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Re: iOS 13 is not bit-accurate; previous versions were
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Re: iOS 13 is not bit-accurate; previous versions were


  • Subject: Re: iOS 13 is not bit-accurate; previous versions were
  • From: Robert Kukuchka via Coreaudio-api <email@hidden>
  • Date: Tue, 26 May 2020 14:01:11 -0700

Hi Andrew,

What mode are you specifying? Have you tried Measurement Mode?

~Rob

> On May 26, 2020, at 1:30 PM, Andrew Smith via Coreaudio-api
> <email@hidden> wrote:
>
> Brian,
>
> I’ve tried both allow mixing with other audio, and not, both work the same.
>
> I’m using the remote IO framework, and I will paste my main setup code below.
> One interesting thing that I’ve found is that if I select playback only mode,
> it works fine! The audio data passes through to the USB output device
> unaltered and the surround files play fine. It’s just when I add in record
> mode (using [avSession setCategory:AVAudioSessionCategoryPlayAndRecord
> mode:mode options:options error:&err]) that the output audio is no longer
> bit-accurate. I do need record and play for audio analysis of the input.
> Again this is only under iOS 13, older iOS work fine.
>
> Best,
> Andrew Smith
>
> In this code sample rate is always 48k and numChannels = 2. This is just
> copied from other sources and may certainly have issues! One thing I find
> weird is that I have to reverse the output busses when I apply the formats,
> ie I have to apply the playback format to the input bus.
>
> #define kOutputBus 0
> #define kInputBus 1
>
>    OSStatus status;
>      // Describe audio component
>     AudioComponentDescription desc;
>     desc.componentType = kAudioUnitType_Output;
>     desc.componentSubType = kAudioUnitSubType_RemoteIO;
>     desc.componentFlags = 0;
>     desc.componentFlagsMask = 0;
>     desc.componentManufacturer = kAudioUnitManufacturer_Apple;
>
>     // Get component
>     AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
>
>     // Get audio units
>     status = AudioComponentInstanceNew(inputComponent, &inRemoteIOUnitAV);
>     [AvHelper checkStatus:status msg:@"AudioComponentInstanceNew"];
>
>     UInt32 flag = 1;
>     // Enable IO for playback
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioOutputUnitProperty_EnableIO,
>                                   kAudioUnitScope_Output,
>                                   kOutputBus,
>                                   &flag,
>                                   sizeof(flag));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 2"];
>
>     // Enable IO for recording
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioOutputUnitProperty_EnableIO,
>                                   kAudioUnitScope_Input,
>                                   kInputBus,
>                                   &flag,
>                                   sizeof(flag));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 1"];
>
>
>     // Describe format
>     int numChannels = 2;
>     AudioStreamBasicDescription audioFormat;
>
>     audioFormat.mSampleRate         = auSampleRate;
>     audioFormat.mFormatID           = kAudioFormatLinearPC
>     audioFormat.mFormatFlags        = kAudioFormatFlagIsSignedInteger |
> kAudioFormatFlagIsPacked;
>     audioFormat.mBytesPerPacket     = 2 * numChannels;
>     audioFormat.mFramesPerPacket    = 1;
>     audioFormat.mBytesPerFrame      = 2 * numChannels;
>     audioFormat.mChannelsPerFrame   = numChannels;
>     audioFormat.mBitsPerChannel     = 16;
>     audioFormat.mReserved           = 0;
>     // Apply format
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioUnitProperty_StreamFormat,
>                                   kAudioUnitScope_Output,
>                                   kInputBus,
>                                   &audioFormat,
>                                   sizeof(audioFormat));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 3"];
>
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioUnitProperty_StreamFormat,
>                                   kAudioUnitScope_Input,
>                                   kOutputBus,
>                                   &audioFormat,
>                                   sizeof(audioFormat));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 4"];
>
>     // Set input callback
>     AURenderCallbackStruct callbackStruct;
>     callbackStruct.inputProc = inRenderProc.inputProc;
>     callbackStruct.inputProcRefCon = inRenderProc.inputProcRefCon;
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioOutputUnitProperty_SetInputCallback,
>                                   kAudioUnitScope_Global,
>                                   kInputBus,
>                                   &callbackStruct,
>                                   sizeof(callbackStruct));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 5"];
>
>     // Set output callback
>     callbackStruct.inputProc = outRenderProc.inputProc;
>     callbackStruct.inputProcRefCon = outRenderProc.inputProcRefCon;
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioUnitProperty_SetRenderCallback,
>                                   kAudioUnitScope_Global,
>                                   kOutputBus,
>                                   &callbackStruct,
>                                   sizeof(callbackStruct));
>     [AvHelper checkStatus:status msg:@"AudioUnitSetProperty 6"];
>
>     // Disable buffer allocation for the recorder (optional - do this if we
> want to pass in our own)
>     flag = 0;
>     status = AudioUnitSetProperty(inRemoteIOUnitAV,
>                                   kAudioUnitProperty_ShouldAllocateBuffer,
>                                   kAudioUnitScope_Output,
>                                   kInputBus,
>                                   &flag,
>                                   sizeof(flag));
>
>
>> On May 21, 2020, at 10:18 PM, Brian Willoughby <email@hidden
>> <mailto:email@hidden>> wrote:
>>
>> On macOS, there was a way to "hog" the audio interface so that no other
>> application's audio would be mixed in. Even at the same sample rate, mixing
>> anything with a surround-encoded stream would destroy it.
>>
>> Are you already using anything like this? If so, and Apple broke it, then
>> that's one thing. If Apple doesn't offer "hog" mode on iOS, then you might
>> need to look for another way to take full control of the stream. I haven't
>> written audio code for iOS in years, but I recall that applications can mark
>> themselves as input-only, output-only, input and output, and various other
>> categories. Perhaps there is a category to say that you need un-manipulated
>> streams.
>>
>> The audio objects seem to keep changing, but there are usually way to query
>> your stream type, the stream type out the output unit, and to do that for
>> the input, output, and perhaps even global scopes (I think global scope is
>> the only one that doesn't have a stream type).
>>
>> Sorry I don't have an immediate answer, but maybe you can give us a few more
>> details on which audio object frameworks you're using, and perhaps the
>> results of some code-based queries of the output stream formats.
>>
>> Brian Willoughby
>>
>>
>> On May 21, 2020, at 4:00 PM, Andrew Smith via Coreaudio-api
>> <email@hidden <mailto:email@hidden>> wrote:
>>> I’m the developer of AudioTools, over 10 years on the app store, lots of
>>> audio test & measurement functions. One in particular is a 5.1 surround
>>> signal generator, which works with our own hardware device,
>>> iAudioInterface2, which happens to have a Toslink digital audio output
>>> connector. The idea is that I pre-encode surround signal audio files in the
>>> Dolby Digital 5.1 format, open the file in the app, and break the data into
>>> packets that get formatted to Dolby specs and sent out the Toslink
>>> connector at 48k sample rate. Surprisingly, this actually works. For many
>>> years it has, anyway. I can send out a pink noise signal to a particular
>>> channel, or send an encoded music file out to the system. I am using basic
>>> remote IO audio processing here. Despite many years in audio and DSP I do
>>> not consider myself an iOS guru, I just use a simple set of iOS tools that
>>> seem to work.
>>>
>>> Starting with iOS 13 however, the surround signal generator no longer
>>> works. This scheme relies on the signal (data) not being altered by iOS at
>>> all, it must be sent through unaltered, as bit-accurate audio. In 13, I get
>>> just random noise, while the same app on iOS 11 or 12 outputs the surround
>>> data correctly. My guess is that Apple has inserted a sample rate
>>> converter, even though as far as I can tell the hardware is still running
>>> at 48k, and I am requesting 48k. I’ve posted a bug report to Apple but
>>> unsurprisingly I’ve had no response to it since I am likely the only person
>>> on Earth who cares about this. Any ideas on something to try or if perhaps
>>> anyone else has knowledge about this I would highly appreciate it!
>>>
>>> Best,
>>> Andrew Smith
>>
>
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  • Follow-Ups:
    • Re: iOS 13 is not bit-accurate; previous versions were
      • From: Andrew Smith via Coreaudio-api <email@hidden>
References: 
 >iOS 13 is not bit-accurate; previous versions were (From: Andrew Smith via Coreaudio-api <email@hidden>)
 >Re: iOS 13 is not bit-accurate; previous versions were (From: Brian Willoughby via Coreaudio-api <email@hidden>)
 >Re: iOS 13 is not bit-accurate; previous versions were (From: Andrew Smith via Coreaudio-api <email@hidden>)

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