mirror of
https://github.com/mackron/miniaudio.git
synced 2026-04-22 00:06:59 +02:00
Update miniaudio.h
fix typos
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+6
-6
@@ -21574,7 +21574,7 @@ static ma_result ma_context_get_device_id_from_MMDevice__wasapi(ma_context* pCon
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size_t idlen = ma_strlen_WCHAR(pDeviceIDString);
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if (idlen+1 > ma_countof(pDeviceID->wasapi)) {
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ma_CoTaskMemFree(pContext, pDeviceIDString);
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MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must haved change and is too long to fit in our fixed sized buffer. */
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MA_ASSERT(MA_FALSE); /* NOTE: If this is triggered, please report it. It means the format of the ID must have changed and is too long to fit in our fixed sized buffer. */
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return MA_ERROR;
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}
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@@ -28705,7 +28705,7 @@ PulseAudio Backend
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******************************************************************************/
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#ifdef MA_HAS_PULSEAUDIO
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/*
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The PulseAudio API, along with Apple's Core Audio, is the worst of the maintream audio APIs. This is a brief description of what's going on
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The PulseAudio API, along with Apple's Core Audio, is the worst of the mainstream audio APIs. This is a brief description of what's going on
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in the PulseAudio backend. I apologize if this gets a bit ranty for your liking - you might want to skip this discussion.
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PulseAudio has something they call the "Simple API", which unfortunately isn't suitable for miniaudio. I've not seen anywhere where it
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@@ -33551,7 +33551,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl
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}
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} else {
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/* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */
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MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should heve been validated at initialization time. */
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MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */
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/*
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For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
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@@ -33882,7 +33882,7 @@ static ma_result ma_context__init_device_tracking__coreaudio(ma_context* pContex
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ma_spinlock_lock(&g_DeviceTrackingInitLock_CoreAudio);
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{
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/* Don't do anything if we've already initializd device tracking. */
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/* Don't do anything if we've already initialized device tracking. */
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if (g_DeviceTrackingInitCounter_CoreAudio == 0) {
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AudioObjectPropertyAddress propAddress;
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propAddress.mScope = kAudioObjectPropertyScopeGlobal;
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@@ -67938,7 +67938,7 @@ static void ma_resource_manager_delete_all_data_buffer_nodes(ma_resource_manager
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ma_resource_manager_data_buffer_node* pDataBufferNode = pResourceManager->pRootDataBufferNode;
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ma_resource_manager_data_buffer_node_remove(pResourceManager, pDataBufferNode);
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/* The data buffer has been removed from the BST, so now we need to free it's data. */
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/* The data buffer has been removed from the BST, so now we need to free its data. */
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ma_resource_manager_data_buffer_node_free(pResourceManager, pDataBufferNode);
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}
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}
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@@ -74425,7 +74425,7 @@ static ma_result ma_engine_node_set_volume(ma_engine_node* pEngineNode, float vo
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/* If we're not smoothing we should bypass the volume gainer entirely. */
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if (pEngineNode->volumeSmoothTimeInPCMFrames == 0) {
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/* We should always have an active spatializer because it can be enabled and disabled dynamically. We can just use that for hodling our volume. */
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/* We should always have an active spatializer because it can be enabled and disabled dynamically. We can just use that for holding our volume. */
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ma_spatializer_set_master_volume(&pEngineNode->spatializer, volume);
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} else {
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/* We're using volume smoothing, so apply the master volume to the gainer. */
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