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miniaudio
Commit 53bbca81 authored by thedmd's avatar thedmd Committed by David Reid
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Remove trailing spaces.

parent e531dd25
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miniaudio.h
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......@@ -38,7 +38,7 @@ A config/init pattern is used throughout the entire library. The idea is that yo
object and pass that into the initialization routine. The advantage to this system is that the
config object can be initialized with logical defaults and new properties added to it without
breaking the API. The config object can be allocated on the stack and does not need to be
maintained after initialization of the corresponding object.
maintained after initialization of the corresponding object.
1.1. Low Level API
......@@ -363,7 +363,7 @@ initialized. The easiest but least flexible way of playing a sound is like so:
This plays what miniaudio calls an "inline" sound. It plays the sound once, and then puts the
internal sound up for recycling. The last parameter is used to specify which sound group the sound
should be associated with which will be explained later. This particular way of playing a sound is
simple, but lacks flexibility and features. A more flexible way of playing a sound is to first
simple, but lacks flexibility and features. A more flexible way of playing a sound is to first
initialize a sound:
```c
......@@ -822,7 +822,7 @@ retrieved like so:
ma_uint32 channels;
ma_uint32 sampleRate;
ma_channel channelMap[MA_MAX_CHANNELS];
result = ma_data_source_get_data_format(pDataSource, &format, &channels, &sampleRate, channelMap, MA_MAX_CHANNELS);
if (result != MA_SUCCESS) {
return result; // Failed to retrieve data format.
......@@ -878,7 +878,7 @@ To do this, you can use chaining:
```
In the example above we're using decoders. When reading from a chain, you always want to read from
the top level data source in the chain. In the example above, `decoder1` is the top level data
the top level data source in the chain. In the example above, `decoder1` is the top level data
source in the chain. When `decoder1` reaches the end, `decoder2` will start seamlessly without any
gaps.
......@@ -971,7 +971,7 @@ base object (`ma_data_source_base`):
void my_data_source_uninit(my_data_source* pMyDataSource)
{
// ... do the uninitialization of your custom data source here ...
// You must uninitialize the base data source.
ma_data_source_uninit(&pMyDataSource->base);
}
......@@ -2561,7 +2561,7 @@ The `ma_decoding_backend_vtable` vtable has the following functions:
```
onInit
onInitFile
onInitFile
onInitFileW
onInitMemory
onUninit
......@@ -3689,7 +3689,7 @@ extern "C" {
#pragma GCC diagnostic ignored "-Wc11-extensions" /* anonymous unions are a C11 extension */
#endif
#endif
#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
......@@ -4089,7 +4089,7 @@ typedef enum
#define MA_MIN_CHANNELS 1
#ifndef MA_MAX_CHANNELS
#ifndef MA_MAX_CHANNELS
#define MA_MAX_CHANNELS 254
#endif
......@@ -18237,7 +18237,7 @@ static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void*
/* The intermediary buffer has just been filled. */
pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap;
}
}
}
}
/* If we're in duplex mode we might need to do a refill of the data. */
......@@ -29558,7 +29558,7 @@ static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_confi
sampleRate = pDescriptorCapture->sampleRate;
}
result = ma_init_pa_mainloop_and_pa_context__pulse(pDevice->pContext, pDevice->pContext->pulse.pApplicationName, pDevice->pContext->pulse.pServerName, MA_FALSE, &pDevice->pulse.pMainLoop, &pDevice->pulse.pPulseContext);
if (result != MA_SUCCESS) {
......@@ -32696,7 +32696,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed to allocate AudioBufferList for capture.\n");
return noErr;
}
pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
MA_ASSERT(pRenderedBufferList);
......@@ -33134,7 +33134,7 @@ static ma_result ma_device__untrack__coreaudio(ma_device* pDevice)
*/
ma_device__on_notification_interruption_began(m_pDevice);
} break;
case AVAudioSessionInterruptionTypeEnded:
{
ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeEnded\n");
......@@ -33188,7 +33188,7 @@ static ma_result ma_device__untrack__coreaudio(ma_device* pDevice)
}
ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_DEBUG, "[Core Audio] Changing Route. inputNumberChannels=%d; outputNumberOfChannels=%d\n", (int)pSession.inputNumberOfChannels, (int)pSession.outputNumberOfChannels);
/* Let the application know about the route change. */
ma_device__on_notification_rerouted(m_pDevice);
}
......@@ -33561,7 +33561,7 @@ static ma_result ma_device_init_internal__coreaudio(ma_context* pContext, ma_dev
@autoreleasepool {
AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
MA_ASSERT(pAudioSession != NULL);
[pAudioSession setPreferredIOBufferDuration:((float)actualPeriodSizeInFrames / pAudioSession.sampleRate) error:nil];
actualPeriodSizeInFrames = ma_next_power_of_2((ma_uint32)(pAudioSession.IOBufferDuration * pAudioSession.sampleRate));
}
......@@ -33802,7 +33802,7 @@ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_c
#if defined(MA_APPLE_DESKTOP)
ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDCapture, sizeof(pDevice->capture.id.coreaudio), pDevice->capture.id.coreaudio);
/*
If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
switch the device in the background.
......@@ -33866,7 +33866,7 @@ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_c
#if defined(MA_APPLE_DESKTOP)
ma_get_AudioObject_uid(pDevice->pContext, pDevice->coreaudio.deviceObjectIDPlayback, sizeof(pDevice->playback.id.coreaudio), pDevice->playback.id.coreaudio);
/*
If we are using the default device we'll need to listen for changes to the system's default device so we can seemlessly
switch the device in the background.
......@@ -37454,7 +37454,7 @@ static ma_result ma_context_init__aaudio(ma_context* pContext, const ma_context_
return result;
}
}
(void)pConfig;
return MA_SUCCESS;
......@@ -39948,7 +39948,7 @@ MA_API ma_result ma_device_job_thread_init(const ma_device_job_thread_config* pC
/* Initialize the job queue before the thread to ensure it's in a valid state. */
jobQueueConfig = ma_job_queue_config_init(pConfig->jobQueueFlags, pConfig->jobQueueCapacity);
jobQueueConfig = ma_job_queue_config_init(pConfig->jobQueueFlags, pConfig->jobQueueCapacity);
result = ma_job_queue_init(&jobQueueConfig, pAllocationCallbacks, &pJobThread->jobQueue);
if (result != MA_SUCCESS) {
......@@ -40756,7 +40756,7 @@ MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pC
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_uint64 intermediaryBufferSizeInBytes;
pDevice->playback.intermediaryBufferLen = 0;
if (pConfig->deviceType == ma_device_type_duplex) {
pDevice->playback.intermediaryBufferCap = pDevice->capture.intermediaryBufferCap; /* In duplex mode, make sure the intermediary buffer is always the same size as the capture side. */
......@@ -40768,7 +40768,7 @@ MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pC
}
intermediaryBufferSizeInBytes = pDevice->playback.intermediaryBufferCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
pDevice->playback.pIntermediaryBuffer = ma_malloc((size_t)intermediaryBufferSizeInBytes, &pContext->allocationCallbacks);
if (pDevice->playback.pIntermediaryBuffer == NULL) {
ma_device_uninit(pDevice);
......@@ -51983,7 +51983,7 @@ MA_API ma_result ma_channel_converter_init_preallocated(const ma_channel_convert
/*
We now need to fill out our weights table. This is determined by the mixing mode.
*/
/* In all cases we need to make sure all channels that are present in both channel maps have a 1:1 mapping. */
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
ma_channel channelPosIn = ma_channel_map_get_channel(pConverter->pChannelMapIn, pConverter->channelsIn, iChannelIn);
......@@ -67080,7 +67080,7 @@ MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_man
pDataBuffer->seekToCursorOnNextRead = MA_TRUE; /* Keep the seek scheduled. We just haven't loaded enough data yet to do the seek properly. */
return MA_BUSY;
}
return result;
}
}
......@@ -67626,7 +67626,7 @@ MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pReso
config.pFilePath = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
}
......@@ -67638,7 +67638,7 @@ MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pRe
config.pFilePathW = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
}
......@@ -68751,7 +68751,7 @@ static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob
There is a hole between here and the where the data connector is initialized where the data
buffer node may have finished initializing. We need to check for this by checking the result of
the data buffer node and whether or not we had an unknown data supply type at the time of
trying to initialize the data connector.
trying to initialize the data connector.
*/
result = ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode);
if (result == MA_BUSY || (result == MA_SUCCESS && isConnectorInitialized == MA_FALSE && dataSupplyType == ma_resource_manager_data_supply_type_unknown)) {
......@@ -70867,7 +70867,7 @@ static ma_result ma_node_read_pcm_frames(ma_node* pNode, ma_uint32 outputBusInde
ma_node_output_bus_set_has_read(&pNodeBase->pOutputBuses[outputBusIndex], MA_TRUE);
}
}
/* Apply volume, if necessary. */
ma_apply_volume_factor_f32(pFramesOut, totalFramesRead * ma_node_get_output_channels(pNodeBase, outputBusIndex), ma_node_output_bus_get_volume(&pNodeBase->pOutputBuses[outputBusIndex]));
......@@ -72470,7 +72470,7 @@ static ma_result ma_engine_node_get_heap_layout(const ma_engine_node_config* pCo
/* Resmapler. */
resamplerConfig = ma_linear_resampler_config_init(ma_format_f32, channelsIn, 1, 1); /* Input and output sample rates don't affect the calculation of the heap size. */
resamplerConfig.lpfOrder = 0;
result = ma_linear_resampler_get_heap_size(&resamplerConfig, &tempHeapSize);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the size of the heap for the resampler. */
......@@ -72706,7 +72706,7 @@ MA_API ma_sound_config ma_sound_config_init_2(ma_engine* pEngine)
} else {
config.monoExpansionMode = ma_mono_expansion_mode_default;
}
config.rangeEndInPCMFrames = ~((ma_uint64)0);
config.loopPointEndInPCMFrames = ~((ma_uint64)0);
......@@ -72814,7 +72814,7 @@ MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEng
#if !defined(MA_NO_DEVICE_IO)
{
pEngine->pDevice = engineConfig.pDevice;
/* If we don't have a device, we need one. */
if (pEngine->pDevice == NULL && engineConfig.noDevice == MA_FALSE) {
ma_device_config deviceConfig;
......@@ -73735,7 +73735,7 @@ MA_API ma_result ma_sound_init_from_file(ma_engine* pEngine, const char* pFilePa
if (pFilePath == NULL) {
return MA_INVALID_ARGS;
}
config = ma_sound_config_init_2(pEngine);
config.pFilePath = pFilePath;
config.flags = flags;
......@@ -73752,7 +73752,7 @@ MA_API ma_result ma_sound_init_from_file_w(ma_engine* pEngine, const wchar_t* pF
if (pFilePath == NULL) {
return MA_INVALID_ARGS;
}
config = ma_sound_config_init_2(pEngine);
config.pFilePathW = pFilePath;
config.flags = flags;
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