Your design is way too simple and fundamentally wrong.
If you want low latency you need to use a pull model (aka callback model)
for audio i/o to the device. Let the device tell you when it wants audio
data, and deliver it,on time, without blocking (which means no on-demand
file i/o in the same thread as the device audio i/o). See
http://www.rossbencina.com/code/real-time-audio-programming-101-time-waits-for-nothing
On Tue, May 17, 2016 at 4:26 AM, David Griffith <dave@email-addr-hidden> wrote:
> On Tue, 17 May 2016, Andrea Del Signore wrote:
>
> On Tue, May 17, 2016 at 12:25 AM, David Griffith <dave@email-addr-hidden> wrote:
>> On Mon, 16 May 2016, Andrea Del Signore wrote:
>>
>> > I'm not simply trying to mix two files. My main project is a
>> > game engine in which two sounds are allowed at any one time.
>> > For instance, there can be constant background music punctuated
>> > by sound effects. I can't get these to mix correctly.
>>
>> Hi,
>>
>> in that case you can just skip the right number of frames before starting
>> playing sounds.
>>
>> I modified my code to take the start time for each file and schedule the
>> play time with frame accuracy.
>>
>> http://pastebin.com/0PMyfPvK
>>
>> If you want your timing to be sample accurate the algorithm is a bit more
>> complex.
>>
>
> That won't work. Your code schedules things ahead of time before anything
> else happens. I need to be able to fire off sound effects the instant the
> player does something to cause them. I can't know in advance.
>
> --
> David Griffith
> dave@email-addr-hidden
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> http://lists.linuxaudio.org/listinfo/linux-audio-dev
>
>
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Received on Tue May 17 16:15:01 2016
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