From 58ba3038f0f88e7724f22d43806ac2fa548ec4f7 Mon Sep 17 00:00:00 2001
From: jsiegle <jsiegle@mit.edu>
Date: Sun, 4 Mar 2012 12:55:00 -0500
Subject: [PATCH] Any sample rate seems to work, as long as tempBuffer is large
enough
---
Source/Audio/AudioComponent.cpp | 2 +-
Source/Processors/ResamplingNode.cpp | 7 +++++--
2 files changed, 6 insertions(+), 3 deletions(-)
diff --git a/Source/Audio/AudioComponent.cpp b/Source/Audio/AudioComponent.cpp
index bd28c5c6f..93119a912 100644
--- a/Source/Audio/AudioComponent.cpp
+++ b/Source/Audio/AudioComponent.cpp
@@ -47,7 +47,7 @@ AudioComponent::AudioComponent() : isPlaying(false)
AudioDeviceManager::AudioDeviceSetup setup;
deviceManager.getAudioDeviceSetup(setup);
- setup.bufferSize = 128; /// larger buffer = fewer empty blocks, but longer latencies
+ setup.bufferSize = 1024; /// larger buffer = fewer empty blocks, but longer latencies
setup.useDefaultInputChannels = false;
setup.inputChannels = 0;
setup.useDefaultOutputChannels = true;
diff --git a/Source/Processors/ResamplingNode.cpp b/Source/Processors/ResamplingNode.cpp
index a77f70e5f..1c17cdd5c 100644
--- a/Source/Processors/ResamplingNode.cpp
+++ b/Source/Processors/ResamplingNode.cpp
@@ -134,7 +134,7 @@ void ResamplingNode::prepareToPlay (double sampleRate_, int estimatedSamplesPerB
if (destBufferIsTempBuffer) {
destBufferSampleRate = sampleRate_;
- tempBuffer->setSize(getNumInputs(), 2048);
+ tempBuffer->setSize(getNumInputs(), 4096);
}
else {
destBufferSampleRate = float(destBufferWidth) / destBufferTimebaseSecs;
@@ -189,7 +189,7 @@ void ResamplingNode::process(AudioSampleBuffer &buffer,
if (destBufferIsTempBuffer) {
ratio = float(nSamps) / float(buffer.getNumSamples());
- valuesNeeded = tempBuffer->getNumSamples();
+ valuesNeeded = buffer.getNumSamples();
} else {
ratio = sourceBufferSampleRate / destBufferSampleRate;
valuesNeeded = (int) buffer.getNumSamples() / ratio;
@@ -278,7 +278,10 @@ void ResamplingNode::process(AudioSampleBuffer &buffer,
if (destBufferIsTempBuffer) {
// copy the temp buffer into the original buffer
+
+
buffer = *tempBuffer;
+ //buffer = AudioSampleBuffer(tempBuffer->getArrayOfChannels(), 2, buffer.getNumSamples());
//buffer.setSize(2,0,true,false,true);
--
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