audacity

I have written an audio WAV (have recorded my voice) file using AVAudioRecorder. The final file format is a WAV file. File was successfully saved and I can hear my voice. I want to send this file to a back End server (web-service). But my server accepting only data and FMT information in WAV . It is not accepting my wav file because of my wav file infromation with FLLR, data, FMT. I have checked my WAV file information in Riffpad tool. It's showing FLLR, data, FMT. But I want only data and fmt. Because my server side accepts only data and FMT. So please advice how to remove FLLR in my wav file

I want to increase a volume on buffer with voice data. The point is I'm using DirectSound and I have one primary and one secondary buffer - all streams mixing is done by hand. In a voice chat all participants can have independent volume levels. I multiply each stream data by a value (gain) and sum it to one buffer. Everything works fine but when I try to multiply data by a value greater than 1.0f - I hear some clipping or what. I've tried using Audacity effect compressor but this doesn't help reducing the strange noise. Probably I should modify gain in some other way? Or just use another post

I am having a problem With SoundPool as it refuses to work with my .ogg files. I am getting this error: AudioFlinger could not create track, status: -12 Error creating AudioTrack I've found a thread concerning this and the (possible) answer is: Make sure you use .ogg media files with constant bitrate! Is this the case? (someone please confirm or cancel) If yes - which application to use (Audacity doesn't support .ogg custom export settings). If not - what else could be wrong? As a side note - before I used MediaPlayer but now I want to play a few sounds parallel. MAJOR EDIT: FOUND A WAY TO FIX