我正在关注这个例子,但它并没有那么有用: https://github.com/xamarin/urho-samples/tree/master/FeatureSamples/Core/29_SoundSynthesis
无论如何我得到一个运行时错误,说:该应用程序尚未配置。 但我做了一个应用程序对象。 错误发生在node = new Node(); 我错过了什么 这是我的班级:
using System;
using Urho.Audio;
using Urho;
using Urho.Resources;
using Urho.Gui;
using System.Diagnostics;
using System.Globalization;
namespace Brain_Entrainment
{
public class IsochronicTones : Urho.Application
{
/// Scene node for the sound component.
Node node;
/// Sound stream that we update.
BufferedSoundStream soundStream;
public double Frequency { get; set; }
public double Beat { get; set; }
public double Amplitude { get; set; }
public float Bufferlength { get; set; }
const int numBuffers = 3;
public IsochronicTones(ApplicationOptions AppOption) : base(AppOption)
{
Amplitude = 1;
Frequency = 100;
Beat = 0;
Bufferlength = Int32.MaxValue;
}
public void play()
{
Start();
}
protected override void OnUpdate(float timeStep)
{
UpdateSound();
base.OnUpdate(timeStep);
}
protected override void Start()
{
base.Start();
CreateSound();
}
void CreateSound()
{
// Sound source needs a node so that it is considered enabled
node = new Node();
SoundSource source = node.CreateComponent();
soundStream = new BufferedSoundStream();
// Set format: 44100 Hz, sixteen bit, mono
soundStream.SetFormat(44100, true, false);
// Start playback. We don't have data in the stream yet, but the
//SoundSource will wait until there is data
// as the stream is by default in the "don't stop at end" mode
source.Play(soundStream);
}
void UpdateSound()
{
// Try to keep 1/10 seconds of sound in the buffer, to avoid both
//dropouts and unnecessary latency
float targetLength = 1.0f / 10.0f;
float requiredLength = targetLength -
Bufferlength;//soundStream.BufferLength;
float w = 0;
if (requiredLength < 0.0f)
return;
uint numSamples = (uint)(soundStream.Frequency * requiredLength);
if (numSamples == 0)
return;
// Allocate a new buffer and fill it with a simple two-oscillator
//algorithm.The sound is over - amplified
// (distorted), clamped to the 16-bit range, and finally lowpass -
//filtered according to the coefficient
var newData = new short[numSamples];
for (int i = 0; i < numSamples; ++i)
{
float newValue = 0;
if (Beat == 0)
{
newValue = (float)(Amplitude * Math.Sin(Math.PI * Frequency * i / 44100D));
}
else
{
w = (float)(1D * Math.Sin(i * Math.PI * Beat / 44100D));
if (w < 0)
{
w = 0;
}
newValue = (float)(Amplitude * Math.Sin(Math.PI * Frequency * i / 44100D));
}
//accumulator = MathHelper.Lerp(accumulator, newValue, filter);
newData[i] = (short)newValue;
}
// Queue buffer to the stream for playback
soundStream.AddData(newData, 0, newData.Length);
}
}
}