【发布时间】：2016-06-08 02:24:23
【问题描述】：
我的第一个应用程序合成了 music audio from a sine look-up table，使用自 iOS 6 以来不推荐使用的方法。我刚刚对其进行了修改，以解决 this blog 帮助下的有关 AudioSession 的警告以及 Apple 对 AVFoundationFramework 的指导方针。音频会话警告现已得到解决，该应用程序会像以前一样产生音频。它目前在 iOS 9 下运行。
但是，该应用偶尔会无缘无故地崩溃。我检查了this SO post，但它似乎处理访问而不是生成原始音频数据，所以它可能没有处理时间问题。我怀疑存在缓冲问题，但在更改或微调代码中的任何内容之前，我需要了解这可能是什么。
我有一个将修改后的应用程序提供给用户的最后期限，因此我非常感谢处理过类似问题的人的来信。
这就是问题所在。该应用程序在模拟器报告中进入调试：
com.apple.coreaudio.AQClient (8):EXC_BAD_ACCESS (code=1, address=0xffffffff10626000)
在调试导航器的线程 8 (com.apple.coreaudio.AQClient (8)) 中，它报告：
0 -[Synth fillBuffer:frames:]
1 -[PlayView audioBufferPlayer:fillBuffer:format:]
2 playCallback
fillBuffer中这行代码高亮显示
float sineValue = (1.0f - b)*sine[a] + b*sine[c];
...audioBufferPlayer 中的这行代码也是如此
int packetsWritten = [synth fillBuffer:buffer->mAudioData frames:packetsPerBuffer];
... 和 playCallBack
[player.delegate audioBufferPlayer:player fillBuffer:inBuffer format:player.audioFormat];
这里是audioBufferPlayer的代码（委托，与above的demo基本相同）。
- (void)audioBufferPlayer:(AudioBufferPlayer*)audioBufferPlayer fillBuffer:(AudioQueueBufferRef)buffer format:(AudioStreamBasicDescription)audioFormat
{
[synthLock lock];
int packetsPerBuffer = buffer->mAudioDataBytesCapacity / audioFormat.mBytesPerPacket;
int packetsWritten = [synth fillBuffer:buffer->mAudioData frames:packetsPerBuffer];
buffer->mAudioDataByteSize = packetsWritten * audioFormat.mBytesPerPacket;
[synthLock unlock];
}
...（在 myViewController 中初始化）
- (id)init
{
if ((self = [super init])) {
// The audio buffer is managed (filled up etc.) within its own thread (Audio Queue thread)
// Since we are also responding to changes from the GUI, we need a lock so both threads
// do not attempt to change the same value independently.
synthLock = [[NSLock alloc] init];
// Synth and the AudioBufferPlayer must use the same sample rate.
float sampleRate = 44100.0f;
// Initialise synth to fill the audio buffer with audio samples.
synth = [[Synth alloc] initWithSampleRate:sampleRate];
// Initialise note buttons
buttons = [[NSMutableArray alloc] init];
// Initialise the audio buffer.
player = [[AudioBufferPlayer alloc] initWithSampleRate:sampleRate channels:1 bitsPerChannel:16 packetsPerBuffer:1024];
player.delegate = self;
player.gain = 0.9f;
[[AVAudioSession sharedInstance] setActive:YES error:nil];
}
return self;
} // initialisation
... 以及用于 playCallback
static void playCallback( void* inUserData, AudioQueueRef inAudioQueue, AudioQueueBufferRef inBuffer)
{
AudioBufferPlayer* player = (AudioBufferPlayer*) inUserData;
if (player.playing){
[player.delegate audioBufferPlayer:player fillBuffer:inBuffer format:player.audioFormat];
AudioQueueEnqueueBuffer(inAudioQueue, inBuffer, 0, NULL);
}
}
...这里是用于合成音频的 fillBuffer 代码
- (int)fillBuffer:(void*)buffer frames:(int)frames
{
SInt16* p = (SInt16*)buffer;
// Loop through the frames (or "block size"), then consider each sample for each tone.
for (int f = 0; f < frames; ++f)
{
float m = 0.0f; // the mixed value for this frame
for (int n = 0; n < MAX_TONE_EVENTS; ++n)
{
if (tones[n].state == STATE_INACTIVE) // only active tones
continue;
// recalculate a 30sec envelope and place in a look-up table
// Longer notes need to interpolate through the envelope
int a = (int)tones[n].envStep; // integer part (like a floored float)
float b = tones[n].envStep - a; // decimal part (like doing a modulo)
// c allows us to calculate if we need to wrap around
int c = a + 1; // (like a ceiling of integer part)
if (c >= envLength) c = a; // don't wrap around
/////////////// LOOK UP ENVELOPE TABLE /////////////////
// uses table look-up with interpolation for both level and pitch envelopes
// 'b' is a value interpolated between 2 successive samples 'a' and 'c')
// first, read values for the level envelope
float envValue = (1.0f - b)*tones[n].levelEnvelope[a] + b*tones[n].levelEnvelope[c];
// then the pitch envelope
float pitchFactorValue = (1.0f - b)*tones[n].pitchEnvelope[a] + b*tones[n].pitchEnvelope[c];
// Advance envelope pointer one step
tones[n].envStep += tones[n].envDelta;
// Turn note off at the end of the envelope.
if (((int)tones[n].envStep) >= envLength){
tones[n].state = STATE_INACTIVE;
continue;
}
// Precalculated Sine look-up table
a = (int)tones[n].phase; // integer part
b = tones[n].phase - a; // decimal part
c = a + 1;
if (c >= sineLength) c -= sineLength; // wrap around
///////////////// LOOK UP OF SINE TABLE ///////////////////
float sineValue = (1.0f - b)*sine[a] + b*sine[c];
// Wrap round when we get to the end of the sine look-up table.
tones[n].phase += (tones[n].frequency * pitchFactorValue); // calculate frequency for each point in the pitch envelope
if (((int)tones[n].phase) >= sineLength)
tones[n].phase -= sineLength;
////////////////// RAMP NOTE OFF IF IT HAS BEEN UNPRESSED
if (tones[n].state == STATE_UNPRESSED) {
tones[n].gain -= 0.0001;
if ( tones[n].gain <= 0 ) {
tones[n].state = STATE_INACTIVE;
}
}
//////////////// FINAL SAMPLE VALUE ///////////////////
float s = sineValue * envValue * gain * tones[n].gain;
// Clip the signal, if needed.
if (s > 1.0f) s = 1.0f;
else if (s < -1.0f) s = -1.0f;
// Add the sample to the out-going signal
m += s;
}
// Write the sample mix to the buffer as a 16-bit word.
p[f] = (SInt16)(m * 0x7FFF);
}
return frames;
}
我不确定这是否是一个红鲱鱼，但我在几个调试寄存器中遇到了 NaN。它似乎在计算fillBuffer 中的正弦查找的相位增量时发生（见上文）。该计算以 44.1 kHz 的采样率对每个样本进行多达十几个部分的计算，并在 iPhone 4 上的 iOS 4 中运行。我在 iOS 9 的模拟器上运行。我所做的唯一更改在这篇文章中进行了描述！
【问题讨论】：
-
只是一个用于帮助我调试我的一些 CoreAudio 代码的 NaN“捕手-消音器”：
if(x != x) x = 0; -
@user3078414 - 抱歉，我不明白 NaN 捕手-消音器如何使用像 'float sineValue = (1.0f - b)*sine[a] + b*sine[ 这样的表达式C];'你的意思是'如果（b！= b）b = 0;' ?或“如果（正弦值！= 正弦值）正弦值 = 0；” ?这不会在波形中插入偶尔的不连续性吗？
-
这样，正如所写的那样，它当然会插入一个不连续性（而 NaN 和其他几个原因也可能使线程崩溃），但这将证明您的代码确实有一个 NaN问题，所以你会限制你的搜索区域。你基本上是在调试一个实时无限循环，所以从实时的角度观察它是有原因的。
-
@user3078414 - 是的，这是一个很好的策略。最好不要假设 NaN 是唯一的问题。也许也可以修改 'if(x != x) x = 0;'并覆盖任何不连续性，例如'如果（x！= x）x = y；'其中 y 是根据 'x[n-2]' 和 'x[n-1]' 之差计算得出的 'x[n]' 的值
-
当然我会先试试你原来的建议
标签： ios objective-c iphone core-audio