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Formatting: Remove double-dots from comments and other strings

This commit is contained in:
reuk 2025-11-17 19:00:05 +00:00
parent 82dc6d1c7e
commit 83e5264c86
No known key found for this signature in database
209 changed files with 508 additions and 509 deletions

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@ -1350,7 +1350,7 @@ double AudioDeviceManager::LevelMeter::getCurrentLevel() const noexcept
void AudioDeviceManager::playTestSound()
{
{ // cunningly nested to swap, unlock and delete in that order.
{ // cunningly nested to swap, unlock and delete in that order
std::unique_ptr<AudioBuffer<float>> oldSound;
{

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@ -105,7 +105,7 @@ void MidiMessageCollector::removeNextBlockOfMessages (MidiBuffer& destBuffer,
if (numSourceSamples > numSamples)
{
// if our list of events is longer than the buffer we're being
// asked for, scale them down to squeeze them all in..
// asked for, scale them down to squeeze them all in
const int maxBlockLengthToUse = numSamples << 5;
auto iter = incomingMessages.cbegin();

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@ -220,7 +220,7 @@ public:
return false;
}
if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw..
if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw
isInterleaved = true;
else if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_NONINTERLEAVED) >= 0)
isInterleaved = false;
@ -286,7 +286,7 @@ public:
|| JUCE_ALSA_FAILED (snd_pcm_hw_params_get_periods (hwParams, &periods, &dir)))
latency = 0;
else
latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency..)
latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency)
JUCE_ALSA_LOG ("frames: " << (int) frames << ", periods: " << (int) periods
<< ", samplesPerPeriod: " << (int) samplesPerPeriod);
@ -551,7 +551,7 @@ public:
currentOutputChans.clear();
// Note that the input device is opened before an output, because we've heard
// of drivers where doing it in the reverse order mysteriously fails.. If this
// of drivers where doing it in the reverse order mysteriously fails. If this
// order also causes problems, let us know and we'll see if we can find a compromise!
if (inputChannelDataForCallback.size() > 0 && inputId.isNotEmpty())
@ -660,7 +660,7 @@ public:
if ((! waitForThreadToExit (400)) && audioIoInProgress && numCallbacks == callbacksToStop)
{
JUCE_ALSA_LOG ("Thread is stuck in i/o.. Is pulseaudio suspended?");
JUCE_ALSA_LOG ("Thread is stuck in i/o. Is pulseaudio suspended?");
if (outputDevice != nullptr) outputDevice->closeNow();
if (inputDevice != nullptr) inputDevice->closeNow();
@ -1240,7 +1240,7 @@ private:
bool isInput = (ioid != "Output");
// alsa is stupid here, it advertises dmix and dsnoop as input/output devices, but
// opening dmix as input, or dsnoop as output will trigger errors..
// opening dmix as input, or dsnoop as output will trigger errors
isInput = isInput && ! id.startsWith ("dmix");
isOutput = isOutput && ! id.startsWith ("dsnoop");

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@ -119,7 +119,7 @@ struct ASIOSampleFormat
case ASIOSTDSDInt8NER8: break; // (unhandled)
default:
jassertfalse; // (not a valid format code..)
jassertfalse; // (not a valid format code)
break;
}
}
@ -357,7 +357,7 @@ public:
void updateSampleRates()
{
// find a list of sample rates..
// find a list of sample rates
Array<double> newRates;
if (asioObject != nullptr)
@ -586,7 +586,7 @@ public:
if (! calledback)
{
error = "Device didn't start correctly";
JUCE_ASIO_LOG ("no callbacks - stopping..");
JUCE_ASIO_LOG ("no callbacks - stopping");
asioObject->stop();
}
}
@ -862,7 +862,7 @@ private:
if (shouldUsePreferredSize)
{
JUCE_ASIO_LOG ("Using preferred size for buffer..");
JUCE_ASIO_LOG ("using preferred size for buffer");
auto err = refreshBufferSizes();
if (err == ASE_OK)
@ -919,7 +919,7 @@ private:
void addBufferSizes (long minSize, long maxSize, long preferredSize, long granularity)
{
// find a list of buffer sizes..
// find a list of buffer sizes
JUCE_ASIO_LOG (String ((int) minSize) + "->" + String ((int) maxSize) + ", "
+ String ((int) preferredSize) + ", " + String ((int) granularity));
@ -964,7 +964,7 @@ private:
if (err == ASE_NoClock && numClockSources > 0)
{
JUCE_ASIO_LOG ("trying to set a clock source..");
JUCE_ASIO_LOG ("trying to set a clock source");
err = asioObject->setClockSource (clocks[0].index);
JUCE_ASIO_LOG_ERROR ("setClockSource2", err);
Thread::sleep (10);
@ -976,7 +976,7 @@ private:
if (err == 0)
currentSampleRate = newRate;
// on fail, ignore the attempt to change rate, and run with the current one..
// on fail, ignore the attempt to change rate, and run with the current one
}
}
@ -1170,7 +1170,7 @@ private:
if (driverError.isEmpty())
{
char buffer[512] = {};
asioObject->getDriverName (buffer); // just in case any flimsy drivers expect this to be called..
asioObject->getDriverName (buffer); // just in case any flimsy drivers expect this to be called
}
return driverError;
@ -1178,7 +1178,7 @@ private:
String openDevice()
{
// open the device and get its info..
// open the device and get its info
JUCE_ASIO_LOG ("opening device: " + getName());
needToReset = false;
@ -1241,11 +1241,11 @@ private:
JUCE_ASIO_LOG ("outputReady true");
updateSampleRates();
readLatencies(); // ..doing these steps because cubase does so at this stage
createDummyBuffers (preferredBufferSize); // in initialisation, and some devices fail if we don't.
readLatencies(); // doing these steps because cubase does so at this stage
createDummyBuffers (preferredBufferSize); // in initialisation, and some devices fail if we don't
readLatencies();
// start and stop because cubase does it..
// start and stop because cubase does it
err = asioObject->start();
// ignore an error here, as it might start later after setting other stuff up
JUCE_ASIO_LOG_ERROR ("start", err);
@ -1485,7 +1485,7 @@ public:
for (int i = deviceNames.size(); --i >= 0;)
if (deviceNames[i].containsIgnoreCase ("asio4all"))
return i; // asio4all is a safe choice for a default..
return i; // asio4all is a safe choice for a default
#if JUCE_DEBUG
if (deviceNames.size() > 1 && deviceNames[0].containsIgnoreCase ("digidesign"))
@ -1503,7 +1503,7 @@ public:
return i;
jassertfalse; // unfortunately you can only have a finite number
// of ASIO devices open at the same time..
// of ASIO devices open at the same time
return -1;
}
@ -1579,7 +1579,7 @@ private:
if (RegQueryValueEx (pathKey, 0, 0, &dtype, (LPBYTE) pathName, &dsize) == ERROR_SUCCESS)
// In older code, this used to check for the existence of the file, but there are situations
// where our process doesn't have access to it, but where the driver still loads ok..
// where our process doesn't have access to it, but where the driver still loads ok.
ok = (pathName[0] != 0);
RegCloseKey (pathKey);

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@ -671,7 +671,7 @@ public:
// Annoyingly, after changing the rate and buffer size, some devices fail to
// correctly report their new settings until some random time in the future, so
// after calling updateDetailsFromDevice, we need to manually bodge these values
// to make sure we're using the correct numbers..
// to make sure we're using the correct numbers.
updateDetailsFromDevice (ins, outs);
sampleRate = newSampleRate;
bufferSize = bufferSizeSamples;
@ -2298,7 +2298,7 @@ public:
jassert (hasScanned); // need to call scanForDevices() before doing this
// if they're asking for any input channels at all, use the default input, so we
// get the built-in mic rather than the built-in output with no inputs..
// get the built-in mic rather than the built-in output with no inputs
AudioObjectPropertyAddress pa;
auto selector = forInput ? kAudioHardwarePropertyDefaultInputDevice

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@ -34,7 +34,7 @@
extern "C"
{
// Declare just the minimum number of interfaces for the DSound objects that we need..
// Declare just the minimum number of interfaces for the DSound objects that we need.
struct DSBUFFERDESC
{
DWORD dwSize;
@ -857,7 +857,7 @@ public:
{
if (! isThreadRunning())
{
// something gone wrong and the thread's stopped..
// something gone wrong and the thread's stopped
isOpen_ = false;
return;
}
@ -1119,7 +1119,7 @@ String DSoundAudioIODevice::openDevice (const BigInteger& inputChannels,
sampleRate = sampleRate_ > 0.0 ? sampleRate_ : 44100.0;
if (bufferSizeSamples_ <= 0)
bufferSizeSamples_ = 960; // use as a default size if none is set.
bufferSizeSamples_ = 960; // use as a default size if none is set
bufferSizeSamples = bufferSizeSamples_ & ~7;

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@ -800,11 +800,11 @@ public:
OpenSLAudioIODevice (const String& deviceName) : AudioIODevice (deviceName, openSLTypeName)
{
// OpenSL has piss-poor support for determining latency, so the only way I can find to
// get a number for this is by asking the AudioTrack/AudioRecord classes..
// get a number for this is by asking the AudioTrack/AudioRecord classes.
AndroidAudioIODevice javaDevice (deviceName);
// this is a total guess about how to calculate the latency, but seems to vaguely agree
// with the devices I've tested.. YMMV
// with the devices I've tested. YMMV
inputLatency = (javaDevice.minBufferSizeIn * 2) / 3;
outputLatency = (javaDevice.minBufferSizeOut * 2) / 3;

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@ -1474,7 +1474,7 @@ public:
if (! isThreadRunning())
{
// something's gone wrong and the thread's stopped..
// something's gone wrong and the thread's stopped
flags &= ~flagOpen;
return;
}

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@ -85,7 +85,7 @@ void AudioSourcePlayer::audioDeviceIOCallbackWithContext (const float* const* in
int numActiveChans = 0, numInputs = 0, numOutputs = 0;
// messy stuff needed to compact the channels down into an array
// of non-zero pointers..
// of non-zero pointers
for (int i = 0; i < totalNumInputChannels; ++i)
{
if (inputChannelData[i] != nullptr)

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@ -269,7 +269,7 @@ void AudioTransportSource::getNextAudioBlock (const AudioSourceChannelInfo& info
if (! playing)
{
// just stopped playing, so fade out the last block..
// just stopped playing, so fade out the last block
for (int i = info.buffer->getNumChannels(); --i >= 0;)
info.buffer->applyGainRamp (i, info.startSample, jmin (256, info.numSamples), 1.0f, 0.0f);