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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: Andrew Smith via Coreaudio-api <email@hidden>
  • Date: Tue, 26 May 2020 14:02:07 -0700

Hi Robert,

Yes I am in measurement mode already.

Best,
Andrew Smith





> On May 26, 2020, at 2:01 PM, Robert Kukuchka <email@hidden> wrote:
>
> 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 <mailto: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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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>)
 >Re: iOS 13 is not bit-accurate; previous versions were (From: Robert Kukuchka via Coreaudio-api <email@hidden>)

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