Here is my problem. I initialize the waveOut API with a sampling rate of 44100 so thatI can play CD-quality audio, but when I try to play any .WAV file that has a sampling rate of 22050 (most built-in windows sounds), the file plays twice as fast. I know that this is suppoised to happen but my question is how can I play files with any sampling rate if I initialize the API with 44100? I am doing streaming and non-streaming audio under Windows. I want my streams to be at 44100 but the other files I play can be encoded at less that that. Is this possible? Regards, Kinfira Lead Programmer KinfiraSoft Makers of Swordplay, a 3D fighting game engine. Check out our home on the web: http://www.kinfirasoft.com

Sorry about the lateness of my reply.

quote:Stoffel said:
However, what do you do with 16kHz and 8kHz files?

At the moment, they will also not play correctly.

quote:
Is closing & reopening the sound device completely out of the question?

The main idea behind all this is that I want to play streamed background music at 44100kHz but some of my sound effects may be encoded at a lower sampling rate. I can''t close and reopen the sound device without terminating the already playing music. One solution would be to require all music and sound be encoded at 44100kHz but I was hoping to avoid that.

quote:Nytegard said:
What you can do:

When reading the wave file into memory, determine the sampling rate before you begin streaming.

Based on the sampling rate, you can still play at 44.1k, but just make sure to send a correct ratio of data to trick it into thinking its playing 44.1k.

Any idea on how to make it do this? Do I simply need to mangle the sound data somehow? BTW I am doing this currently with uncompressed PCM data.

Thank for the help given so far.



Regards,

Kinfira
Lead Programmer
KinfiraSoft
Makers of Swordplay, a 3D fighting game engine.

Check out our home on the web:
http://www.kinfirasoft.com