Core Audio: Improvements to sample rate selection on desktop.

miniaudio.h

@@ -3197,6 +3197,10 @@ typedef struct
         const char* pStreamNamePlayback;
         const char* pStreamNameCapture;
     } pulse;
+    struct
+    {
+        ma_bool32 allowNominalSampleRateChange; /* Desktop only. When enabled, allows changing of the sample rate at the operating system level. */
+    } coreaudio;
 } ma_device_config;
 
 typedef struct
@@ -4601,6 +4605,13 @@ then be set directly on the structure. Below are the members of the `ma_device_c
     pulse.pStreamNameCapture
         PulseAudio only. Sets the stream name for capture.
  
+    coreaudio.allowNominalSampleRateChange
+        Core Audio only. Desktop only. When enabled, allows the sample rate of the device to be changed at the operating system level. This
+        is disabled by default in order to prevent intrusive changes to the user's system. This is useful if you want to use a sample rate
+        that is known to be natively supported by the hardware thereby avoiding the cost of resampling. When set to true, miniaudio will
+        find the closest match between the sample rate requested in the device config and the sample rates natively supported by the
+        hardware. When set to false, the sample rate currently set by the operating system will always be used.
+
 
 Once initialized, the device's config is immutable. If you need to change the config you will need to initialize a new device.
 
@@ -24816,6 +24827,8 @@ static void ma_device_uninit__coreaudio(ma_device* pDevice)
 
 typedef struct
 {
+    ma_bool32 allowNominalSampleRateChange;
+
     /* Input. */
     ma_format formatIn;
     ma_uint32 channelsIn;
@@ -24990,16 +25003,6 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
         }
         
         /*
-        Update 2020-10-10:
-        
-        I cannot remember where I read this in the documentation and I cannot find it again. For now I'm going to remove this
-        and see what the feedback from the community is like. If this results in issues we can add it back in again. The idea
-        is that the closest sample rate natively supported by the backend to the requested sample rate should be used if possible.
-        
-        Update 2020-11-06:
-        
-        I have found the documentation that talks about keeping the sample rate consistent:
-        
         Technical Note TN2091: Device input using the HAL Output Audio Unit
             https://developer.apple.com/library/archive/technotes/tn2091/_index.html
         
@@ -25016,11 +25019,34 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
         
         I have tried going against the documentation by setting the sample rate anyway, but this just results in AudioUnitRender()
         returning a result code of -10863. I have also tried changing the format directly on the input scope on the input bus, but
-        this just results in `ca_require: IsStreamFormatWritable(inScope, inElement) NotWritable` when trying to set the format. At
+        this just results in `ca_require: IsStreamFormatWritable(inScope, inElement) NotWritable` when trying to set the format.
-        this point I'm not sure how to change the sample rate on the device even if the device reports native support for it. If
+        
-        anybody has any suggestions on this please let me know.
+        Something that does seem to work, however, has been setting the nominal sample rate on the deivce object. The problem with
+        this, however, is that it actually changes the sample rate at the operating system level and not just the application. This
+        could be intrusive to the user, however, so I don't think it's wise to make this the default. Instead I'm making this a
+        configuration option. When the `coreaudio.allowNominalSampleRateChange` config option is set to true, changing the sample
+        rate will be allowed. Otherwise it'll be fixed to the current sample rate. To check the system-defined sample rate, run
+        the Audio MIDI Setup program that comes installed on macOS and observe how the sample rate changes as the sample rate is
+        changed by miniaudio.
         */
+        if (pData->allowNominalSampleRateChange) {
+            AudioValueRange sampleRateRange;
+            AudioObjectPropertyAddress propAddress;
+            
+            sampleRateRange.mMinimum = bestFormat.mSampleRate;
+            sampleRateRange.mMaximum = bestFormat.mSampleRate;
+            
+            propAddress.mSelector = kAudioDevicePropertyNominalSampleRate;
+            propAddress.mScope    = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
+            propAddress.mElement  = kAudioObjectPropertyElementMaster;
+            
+            status = ((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(sampleRateRange), &sampleRateRange);
+            if (status != noErr) {
                 bestFormat.mSampleRate = origFormat.mSampleRate;
+            }
+        } else {
+            bestFormat.mSampleRate = origFormat.mSampleRate;
+        }
         
         status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
         if (status != noErr) {
@@ -25265,6 +25291,8 @@ static ma_result ma_device_reinit_internal__coreaudio(ma_device* pDevice, ma_dev
         return MA_INVALID_ARGS;
     }
     
+    data.allowNominalSampleRateChange = MA_FALSE;   /* Don't change the nominal sample rate when switching devices. */
+
     if (deviceType == ma_device_type_capture) {
         data.formatIn               = pDevice->capture.format;
         data.channelsIn             = pDevice->capture.channels;
@@ -25368,6 +25396,7 @@ static ma_result ma_device_init__coreaudio(ma_context* pContext, const ma_device
     /* Capture needs to be initialized first. */
     if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
         ma_device_init_internal_data__coreaudio data;
+        data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
         data.formatIn                     = pConfig->capture.format;
         data.channelsIn                   = pConfig->capture.channels;
         data.sampleRateIn                 = pConfig->sampleRate;
@@ -25421,6 +25450,7 @@ static ma_result ma_device_init__coreaudio(ma_context* pContext, const ma_device
     /* Playback. */
     if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
         ma_device_init_internal_data__coreaudio data;
+        data.allowNominalSampleRateChange = pConfig->coreaudio.allowNominalSampleRateChange;
         data.formatIn                     = pConfig->playback.format;
         data.channelsIn                   = pConfig->playback.channels;
         data.sampleRateIn                 = pConfig->sampleRate;
@@ -62960,6 +62990,7 @@ v0.10.22 - TBD
   - Fix some compilation warnings on GCC and Clang relating to the Speex resampler.
   - Fix a compilation error for the Linux build when the ALSA and JACK backends are both disabled.
   - ALSA: Fix a bug in `ma_context_get_device_info()` where the PCM handle is left open in the event of an error.
+  - Core Audio: Further improvements to sample rate selection.
   - Add support for detecting default devices during device enumeration and with `ma_context_get_device_info()`.
   - Add documentation for `MA_NO_RUNTIME_LINKING`.