To answer my own question:
The issue was indeed that I was not handling the interleaving correctly and
thus duplicating samples by pushing both L & R onto the same channel.
I adjusted my data structure and render callback code to handle left and right
channels correctly now:
#define CHANNEL_L 0
#define CHANNEL_R 1
static OSStatus multiChannelMixerRenderCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
__unsafe_unretained SWAudioEngine *self = (__bridge SWAudioEngine
*)inRefCon;
Float32 *out0 = (Float32 *)ioData->mBuffers[CHANNEL_L].mData;
Float32 *out1 = (Float32 *)ioData->mBuffers[CHANNEL_R].mData;
for (UInt32 i = 0; i < inNumberFrames; i++) {
MTSampleData data = MT_PopSample(inBusNumber);
out0[i] = data.sampleL;
out1[i] = data.sampleR;
}
return noErr;
}
With these adjustments the pitch is now correct, and actually plays back on
both channels. Thanks so much Chad and Steve for putting me on the right track.
-António
> On 16 Apr 2015, at 19:56, Antonio Nunes <[email protected]> wrote:
>
> Ok, thanks for the lead. The files are multichannel indeed (stereo). So maybe
> it is an interleaving issue.
>
> (Originally I was using WAV files with 16 bit integers, but I changed them to
> float recently, because these days I should be able to use that format across
> Mac OS and iOS interchangeably. The wav files played back at the right pitch.)
>
> Below is my render callback, after adjusting it to Float32. Does it look like
> I am interleaving? (I suppose it may, since I am handling only a single
> channel, but how do I adjust this to handle both channels correctly?)
>
> static OSStatus multiChannelMixerRenderCallback(void *inRefCon,
>
> AudioUnitRenderActionFlags *ioActionFlags,
>
> const AudioTimeStamp *inTimeStamp,
>
> UInt32 inBusNumber,
>
> UInt32 inNumberFrames,
>
> AudioBufferList *ioData)
> {
> __unsafe_unretained SWAudioEngine *self = (__bridge SWAudioEngine
> *)inRefCon;
> uint32_t channel = 0;
> Float32 *out = (Float32 *)ioData->mBuffers[channel].mData;
>
> for (UInt32 i = 0; i < inNumberFrames; ++i) {
> MTSampleData data = MT_PopSample(inBusNumber);
> out[i] = data.sample;
> }
>
> return noErr;
> }
>
> MT_PopSample pops a data from a ring buffer. That data contains, among other
> info one sample retrieved from the audio file. The sample is retrieved from
> the audio file thus: self->audioData[self->packetIndex[voice]++]; So, a
> straight packet from the audioData that was read into memory when the file
> was loaded.
>
> -António
>
>> On 16 Apr 2015, at 19:26, Chad Wagner <[email protected]> wrote:
>>
>> Re: Steve's point #2:
>> I assume the files are multichannel (stereo)? You may also want to make sure
>> you're not interleaving 2 channels when you shouldn't be, and/or are setting
>> mNumberChannels correctly in your AudioBuffer.
>>
>>> On Apr 16, 2015, at 11:11 AM, Steve Bird <[email protected]> wrote:
>>>
>>>
>>>> On Apr 16, 2015, at 12:46 PM, Antonio Nunes <[email protected]>
>>>> wrote:
>>>> When my app plays the files, they sound fine, except for the fact that
>>>> they sound an octave (I guess) lower than they are supposed to. I suppose
>>>> something is not right with my conversion method. Maybe I shouldn’t be
>>>> using NSConvertHostFloatToSwapped to get the sample data into an
>>>> NSSwappedFloat format? Any ideas on achieving correct results?
>>>
>>>
>>> If they sound OK, except for the pitch, then you have one of two problems:
>>>
>>> 1… you've mishandled the sample rate somewhere. It’s recorded at 44100 and
>>> you’re playing back at 22050, or something similar.
>>> 2… you’re duplicating samples somewhere, reading one sample and turning it
>>> into two samples of playback.
>>>
>>>
>>>
>>> ----------------------------------------------------------------
>>> Steve Bird
>>> Culverson Software - Elegant software that is a pleasure to use.
>>> www.Culverson.com (toll free) 1-877-676-8175
>>>
>>>
>>>
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>>
>
>
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