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https://github.com/juce-framework/JUCE.git
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Formatting: Remove double-dots from comments and other strings
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82dc6d1c7e
commit
83e5264c86
209 changed files with 508 additions and 509 deletions
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@ -1350,7 +1350,7 @@ double AudioDeviceManager::LevelMeter::getCurrentLevel() const noexcept
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void AudioDeviceManager::playTestSound()
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{
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{ // cunningly nested to swap, unlock and delete in that order.
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{ // cunningly nested to swap, unlock and delete in that order
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std::unique_ptr<AudioBuffer<float>> oldSound;
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{
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@ -105,7 +105,7 @@ void MidiMessageCollector::removeNextBlockOfMessages (MidiBuffer& destBuffer,
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if (numSourceSamples > numSamples)
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{
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// if our list of events is longer than the buffer we're being
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// asked for, scale them down to squeeze them all in..
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// asked for, scale them down to squeeze them all in
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const int maxBlockLengthToUse = numSamples << 5;
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auto iter = incomingMessages.cbegin();
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@ -220,7 +220,7 @@ public:
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return false;
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}
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if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw..
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if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0) // works better for plughw
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isInterleaved = true;
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else if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_NONINTERLEAVED) >= 0)
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isInterleaved = false;
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@ -286,7 +286,7 @@ public:
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|| JUCE_ALSA_FAILED (snd_pcm_hw_params_get_periods (hwParams, &periods, &dir)))
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latency = 0;
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else
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latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency..)
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latency = (int) frames * ((int) periods - 1); // (this is the method JACK uses to guess the latency)
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JUCE_ALSA_LOG ("frames: " << (int) frames << ", periods: " << (int) periods
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<< ", samplesPerPeriod: " << (int) samplesPerPeriod);
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@ -551,7 +551,7 @@ public:
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currentOutputChans.clear();
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// Note that the input device is opened before an output, because we've heard
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// of drivers where doing it in the reverse order mysteriously fails.. If this
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// of drivers where doing it in the reverse order mysteriously fails. If this
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// order also causes problems, let us know and we'll see if we can find a compromise!
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if (inputChannelDataForCallback.size() > 0 && inputId.isNotEmpty())
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@ -660,7 +660,7 @@ public:
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if ((! waitForThreadToExit (400)) && audioIoInProgress && numCallbacks == callbacksToStop)
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{
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JUCE_ALSA_LOG ("Thread is stuck in i/o.. Is pulseaudio suspended?");
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JUCE_ALSA_LOG ("Thread is stuck in i/o. Is pulseaudio suspended?");
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if (outputDevice != nullptr) outputDevice->closeNow();
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if (inputDevice != nullptr) inputDevice->closeNow();
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@ -1240,7 +1240,7 @@ private:
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bool isInput = (ioid != "Output");
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// alsa is stupid here, it advertises dmix and dsnoop as input/output devices, but
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// opening dmix as input, or dsnoop as output will trigger errors..
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// opening dmix as input, or dsnoop as output will trigger errors
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isInput = isInput && ! id.startsWith ("dmix");
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isOutput = isOutput && ! id.startsWith ("dsnoop");
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@ -119,7 +119,7 @@ struct ASIOSampleFormat
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case ASIOSTDSDInt8NER8: break; // (unhandled)
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default:
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jassertfalse; // (not a valid format code..)
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jassertfalse; // (not a valid format code)
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break;
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}
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}
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@ -357,7 +357,7 @@ public:
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void updateSampleRates()
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{
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// find a list of sample rates..
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// find a list of sample rates
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Array<double> newRates;
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if (asioObject != nullptr)
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@ -586,7 +586,7 @@ public:
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if (! calledback)
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{
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error = "Device didn't start correctly";
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JUCE_ASIO_LOG ("no callbacks - stopping..");
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JUCE_ASIO_LOG ("no callbacks - stopping");
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asioObject->stop();
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}
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}
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@ -862,7 +862,7 @@ private:
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if (shouldUsePreferredSize)
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{
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JUCE_ASIO_LOG ("Using preferred size for buffer..");
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JUCE_ASIO_LOG ("using preferred size for buffer");
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auto err = refreshBufferSizes();
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if (err == ASE_OK)
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@ -919,7 +919,7 @@ private:
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void addBufferSizes (long minSize, long maxSize, long preferredSize, long granularity)
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{
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// find a list of buffer sizes..
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// find a list of buffer sizes
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JUCE_ASIO_LOG (String ((int) minSize) + "->" + String ((int) maxSize) + ", "
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+ String ((int) preferredSize) + ", " + String ((int) granularity));
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@ -964,7 +964,7 @@ private:
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if (err == ASE_NoClock && numClockSources > 0)
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{
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JUCE_ASIO_LOG ("trying to set a clock source..");
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JUCE_ASIO_LOG ("trying to set a clock source");
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err = asioObject->setClockSource (clocks[0].index);
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JUCE_ASIO_LOG_ERROR ("setClockSource2", err);
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Thread::sleep (10);
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@ -976,7 +976,7 @@ private:
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if (err == 0)
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currentSampleRate = newRate;
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// on fail, ignore the attempt to change rate, and run with the current one..
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// on fail, ignore the attempt to change rate, and run with the current one
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}
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}
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@ -1170,7 +1170,7 @@ private:
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if (driverError.isEmpty())
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{
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char buffer[512] = {};
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asioObject->getDriverName (buffer); // just in case any flimsy drivers expect this to be called..
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asioObject->getDriverName (buffer); // just in case any flimsy drivers expect this to be called
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}
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return driverError;
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@ -1178,7 +1178,7 @@ private:
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String openDevice()
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{
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// open the device and get its info..
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// open the device and get its info
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JUCE_ASIO_LOG ("opening device: " + getName());
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needToReset = false;
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@ -1241,11 +1241,11 @@ private:
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JUCE_ASIO_LOG ("outputReady true");
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updateSampleRates();
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readLatencies(); // ..doing these steps because cubase does so at this stage
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createDummyBuffers (preferredBufferSize); // in initialisation, and some devices fail if we don't.
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readLatencies(); // doing these steps because cubase does so at this stage
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createDummyBuffers (preferredBufferSize); // in initialisation, and some devices fail if we don't
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readLatencies();
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// start and stop because cubase does it..
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// start and stop because cubase does it
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err = asioObject->start();
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// ignore an error here, as it might start later after setting other stuff up
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JUCE_ASIO_LOG_ERROR ("start", err);
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@ -1485,7 +1485,7 @@ public:
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for (int i = deviceNames.size(); --i >= 0;)
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if (deviceNames[i].containsIgnoreCase ("asio4all"))
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return i; // asio4all is a safe choice for a default..
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return i; // asio4all is a safe choice for a default
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#if JUCE_DEBUG
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if (deviceNames.size() > 1 && deviceNames[0].containsIgnoreCase ("digidesign"))
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@ -1503,7 +1503,7 @@ public:
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return i;
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jassertfalse; // unfortunately you can only have a finite number
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// of ASIO devices open at the same time..
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// of ASIO devices open at the same time
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return -1;
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}
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@ -1579,7 +1579,7 @@ private:
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if (RegQueryValueEx (pathKey, 0, 0, &dtype, (LPBYTE) pathName, &dsize) == ERROR_SUCCESS)
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// In older code, this used to check for the existence of the file, but there are situations
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// where our process doesn't have access to it, but where the driver still loads ok..
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// where our process doesn't have access to it, but where the driver still loads ok.
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ok = (pathName[0] != 0);
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RegCloseKey (pathKey);
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@ -671,7 +671,7 @@ public:
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// Annoyingly, after changing the rate and buffer size, some devices fail to
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// correctly report their new settings until some random time in the future, so
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// after calling updateDetailsFromDevice, we need to manually bodge these values
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// to make sure we're using the correct numbers..
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// to make sure we're using the correct numbers.
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updateDetailsFromDevice (ins, outs);
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sampleRate = newSampleRate;
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bufferSize = bufferSizeSamples;
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@ -2298,7 +2298,7 @@ public:
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jassert (hasScanned); // need to call scanForDevices() before doing this
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// if they're asking for any input channels at all, use the default input, so we
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// get the built-in mic rather than the built-in output with no inputs..
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// get the built-in mic rather than the built-in output with no inputs
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AudioObjectPropertyAddress pa;
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auto selector = forInput ? kAudioHardwarePropertyDefaultInputDevice
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@ -34,7 +34,7 @@
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extern "C"
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{
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// Declare just the minimum number of interfaces for the DSound objects that we need..
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// Declare just the minimum number of interfaces for the DSound objects that we need.
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struct DSBUFFERDESC
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{
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DWORD dwSize;
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@ -857,7 +857,7 @@ public:
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{
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if (! isThreadRunning())
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{
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// something gone wrong and the thread's stopped..
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// something gone wrong and the thread's stopped
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isOpen_ = false;
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return;
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}
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@ -1119,7 +1119,7 @@ String DSoundAudioIODevice::openDevice (const BigInteger& inputChannels,
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sampleRate = sampleRate_ > 0.0 ? sampleRate_ : 44100.0;
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if (bufferSizeSamples_ <= 0)
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bufferSizeSamples_ = 960; // use as a default size if none is set.
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bufferSizeSamples_ = 960; // use as a default size if none is set
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bufferSizeSamples = bufferSizeSamples_ & ~7;
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@ -800,11 +800,11 @@ public:
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OpenSLAudioIODevice (const String& deviceName) : AudioIODevice (deviceName, openSLTypeName)
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{
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// OpenSL has piss-poor support for determining latency, so the only way I can find to
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// get a number for this is by asking the AudioTrack/AudioRecord classes..
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// get a number for this is by asking the AudioTrack/AudioRecord classes.
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AndroidAudioIODevice javaDevice (deviceName);
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// this is a total guess about how to calculate the latency, but seems to vaguely agree
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// with the devices I've tested.. YMMV
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// with the devices I've tested. YMMV
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inputLatency = (javaDevice.minBufferSizeIn * 2) / 3;
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outputLatency = (javaDevice.minBufferSizeOut * 2) / 3;
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@ -1474,7 +1474,7 @@ public:
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if (! isThreadRunning())
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{
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// something's gone wrong and the thread's stopped..
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// something's gone wrong and the thread's stopped
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flags &= ~flagOpen;
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return;
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}
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@ -85,7 +85,7 @@ void AudioSourcePlayer::audioDeviceIOCallbackWithContext (const float* const* in
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int numActiveChans = 0, numInputs = 0, numOutputs = 0;
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// messy stuff needed to compact the channels down into an array
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// of non-zero pointers..
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// of non-zero pointers
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for (int i = 0; i < totalNumInputChannels; ++i)
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{
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if (inputChannelData[i] != nullptr)
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@ -269,7 +269,7 @@ void AudioTransportSource::getNextAudioBlock (const AudioSourceChannelInfo& info
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if (! playing)
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{
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// just stopped playing, so fade out the last block..
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// just stopped playing, so fade out the last block
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for (int i = info.buffer->getNumChannels(); --i >= 0;)
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info.buffer->applyGainRamp (i, info.startSample, jmin (256, info.numSamples), 1.0f, 0.0f);
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