This fixes a few issues with the OS X version of TAAE2. With these changes, I was able to get a basic Mac app working.

The first fix is in -[AEIOAudioUnit streamFormatForDefaultDeviceScope:], it was trying write an AudioStreamBasicDescription into an AudioDeviceID (likely because it was copy-pasted from the method above it).

The second fix guards out certain parts of -[AEIOAudioUnit setup:] that don't apply on OS X. It skips setting kAudioOutputUnitProperty_EnableIO (only applies to AURemoteIO unit on iOS) and skips setting the render callback or input callback, depending on whether we're setting up for input or output. I'm not sure if the conditions I'm using are ideal (if (TARGET_OS_IPHONE || self.outputEnabled) etc), so that might need to be tweaked to always check for input/output enabled, even on iOS.

I would like to see a UInt64 frameIndex added to the rendercontext which basically would be the equivalent of timestamp->mSampleTime minus the pitfalls of the AudioTimeStamp and floatingpoint bs.

Hey Michael,

First off thanks for all of the hard work that you put into the original TAAE, this is my first time using it but have been a fan and following the progress over the years. Was really interested in this re-write, so wanted to start testing it.

I had no problem installing the original TAAE via Cocoapods in a project a few minutes ago, but having trouble installing this one. I dragged the files over from the sample, but i'm getting "file not found" errors in TheAmazingAudioEngine.h file.

Am I understanding correctly that this is a new library, or is it still utilizing the old one? Would love to install this ASAP so I can continue testing.

Thanks.

Hey Michael,

TAAE2 Lib seems to have a bug with the headphones. You can reproduce easily the bug with your sample. Just kill the sample app, Plug your headphones launch the sample app and play a beat The beat seems to be 2 times faster

I cleaned up the files a bit as per your suggestions first, then merged just the AEMeteringModule.h/m files into master. I updated TheAmazingAudioEngine.h to import the new files.

I noticed some odd behavior when using an AEArray inside a renderer.block today (which may be intentional actually). Let me know if this is actually desired behavior though, if not, it should be a pretty easy fix that I can throw into a PR.

1. Create an AEArray of NSNumbers of type AESeconds for demonstration purposes

self.durationArray = [[AEArray alloc] init]; NSMutableArray *intervalArray = [[NSMutableArray alloc] init]; for (int i = 0; i < 3; i++) { [intervalArray addObject:@((AESeconds)i)]; } [self.durationArray updateWithContentsOfArray:intervalArray];

2. Now if you use a subscript to access something out of bounds of the array, I would expect an exception to be thrown. Instead, it appears that the thread just gets locked and the execution stops at the line in which the index is being accessed out of bounds. For instance, if I put a breakpoint on the following line, which is out of bounds of the array, the thread does not continue past that line of execution:

AESeconds seconds = [self.durationArray[6] doubleValue];

Is this expected or should there be an exception thrown? Maybe I'm just getting odd behavior since I have only tested this inside the renderer.block which is happening on the core audio thread, so you may have some insight into what is going on. Thanks!

The docs for AEManagedValueCommitPendingUpdates state:

"If you are not using AEAudioUnitOutput, then you must call the AEManagedValueCommitPendingUpdates function at the beginning of your main render loop."

But in reading AEAudioFileOutput, I don't see where AEManagedValueCommitPendingUpdates is called. Should it be called somewhere near line 71 of AEAudioFileOutput.m? (Top of the while loop inside of |AEAudioFileOutput runForDuration:completionBlock:|

Hey,

I get a crash when I use AEAudioFilePlayerModule to play audio stored in a file.

Steps to reproduce:

1. Start playing content of the audio file using AEAudioFilePlayerModule class.
2. Set all references to the instance of the AEAudioFilePlayerModule class to nil in the completion block of the aforementioned class. In most cases there application keeps only one instance to that object.

Block of the renderer looks in the following way:

__weak typeof(self) weakSelf = self;
_renderer.block = ^(const AERenderContext * _Nonnull context) {
     __strong typeof(weakSelf) self = weakSelf;
    AEModuleProcess(self.audioFilePlayerModule, context);
    AERenderContextOutput(context, 1);
}

Result:

1. Application crashes.

Expected result:

1. No crash. 😀

I ran into a problem of frame sizes being a power of 2 on iOS (tested on iPhone 6+).

For example, setting setPreferredIOBufferDuration:0.01 in "TAAESample/Classes/AEAudioController.m" will result in a buffer size of 512 on the iPhone 6+ (sample rate is 44100), which is ~11.6ms of data vs the 10ms that I'd like the buffer to contain (so a buffer size of 441).

Is this possible? If so, how would I force each buffer duration to be the specified length? Basically, I'd like to get 10ms buffers of microphone data.

Hey, your library is really interesting.

The only problem I found was the README.md, which lacks information. I created this iOS Open source Readme Template so you can take a look on how to better organize.

If you want, I can help you to do it.

What are your thoughts?

I think the library has a main issue which that is impossible to record only the input, it causes a EXC_BAD_ACCESS on "ExtAudioFileWriteAsync"

i need play 2 beat with different time ,first on the 1s, tow with 3s . how to use playAtTime.

i try use

 [loop1 playAtTime:AETimeStampNone];

 AudioTimeStamp time = AETimeStampWithSamples(5 *self.saveRenderer.sampleRate);

[loop2 playAtTime:time];

loop1 work done! but loop2 it not work!

Hi, I'm aware of the retired situation of TAAE2, but I wonder if someone could help making a podspec file for TAAE2 similar to the podspec file in TAAE 1
thank you!

Hello guys,

Is there anyone who converted this library into React Native?

I would love to use this on React Native... https://facebook.github.io/react-native/

I am getting echo effect when recording from Mic. But when record using only instruments, it's don't have any echo effect. How to remove echo effect. Attached sample file. Recording.m4a.zip

Is it possible to access rear mic and do FFT with audio data on any of the mic data? Let me know. I came across your remoteio page and I am trying to record audio data but have not been successful. I am using auriotouch source code and I want to record and play audio data from rear microphone..