<div>Hi!</div><div><br></div><div>I'm not really sure which side is responsible for the problem... More like that I need your advice what to do and investigate next.</div><div><br></div><div>I use live555 to receive video and audio streams using RTSP client. The problem is: when I fill audio buffer with audio-frames according to their timestamps, I get a small gaps between frames in the buffer. That causes unacceptable artefacts while playing. From the other side, if I simply concatenate incoming audio-frames, I got clear audio without artefacts. The problem is protocol-independent, I can hear the same artefacts on both TCP and UDP protocols.</div>
<div><br></div><div>Audio is ì-law encoded.</div><div><br></div><div>Here (<a href="http://www.sendspace.com/file/ba7f3n">http://www.sendspace.com/file/ba7f3n</a>) you can download small archive (~200 kilobytes) with two samples:</div>
<div><br></div><div>"solid.wav" created via simple concatenation of incoming audio-frames. Plays clear, no artefacts.<br><br>"assembled.wav" created using prerequisite buffer, where incoming audio-frames were arranged according to their timestamps. Plays with artefacts. If you'll open this file in hex-editor you will see small null-filled gaps.</div>
<div><br></div><div>I can't use simple concatenation method because of synchronization issue. What should I do, what solutions can fix that problem?<br></div>