On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely.
Any ideas what could be going wrong?
Here's a minimum code example to demonstrate:
#include <AudioToolbox/AudioToolbox.h>
#include <stdint.h>
#define RENDER_BUFFER_COUNT 2
#define RENDER_FRAMES_PER_BUFFER 128
// mono linear PCM audio data at 48kHz
#define RENDER_SAMPLE_RATE 48000
#define RENDER_CHANNEL_COUNT 1
#define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32))
void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount)
{
static bool isInverted = false;
float scalar = isInverted ? -1.f : 1.f;
for (uint32_t frame = 0; frame < frameCount; ++frame)
{
for (uint32_t channel = 0; channel < channelCount; ++channel)
{
// series of ramps, alternating up and down.
outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount);
}
}
isInverted = !isInverted;
}
AudioStreamBasicDescription coreAudioDesc = { 0 };
AudioQueueRef coreAudioQueue = NULL;
AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL };
void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer)
{
// 0's here indicate no fancy packet magic
AudioQueueEnqueueBuffer(queue, buffer, 0, 0);
}
int main(void)
{
const UInt32 BytesPerSample = sizeof(float);
coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE;
coreAudioDesc.mFormatID = kAudioFormatLinearPCM;
coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked;
coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample;
coreAudioDesc.mFramesPerPacket = 1;
coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample;
coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT;
coreAudioDesc.mBitsPerChannel = BytesPerSample * 8;
coreAudioQueue = NULL;
OSStatus result;
// most of the 0 and NULL params here are for compressed sound formats etc.
result = AudioQueueNewOutput(&coreAudioDesc, &coreAudioCallback, NULL, 0, 0, 0, &coreAudioQueue);
if (result != noErr)
{
assert(false == "AudioQueueNewOutput failed!");
abort();
}
for (int i = 0; i < RENDER_BUFFER_COUNT; ++i)
{
uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER;
result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &(coreAudioBuffers[i]));
if (result != noErr)
{
assert(false == "AudioQueueAllocateBuffer failed!");
abort();
}
}
for (int i = 0; i < RENDER_BUFFER_COUNT; ++i)
{
RenderAudioSaw(coreAudioBuffers[i]->mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT);
coreAudioBuffers[i]->mAudioDataByteSize = coreAudioBuffers[i]->mAudioDataBytesCapacity;
AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0);
}
AudioQueueStart(coreAudioQueue, NULL);
sleep(10); // some time to hear the audio
AudioQueueStop(coreAudioQueue, true);
AudioQueueDispose(coreAudioQueue, true);
return 0;
}
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