问题
I am able to stream and play m4a files using Audio File Services + Audio Queue Services. Bitrate information of the file is not available to the Audio Queue because of file type.
After downloading the all of the audio packets I feed them to the player.
When I choose a buffer size around 32768 or 16384 since callbacks are called less often and and each buffer size is big, it seems it is playing almost at regular speed. Problem is sometimes I have to play small files as well but when I choose a small buffer size -512 or 1024 or 2048 up to 8192- audio plays really fast and with occasional glitches.
I know calling objective-c function in c callback is not a great idea but for readability and easiness I do that. Regardless I think that is not the problem.
// allocate the buffers and prime the queue with some data before starting
AudioQueueBufferRef buffers[XMNumberPlaybackBuffers];
int i;
for (i = 0; i < XMNumberPlaybackBuffers; ++i)
{
err=AudioQueueAllocateBuffer(queue, XMAQDefaultBufSize, &buffers[i]);
if (err) {
[self failWithErrorCode:err customError:AP_AUDIO_QUEUE_BUFFER_ALLOCATION_FAILED];
}
@synchronized(self)
{
state=AP_WAITING_FOR_QUEUE_TO_START;
}
// manually invoke callback to fill buffers with data
MyAQOutputCallBack((__bridge void *)(self), queue, buffers[i]);
}
I also get audio packets from a mutablearray of dictionaries...
#define XMNumberPlaybackBuffers 4
#define XMAQDefaultBufSize 8192
#pragma mark playback callback function
static void MyAQOutputCallBack(void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inCompleteAQBuffer)
{
// this is called by the audio queue when it has finished decoding our data.
// The buffer is now free to be reused.
NSLog(@"MyAQOutputCallBack..");
//printf("MyAQOutputCallBack...\n");
XMAudioPlayer* player = (__bridge XMAudioPlayer *)inUserData;
[player handleBufferCompleteForQueue:inAQ buffer:inCompleteAQBuffer];
//printf("##################\n");
}
- (void)handleBufferCompleteForQueue:(AudioQueueRef)inAQ
buffer:(AudioQueueBufferRef)inBuffer
{
//NSLog(@"######################\n");
AudioTimeStamp queueTime;
Boolean discontinuity;
err = AudioQueueGetCurrentTime(queue, NULL, &queueTime, &discontinuity);
printf("queueTime.mSampleTime %.2f\n",queueTime.mSampleTime/dataFormat.mSampleRate);
AudioStreamPacketDescription packetDescs[XMAQMaxPacketDescs]; // packet descriptions for enqueuing audio
BOOL isBufferFilled=NO;
size_t bytesFilled=0; // how many bytes have been filled
size_t packetsFilled=0; // how many packets have been filled
size_t bufSpaceRemaining;
while (isBufferFilled==NO && isEOF==NO) {
if (currentlyReadingBufferIndex<[sharedCache.audioCache count]) {
//loop thru untill buffer is enqued
if (sharedCache.audioCache) {
NSMutableDictionary *myDict= [[NSMutableDictionary alloc] init];
myDict=[sharedCache.audioCache objectAtIndex:currentlyReadingBufferIndex];
//why I cant use this info?
//UInt32 inNumberBytes =[[myDict objectForKey:@"inNumberBytes"] intValue];
UInt32 inNumberPackets =[[myDict objectForKey:@"inNumberPackets"] intValue];
NSData *convert=[myDict objectForKey:@"inInputData"];
const void *inInputData=(const char *)[convert bytes];
//AudioStreamPacketDescription *inPacketDescriptions;
AudioStreamPacketDescription *inPacketDescriptions= malloc(sizeof(AudioStreamPacketDescription));
NSNumber *mStartOffset = [myDict objectForKey:@"mStartOffset"];
NSNumber *mDataByteSize = [myDict objectForKey:@"mDataByteSize"];
NSNumber *mVariableFramesInPacket = [myDict objectForKey:@"mVariableFramesInPacket"];
inPacketDescriptions->mVariableFramesInPacket=[mVariableFramesInPacket intValue];
inPacketDescriptions->mStartOffset=[mStartOffset intValue];
inPacketDescriptions->mDataByteSize=[mDataByteSize intValue];
for (int i = 0; i < inNumberPackets; ++i)
{
SInt64 packetOffset = [mStartOffset intValue];
SInt64 packetSize = [mDataByteSize intValue];
//printf("packetOffset %lli\n",packetOffset);
//printf("packetSize %lli\n",packetSize);
currentlyReadingBufferIndex++;
if (packetSize > packetBufferSize)
{
//[self failWithErrorCode:AS_AUDIO_BUFFER_TOO_SMALL];
}
bufSpaceRemaining = packetBufferSize - bytesFilled;
//printf("bufSpaceRemaining %zu\n",bufSpaceRemaining);
// if the space remaining in the buffer is not enough for this packet, then enqueue the buffer.
if (bufSpaceRemaining < packetSize)
{
inBuffer->mAudioDataByteSize = (UInt32)bytesFilled;
err=AudioQueueEnqueueBuffer(inAQ,inBuffer,(UInt32)packetsFilled,packetDescs);
if (err) {
[self failWithErrorCode:err customError:AP_AUDIO_QUEUE_ENQUEUE_FAILED];
}
isBufferFilled=YES;
[self incrementBufferUsedCount];
return;
}
@synchronized(self)
{
//
// If there was some kind of issue with enqueueBuffer and we didn't
// make space for the new audio data then back out
//
if (bytesFilled + packetSize > packetBufferSize)
{
return;
}
// copy data to the audio queue buffer
//error -66686 refers to
//kAudioQueueErr_BufferEmpty = -66686
//memcpy((char*)inBuffer->mAudioData + bytesFilled, (const char*)inInputData + packetOffset, packetSize);
memcpy(inBuffer->mAudioData + bytesFilled, (const char*)inInputData + packetOffset, packetSize);
// fill out packet description
packetDescs[packetsFilled] = inPacketDescriptions[0];
packetDescs[packetsFilled].mStartOffset = bytesFilled;
bytesFilled += packetSize;
packetsFilled += 1;
free(inPacketDescriptions);
}
// if that was the last free packet description, then enqueue the buffer.
// size_t packetsDescsRemaining = kAQMaxPacketDescs - packetsFilled;
// if (packetsDescsRemaining == 0) {
//
// }
if (sharedCache.numberOfToTalPackets>0)
{
if (currentlyReadingBufferIndex==[sharedCache.audioCache count]-1) {
if (loop==NO) {
inBuffer->mAudioDataByteSize = (UInt32)bytesFilled;
lastEnqueudBufferSize=bytesFilled;
lastbufferPacketCount=(int)packetsFilled;
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;
}
else
{
//if loop is yes return to first packet pointer and fill the rest of the buffer before enqueing it
//set the currently reading to zero
//check the space in buffer
//if space is avaialbele create a while loop till it is filled
//then enqueu the buffer
currentlyReadingBufferIndex=0;
}
}
}
}
}
}
}
}
EDIT:
For anyone who is visiting this in the future, turns out my exact problem was AudioStreamPacketDescription packetDescs[XMAQMaxPacketDescs]; so XMAQMaxPacketDescs here is 512 when I choose bigger buffer sizes I was enqueueing closer numbers to 512 packets for each buffer so it was playing at normal speed
However for small buffer sizes like 1024 this is only 2-3 packets total so rest of the 508 packets were 0, and player was trying to play all the packetdescriptions 512 of them an that's why it was too fast.
I solved the problem by counting the number of total number of packets that I put the buffers then I created a dynamic AudioStreamPacketDescription description array..
AudioStreamPacketDescription * tempDesc = (AudioStreamPacketDescription *)(malloc(packetsFilledDesc * sizeof(AudioStreamPacketDescription)));
memcpy(tempDesc,packetDescs, packetsFilledDesc*sizeof(AudioStreamPacketDescription));
err = AudioQueueEnqueueBuffer(inAQ,inBuffer,packetsFilledDesc,tempDesc);
if (err) {
[self failWithErrorCode:err customError:AP_AUDIO_QUEUE_ENQUEUE_FAILED];
}
However I accepted and rewarded 100 points to DAVE answer's below, soon I realized my problem was different.....
回答1:
When you allocate your queue for variable bit rate, instead of using XMAQDefaultBufSize, for variable bit rate, you need to calculate the packet size. I pulled a method from this tutorial from this book that shows how it's done.
void DeriveBufferSize (AudioQueueRef audioQueue, AudioStreamBasicDescription ASBDescription, Float64 seconds, UInt32 *outBufferSize)
{
static const int maxBufferSize = 0x50000; // punting with 50k
int maxPacketSize = ASBDescription.mBytesPerPacket;
if (maxPacketSize == 0)
{
UInt32 maxVBRPacketSize = sizeof(maxPacketSize);
AudioQueueGetProperty(audioQueue, kAudioConverterPropertyMaximumOutputPacketSize, &maxPacketSize, &maxVBRPacketSize);
}
Float64 numBytesForTime = ASBDescription.mSampleRate * maxPacketSize * seconds;
*outBufferSize = (UInt32)((numBytesForTime < maxBufferSize) ? numBytesForTime : maxBufferSize);
}
You would use it like this.
Float64 bufferDurSeconds = 0.54321;
AudioStreamBasicDescription myAsbd = self.format; // or something
UInt32 bufferByteSize;
DeriveBufferSize(recordState.queue, myAsbd, bufferDurSeconds, &bufferByteSize);
AudioQueueAllocateBuffer(queue, bufferByteSize, &buffers[i]);
Using kAudioConverterPropertyMaximumOutputPacketSize, you calculate the smallest buffer size you can safely use for the unpredictable variable bit rate file. If your file is too small, you just need to identify which samples are padding for the codec.
来源：https://stackoverflow.com/questions/30533892/audio-queue-is-playing-too-fast-when-the-buffer-size-is-small