CODEWITHSUNDEEP
ioswavaudioqueue
SpaceDust__
SpaceDust__

Reputation: 4914

AudioStreamBasicDescription setting values for wav

I am trying to play a simple PCM file on iOS but couldn't wrap my head around AudioStreamBasicDescription and this link does not provide enough information.

I get this values from terminal

afinfo BlameItOnTheNight.wav
File:           BlameItOnTheNight.wav
File type ID:   WAVE
Num Tracks:     1
----
Data format:     2 ch,  44100 Hz, 'lpcm' (0x0000000C) 16-bit little-endian signed integer
                no channel layout.
estimated duration: 9.938141 sec
audio bytes: 1753088
audio packets: 438272
bit rate: 1411200 bits per second
packet size upper bound: 4
maximum packet size: 4
audio data file offset: 44
optimized
source bit depth: I16
----

Then I choose values in code 

- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
    format->mSampleRate = 44100.0;
    format->mFormatID = kAudioFormatLinearPCM;
    format->mFramesPerPacket = 1;
    format->mChannelsPerFrame = 2;
    format->mBytesPerFrame = format->mChannelsPerFrame * sizeof(Float32);
    format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
    format->mBitsPerChannel = sizeof(Float32) * 8;
    format->mReserved = 0;
    format->mFormatFlags =  kAudioFormatFlagIsSignedInteger |
    kAudioFormatFlagsNativeEndian |
    kAudioFormatFlagIsPacked;
}

Audio plays really fast.

Whats the correct way the calculate this values based on actual audio file?

Upvotes: 3

Views: 3980

Answers (1)

SpaceDust__
SpaceDust__

Reputation: 4914

when I changed the values I was getting following error.

error for object 0x7fba72c50db8: incorrect checksum for freed object - object was probably modified after being freed.
*** set a breakpoint in malloc_error_break to debug

then finally I figured out that my AudioStreamBasicDescription bitsperchannel values was not correct also the buffer size was not enough.

So first I have changed the values to 

- (void)setupAudioFormat:(AudioStreamBasicDescription*)format
{
    format->mSampleRate = 44100.0;
    format->mFormatID = kAudioFormatLinearPCM;
    format->mFramesPerPacket = 1; //For uncompressed audio, the value is 1. For variable bit-rate formats, the value is a larger fixed number, such as 1024 for AAC
    format->mChannelsPerFrame = 2;
    format->mBytesPerFrame = format->mChannelsPerFrame * 2;
    format->mBytesPerPacket = format->mFramesPerPacket * format->mBytesPerFrame;
    format->mBitsPerChannel = 16;
    format->mReserved = 0;
    format->mFormatFlags =  kAudioFormatFlagIsSignedInteger |
    kAudioFormatFlagsNativeEndian |
    kLinearPCMFormatFlagIsPacked;
}

then when I allocate buffer I increased the size 

// Allocate and prime playback buffers
            playState.playing = true;
            for (int i = 0; i < NUM_BUFFERS && playState.playing; i++)
            {
                AudioQueueAllocateBuffer(playState.queue, 32000, &playState.buffers[i]);
                AudioOutputCallback(&playState, playState.queue, playState.buffers[i]);
            }

In my original code it was set to 8000, now changing it to 32000 solves the problem.

Upvotes: 4

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