我需要在播放最后一个缓冲区后的150ms-200ms内通知一个函数…
通过回调方法我知道有多少缓冲区被排队
我知道缓冲区大小,我知道上一个缓冲区填充的字节数.
首先,我初始化一些缓冲区,然后用音频数据填充缓冲区,然后将它们排队.当音频队列需要填充缓冲区时,它会调用回调并用数据填充缓冲区.
当没有更多可用的音频数据时,Audio Queue会向我发送最后一个空缓冲区,所以我用我拥有的任何数据填充它：
if (sharedCache.numberOfToTalPackets>0)
{
if (currentlyReadingBufferIndex==[sharedCache.baseAudioCache count]-1) {
inBuffer->mAudioDataByteSize = (UInt32)bytesFilled;
lastEnqueudBufferSize=bytesFilled;
err=AudioQueueEnqueueBuffer(inAQ,inBuffer,(UInt32)packetsFilled,packetDescs);
if (err) {
[self failWithErrorCode:err customError:AP_AUdio_QUEUE_ENQUEUE_Failed];
}
printf("if that was the last free packet description,then enqueue the buffer\n");
//go to the next item on keepbuffer array
isBufferFilled=YES;
[self incrementBufferUsedCount];
return;
}
}
当Audio Queue通过回调请求更多数据而我没有更多数据时,我开始倒计时缓冲区.如果缓冲区计数等于零,这意味着要播放的航班上只剩下一个缓冲区,则完成片刻播放时我会尝试停止音频队列.
-(void)decrementBufferUsedCount
{
if (buffersUsed>0) {
buffersUsed--;
printf("buffer on the queue %i\n",buffersUsed);
if (buffersUsed==0) {
NSLog(@"playback is finished\n");
// end playback
isPlayBackDone=YES;
double sampleRate = dataFormat.mSampleRate;
double bufferDuration = lastEnqueudBufferSize/ sampleRate;
double estimatedTimeNeded=bufferDuration*1;
[self performSelector:@selector(stopPlayer) withObject:nil afterDelay:estimatedTimeNeded];
}
}
}
-(void)stopPlayer
{
@synchronized(self)
{
state=AP_STOPPING;
}
err=AudioQueueStop(queue,TRUE);
if (err) {
[self failWithErrorCode:err customError:AP_AUdio_QUEUE_STOP_Failed];
}
else
{
@synchronized(self)
{
state=AP_STOPPED;
NSLog(@"Stopped\n");
}
然而,似乎我无法在这里得到精确的时间.上面的代码会提前阻止玩家
如果我也提前跟进音频切换
double bufferDuration = XMAQDefaultBufSize/ sampleRate; double estimatedTimeNeded=bufferDuration*1;
如果增加1到2,因为缓冲区大小很大我得到一些延迟,似乎1.5是现在的最佳值但我不明白为什么lastEnqueudBufferSize / sampleRate不是wotking
音频文件和缓冲区的详细信息：
Audio file has 22050 sample rate #define kNumberPlaybackBuffers 4 #define kAQDefaultBufSize 16384 it is a vbr file format with no bitrate information available
解决方法
我找到了一种更简单的方法来获得相同的结果(/ -10ms).使用AudioQueueNewOutput()设置输出队列后,初始化要在输出回调中使用的AudioQueueTimelineRef. (我的第一个方法中包含ticksToSeconds函数)不要忘记导入< mach / mach_time.h>
//After AudioQueueNewOutput() AudioQueueTimelineRef timeLine; //ivar AudioQueueCreateTimeline(queue,self.timeLine);
然后在输出回调中调用AudioQueueGetCurrentTime().警告：队列必须播放有效的时间戳.因此,对于非常短的文件,您可能需要使用下面的AudioQueueProcessingTap方法.
AudioTimeStamp timestamp; AudioQueueGetCurrentTime(queue,self->timeLine,×tamp,NULL);
时间戳将当前播放的样本与当前机器时间联系在一起.有了这些信息,我们可以在将来播放最后一个样本时获得准确的机器时间.
Float64 samplesLeft = self->frameCount - timestamp.mSampleTime;//samples in file - current sample Float64 secondsLeft = samplesLeft / self->sampleRate; //seconds of audio to play UInt64 ticksLeft = secondsLeft / ticksToSeconds(); //seconds converted to machine ticks UInt64 machTimeFinish = timestamp.mHostTime + ticksLeft; //machine time of first sample + ticks left
现在我们拥有了这个未来的机器时间,我们可以使用它来准确地计算您想要做的任何事情.
UInt64 currentMachTime = mach_absolute_time();
Uint64 ticksFromNow = machTimeFinish - currentMachTime;
float secondsFromNow = ticksFromNow * ticksToSeconds();
dispatch_after(dispatch_time(disPATCH_TIME_Now,(int64_t)(secondsFromNow * NSEC_PER_SEC)),dispatch_get_main_queue(),^{
//do the thing!!!
printf("Giggety");
});
如果GCD dispatch_async不够准确,有办法设置precision timer
使用AudioQueueProcessingTap
您可以从AudioQueueProcessingTap获得相当低的响应时间.首先,你的回调基本上会置于音频流之间. MyObject类型就是代码中的self(这是ARC桥接在这里以获得函数内部的自我).检查ioFlags会告诉您流何时开始并完成.输出回调的ioTimeStamp描述了回调中的第一个样本将来会触及发言者的时间.所以,如果你想在这里得到准确的话.我添加了一些便利功能,用于将机器时间转换为秒.
#import <mach/mach_time.h>
double getTimeConversion(){
double timecon;
mach_timebase_info_data_t tinfo;
kern_return_t kerror;
kerror = mach_timebase_info(&tinfo);
timecon = (double)tinfo.numer / (double)tinfo.denom;
return timecon;
}
double ticksToSeconds(){
static double ticksToSeconds = 0;
if (!ticksToSeconds) {
ticksToSeconds = getTimeConversion() * 0.000000001;
}
return ticksToSeconds;
}
void processingTapCallback(
void * inClientData,AudioQueueProcessingTapRef inAQTap,UInt32 inNumberFrames,AudioTimeStamp * ioTimeStamp,UInt32 * ioFlags,UInt32 * outNumberFrames,audiobufferlist * ioData){
MyObject *self = (__bridge Object *)inClientData;
AudioQueueProcessingTapGetSourceAudio(inAQTap,inNumberFrames,ioTimeStamp,ioFlags,outNumberFrames,ioData);
if (*ioFlags == kAudioQueueProcessingTap_EndOfStream) {
Float64 sampTime;
UInt32 frameCount;
AudioQueueProcessingTapGetQueueTime(inAQTap,&sampTime,&frameCount);
Float64 samplesInThisCallback = self->frameCount - sampleTime;//file sampleCount - queue current sample
//double secondsInCallback = outNumberFrames / (double)self->sampleRate; outNumberFrames was inaccurate
double secondsInCallback = * samplesInThisCallback / (double)self->sampleRate;
uint64_t timeOfLastSampleLeavingSpeaker = ioTimeStamp->mHostTime + (secondsInCallback / ticksToSeconds());
[self lastSampleDoneAt:timeOfLastSampleLeavingSpeaker];
}
}
-(void)lastSampleDoneAt:(uint64_t)lastSampTime{
uint64_t currentTime = mach_absolute_time();
if (lastSampTime > currentTime) {
double secondsFromNow = (lastSampTime - currentTime) * ticksToSeconds();
dispatch_after(dispatch_time(disPATCH_TIME_Now,^{
//do the thing!!!
});
}
else{
//do the thing!!!
}
}
您可以在AudioQueueNewOutput之后和AudioQueueStart之前将其设置为这样.注意将桥接self传递给inClientData参数.队列实际上将self保持为void *以在回调中使用,我们将其桥接回回调中的objective-C对象.
AudioStreamBasicDescription format; AudioQueueProcessingTapRef tapRef; UInt32 maxFrames = 0; AudioQueueProcessingTapNew(queue,processingTapCallback,(__bridge void *)self,kAudioQueueProcessingTap_PostEffects,&maxFrames,&format,&tapRef);
一旦文件启动,您就可以获得最终机器时间.还有点清洁.
void processingTapCallback(
void * inClientData,ioData);
if (*ioFlags == kAudioQueueProcessingTap_StartOfStream) {
uint64_t timeOfLastSampleLeavingSpeaker = ioTimeStamp->mHostTime + (self->audioDurSeconds / ticksToSeconds());
[self lastSampleDoneAt:timeOfLastSampleLeavingSpeaker];
}
}