final solution will depend on your needs. For example, if you're constrained to, or have a preference for "playing" via AVPlayer, you can use an MTAudioProcessingTap (See apple's AudioTapProcessor example) to gain access to the "Raw Data". Similarly, you can build an AudioGraph and use an AudioUnit as the "player" along with Audio File Services to both read (which will give you access to the "Raw Data") and write the file. There are various alternatives depending on your needs and, in some cases, your preference. Providing some code or a better explanation of your needs, might solicit a more comprehensive answer.
I am using the ExtAudioFile interface to decode audio, and would like to know whether it ever makes use of hardware-assisted audio decoding.
The docs for the kExtAudioFileProperty_CodecManufacturer property say:
Use this property in iOS to choose between a hardware or software encoder, by specifying kAppleHardwareAudioCodecManufacturer or kAppleSoftwareAudioCodecManufacturer.
This seems to indicate that ExtAudioFile can indeed make use of decoding hardware.
Elsewhere, in the Audio Format Services docs, I find:
Hardware-based codecs can be used only when playing or recording using Audio Queue Services or using interfaces, such as AV Foundation, which use Audio Queue Services. In particular, you cannot use hardware-based audio codecs with OpenAL or when using the I/O audio unit.
... which is not completely clear; if ExtAudioFile uses Audio Queue Services in its implementation, then maybe it can make use of hardware, but we don't actually know how it is implemented.
I tried to test at runtime whether the hardware was being used, but this itself proved difficult. One method, given in the Audio Format Services reference, is to use AudioFormatGetProperty to test the kAudioFormatProperty_HardwareCodecCapabilities property. But the sample code does not work, always returning kAudioFormatUnsupportedPropertyError. (After searching online, I found a few other questions from people who have this problem, but no reports of ever using it successfully.)
So... I'm wondering if anyone knows of any way to test for certain whether the hardware decoder is currently active (in which case I could test for myself whether ExtAudioFile is using it). Or alternatively if anyone has any definitive knowledge of whether ExtAudioFile does use hardware (not based solely on the vague mentions in the Apple docs).
Specifying kAppleHardwareAudioCodecManufacturer with ExtAudioFileSetProperty() appears to enable hardware decoding, because it fails with kAudioConverterErr_HardwareInUse when there is already one audio file open with that codec set, and with kAudioQueueErr_InvalidCodecAccess when the audio category is set to one that does not (according to the documentation) enable hardware decoding.
However, after profiling with Instruments, I found that performance was slightly worse with hardware decoding enabled, which I still can't explain...
guys.
I'm working on some audio services on iOS.
I trying to search any examples or tutorials about
how audio service or stream can read a existing audio file than
process something like filter, than write another file.
Is there any body who can help me?
Dirac3LE (by Stephan M. Bernsee) is a great library for this job.
There are examples and manual included in the download.
It is particulary inteded for time and pitch manipulation
but in your case you'll be interested in its EAFRead and EAFWrite
classes.
If you want to get familiar with the lower level library that you can also use for microphone input/sound output, and that you can get raw samples into and out of, I would suggest taking a look at Audio Queue Services.
I used it in my side project to get audio from the microphone, and I also wrote some code you might find useful to do fast vectorized, FFT based FIR filtering on input audio. You can find the code here https://github.com/jamescarlson/FreeAPRS
I want to make a really simple synth.
In short, i want to play a wav file, and have it loop at certain points until touch is released.
I am looking for some example code, (doesn't need to be free).
Sorry for such a basic question, i have been googling this, though there seems to be nothing on this exact topic, unless I'm missing some important term.
Also, is what i'm describing, a wavetable synth, or a soundboard?
I'd call it a sampler.
Here's a sample project that will get you started:
https://sites.google.com/site/iphonecoreaudiodevelopment/remoteio-playback
See also:
The Audio Programming Book
The Core Audio Book
A sample project of mine
You need to store the sound data in memory, and have some sort of read() command that fills an array of bytes to send to the sound card. The read() command has to keep track of its position between reads, so a persistent pointer must be maintained. You will test the position of the pointer and see if you've reached the end or not, and reset to the beginning when needed.
The specifics are going to depend upon your chosen language, of course.
I did this with Java, with the added the possibility of playback at different speeds.
http://www.hexara.com/VSL/VSL2.htm
It's a little laggy. I've learned a bit since posting that, but haven't gone back to fix it yet. The program asks permission and has you load a wav file from your computer. It should be 16-bit, stereo, 44100fps, little-endian.
WaveTable synthesis is a bit different, in that only a single iteration of a wave is stored and used as source data.
Here is a short discussion, from Stanford's CCRMA website:
https://ccrma.stanford.edu/~bilbao/booktop/node9.html
I used this method to make a Java "Theremin"
http://www.hexara.com/VSL/JTheremin.htm
With a WaveTable, you decide on the size of the array. If it is a power of 2, one can bitmask the pointer after every increment, which is faster than doing a comparison and reset.