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AUHAL: accept Vivox's playback format and trace recorded levels

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The issue #75 trace shows recording now runs, but Vivox's playback unit
(mic test loopback, other players' voices) sets interleaved Float32 stereo
at 32 kHz on input scope element 0 and got -10868. Apply the relaxed rule
to both client sides (input scope element 0, output scope element 1); the
playback path already interleaves for the device and PulseAudio resamples.

AudioUnitRender(bus 1) now traces frames and peak level (no samples) so a
log shows whether the client pulls sound or silence. The input check also
sets Vivox's playback format and expects unsigned 16-bit to be refused.

Co-Authored-By: Claude Opus 5.5 <[email protected]>
This commit is contained in:
lucannez64andClaude Opus 5.5 committed 2026-10-19 00:41:10 +02:00
1 parent 2462b2b755
commit d2ec718c48
2 files changed
+62 -24
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patches/darling.patch
+44 -24
View File
@@ -2572,7 +2572,7 @@ index ff83d1e15..1e86f9e99 100644
bool m_suspended = true;
};
diff --git a/src/CoreAudio/CoreAudioComponent/AUHAL.cpp b/src/CoreAudio/CoreAudioComponent/AUHAL.cpp
index fe4d1eaa1..1ee1da9c5 100644
index fe4d1eaa1..769b06586 100644
--- a/src/CoreAudio/CoreAudioComponent/AUHAL.cpp
+++ b/src/CoreAudio/CoreAudioComponent/AUHAL.cpp
@@ -1,3 +1,4 @@
@@ -2647,7 +2647,7 @@ index fe4d1eaa1..1ee1da9c5 100644
bool AUHAL::CanScheduleParameters() const
{
return true;
@@ -49,7 +93,33 @@ bool AUHAL::CanScheduleParameters() const
@@ -49,7 +93,36 @@ bool AUHAL::CanScheduleParameters() const
bool AUHAL::StreamFormatWritable(AudioUnitScope scope, AudioUnitElement element)
{
@@ -2656,15 +2656,18 @@ index fe4d1eaa1..1ee1da9c5 100644
+}
+
+// AUBase only accepts the canonical layout (Float32, non-interleaved). A real AUHAL
+// converts any linear PCM, and voice engines (Vivox) ask for interleaved Float32 on the
+// recording side, so element 1 accepts whatever PulseAudio records and RenderBus() copies:
+// native-endian float32 or signed 16/32-bit integers, either layout. RenderBus() pads with
+// zeros, which is only silence for float and signed samples, so unsigned (and 8-bit) formats
+// are left out; so is 24-bit (PulseAudio may read 4-byte samples while RenderBus() copies 3).
+// Element 0 (playback) keeps the canonical rule.
+// converts any linear PCM, and Vivox asks for interleaved Float32 on both client sides
+// (recording: output scope element 1; playback: input scope element 0, at its own rate).
+// Those accept what PulseAudio converts and our copies handle: native-endian float32 or
+// signed 16/32-bit integers, either layout. Missing frames are padded with zeros, which is
+// only silence for float and signed samples, so unsigned (and 8-bit) formats are left out;
+// so is 24-bit (PulseAudio may use 4-byte samples while we copy 3). Device sides keep the
+// canonical rule.
+bool AUHAL::ValidFormat(AudioUnitScope scope, AudioUnitElement element, const CAStreamBasicDescription& format)
+{
+ if (element != kInputBus || scope == kAudioUnitScope_Global)
+ const bool clientSide = (scope == kAudioUnitScope_Output && element == kInputBus)
+ || (scope == kAudioUnitScope_Input && element == kOutputBus);
+ if (!clientSide)
+ return AUBase::ValidFormat(scope, element, format);
+
+ const UInt32 flags = format.mFormatFlags;
@@ -2682,7 +2685,7 @@ index fe4d1eaa1..1ee1da9c5 100644
}
OSStatus AUHAL::Version()
@@ -57,45 +127,112 @@ OSStatus AUHAL::Version()
@@ -57,45 +130,129 @@ OSStatus AUHAL::Version()
return 1;
}
@@ -2709,11 +2712,7 @@ index fe4d1eaa1..1ee1da9c5 100644
+ std::lock_guard<std::mutex> lk(m_dataAvailableMutex);
+ const UInt32 frames = std::min<UInt32>(inNumberFrames, m_bufferUsed / frameBytes);
+ const uint8_t* source = m_buffer.get();
- // TODO: Prepare for non-interleaved audio
- UInt32 howMuch = std::min<UInt32>(abl.mBuffers[0].mDataByteSize, m_bufferUsed);
- memcpy(abl.mBuffers[0].mData, m_buffer.get(), howMuch);
- abl.mBuffers[0].mDataByteSize = howMuch;
+
+ if (planar)
+ {
+ for (UInt32 c = 0; c < abl.mNumberBuffers && c < desc.mChannelsPerFrame; c++)
@@ -2731,6 +2730,27 @@ index fe4d1eaa1..1ee1da9c5 100644
+ memcpy(destination, source, frames * frameBytes);
+ memset(destination + frames * frameBytes, 0, (inNumberFrames - frames) * frameBytes);
+ abl.mBuffers[0].mDataByteSize = inNumberFrames * frameBytes;
+ }
- // TODO: Prepare for non-interleaved audio
- UInt32 howMuch = std::min<UInt32>(abl.mBuffers[0].mDataByteSize, m_bufferUsed);
- memcpy(abl.mBuffers[0].mData, m_buffer.get(), howMuch);
- abl.mBuffers[0].mDataByteSize = howMuch;
+ // Level only (no samples): shows whether the client pulls real sound or silence.
+ static std::atomic<unsigned> renders{0};
+ const unsigned n = renders++;
+ if (rclolAudioTraceEnabled() && (n < 40 || n % 1000 == 0))
+ {
+ float peak = 0;
+ const UInt32 count = frames * desc.mChannelsPerFrame;
+ if (desc.IsFloat() && sampleBytes == 4)
+ for (UInt32 i = 0; i < count; i++)
+ peak = std::max(peak, std::abs(reinterpret_cast<const float*>(source)[i]));
+ else if (sampleBytes == 2)
+ for (UInt32 i = 0; i < count; i++)
+ peak = std::max(peak, std::abs(reinterpret_cast<const int16_t*>(source)[i] / 32768.0f));
+ rclolAudioTrace("AUDIO_INPUT_RENDER n=%u frames=%u available=%u planar=%d peak=%.4f\n",
+ n, inNumberFrames, frames, (int) planar, peak);
+ }
return noErr;
@@ -2810,7 +2830,7 @@ index fe4d1eaa1..1ee1da9c5 100644
}
m_running = m_enableOutput || m_enableInput;
@@ -111,20 +248,49 @@ OSStatus AUHAL::Stop()
@@ -111,20 +268,49 @@ OSStatus AUHAL::Stop()
{
AudioDeviceStop(m_outputDevice, m_outputProcID);
AudioDeviceDestroyIOProcID(m_outputDevice, m_outputProcID);
@@ -2861,7 +2881,7 @@ index fe4d1eaa1..1ee1da9c5 100644
case kAudioOutputUnitProperty_SetInputCallback:
{
ca_require(inDataSize == sizeof(AURenderCallbackStruct), InvalidPropertyValue);
@@ -161,7 +327,14 @@ OSStatus AUHAL::SetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
@@ -161,7 +347,14 @@ OSStatus AUHAL::SetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
ca_require(inDataSize == sizeof(AudioDeviceID), InvalidPropertyValue);
const AudioDeviceID* dev = static_cast<const AudioDeviceID*>(inData);
@@ -2877,7 +2897,7 @@ index fe4d1eaa1..1ee1da9c5 100644
{
m_outputDevice = *dev;
PropertyChanged(inID, inScope, inElement);
@@ -176,20 +349,28 @@ OSStatus AUHAL::SetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
@@ -176,20 +369,28 @@ OSStatus AUHAL::SetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
return noErr;
}
}
@@ -2908,7 +2928,7 @@ index fe4d1eaa1..1ee1da9c5 100644
}
case kAudioOutputUnitProperty_EnableIO:
{
@@ -210,13 +391,23 @@ OSStatus AUHAL::GetPropertyInfo(AudioUnitPropertyID inID, AudioUnitScope inScope
@@ -210,13 +411,23 @@ OSStatus AUHAL::GetPropertyInfo(AudioUnitPropertyID inID, AudioUnitScope inScope
return noErr;
}
}
@@ -2933,7 +2953,7 @@ index fe4d1eaa1..1ee1da9c5 100644
case kAudioOutputUnitProperty_SetInputCallback:
{
memcpy(outData, &m_outputAvailableCb, sizeof(m_outputAvailableCb));
@@ -228,12 +419,12 @@ OSStatus AUHAL::GetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
@@ -228,12 +439,12 @@ OSStatus AUHAL::GetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
if (inElement == kOutputBus)
{
UInt32 value = m_enableOutput;
@@ -2948,7 +2968,7 @@ index fe4d1eaa1..1ee1da9c5 100644
}
else
return kAudioUnitErr_InvalidElement;
@@ -241,7 +432,11 @@ OSStatus AUHAL::GetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
@@ -241,7 +452,11 @@ OSStatus AUHAL::GetProperty(AudioUnitPropertyID inID, AudioUnitScope inScope, Au
}
case kAudioOutputUnitProperty_CurrentDevice:
{
@@ -2961,7 +2981,7 @@ index fe4d1eaa1..1ee1da9c5 100644
{
memcpy(outData, &m_outputDevice, sizeof(m_outputDevice));
}
@@ -277,9 +472,89 @@ OSStatus AUHAL::recordCallback(AudioObjectID inObjectID,
@@ -277,9 +492,89 @@ OSStatus AUHAL::recordCallback(AudioObjectID inObjectID,
return This->doRecord(inNow, inInputData, inInputTime);
}
@@ -3051,7 +3071,7 @@ index fe4d1eaa1..1ee1da9c5 100644
if (!HasInput(0))
{
// std::cerr << "No connection\n";
@@ -290,9 +565,75 @@ OSStatus AUHAL::doPlayback(const AudioTimeStamp* inNow, AudioBufferList* outOutp
@@ -290,9 +585,75 @@ OSStatus AUHAL::doPlayback(const AudioTimeStamp* inNow, AudioBufferList* outOutp
AudioUnitRenderActionFlags flags = kAudioUnitRenderAction_PreRender;
const CAStreamBasicDescription& desc = GetStreamFormat(kAudioUnitScope_Input, kOutputBus);
@@ -3130,7 +3150,7 @@ index fe4d1eaa1..1ee1da9c5 100644
// std::cout << "Pull result: " << result << std::endl;
// std::cout << "Bytes: " << outOutputData->mBuffers[0].mDataByteSize << std::endl;
// m_auhalData.write((char*) outOutputData->mBuffers[0].mData, outOutputData->mBuffers[0].mDataByteSize);
@@ -303,30 +644,46 @@ OSStatus AUHAL::doPlayback(const AudioTimeStamp* inNow, AudioBufferList* outOutp
@@ -303,30 +664,46 @@ OSStatus AUHAL::doPlayback(const AudioTimeStamp* inNow, AudioBufferList* outOutp
OSStatus AUHAL::doRecord(const AudioTimeStamp* inNow, const AudioBufferList* inInputData, const AudioTimeStamp* inInputTime)
{
tools/riot-audio-input-check.cpp
+18
View File
@@ -142,6 +142,23 @@ static int record(AudioDeviceID id,bool nonInterleaved,unsigned channels,unsigne
|| seconds<milliseconds/2000.0;
}
// Vivox's playback unit sets interleaved Float32 stereo at 32 kHz on input scope element 0;
// only the format is checked, nothing is played.
static int playbackFormat() {
AudioComponentDescription description={kAudioUnitType_Output,kAudioUnitSubType_HALOutput,kAudioUnitManufacturer_Apple,0,0};
AudioComponent component=AudioComponentFindNext(nullptr,&description);
AudioUnit playback=nullptr;
if(!component || AudioComponentInstanceNew(component,&playback) || !playback) return 1;
AudioStreamBasicDescription format={32000,kAudioFormatLinearPCM,kAudioFormatFlagIsFloat|kAudioFormatFlagIsPacked,8,1,8,2,32,0};
OSStatus status=AudioUnitSetProperty(playback,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input,0,&format,sizeof(format));
AudioStreamBasicDescription unsignedFormat={44100,kAudioFormatLinearPCM,kAudioFormatFlagIsPacked,4,1,4,2,16,0};
OSStatus rejected=AudioUnitSetProperty(playback,kAudioUnitProperty_StreamFormat,kAudioUnitScope_Input,0,&unsignedFormat,sizeof(unsignedFormat));
AudioComponentInstanceDispose(playback);
printf("AUDIO_PLAYBACK_FORMAT interleaved_status=%d unsigned_status=%d\n",(int)status,(int)rejected);
fflush(stdout);
return status!=noErr || rejected==noErr;
}
int main() {
int failures=0;
failures+=authorization()!=0;
@@ -156,6 +173,7 @@ int main() {
result=record(id,false,2,400);
if(result) printf("AUDIO_INPUT_RECORD failed step=%d interleaved_stereo=1\n",result);
failures+=result!=0;
failures+=playbackFormat()!=0;
printf("AUDIO_INPUT_CHECK failures=%d\n",failures);
return failures;
}