Lower queued decode unit limit to resync faster if the renderers get behind. Lower the audio receive buffer size since it was unneccessarily large.

This commit is contained in:
Cameron Gutman 2014-01-22 17:01:37 -05:00
parent 46f4f5ccbe
commit 932bb1145b
3 changed files with 3 additions and 3 deletions

View File

@ -7,7 +7,7 @@ import com.limelight.nvstream.av.RtpPacket;
public class AudioDepacketizer {
private static final int DU_LIMIT = 15;
private static final int DU_LIMIT = 5;
private LinkedBlockingQueue<ByteBufferDescriptor> decodedUnits =
new LinkedBlockingQueue<ByteBufferDescriptor>(DU_LIMIT);

View File

@ -16,7 +16,7 @@ public class AudioStream {
public static final int RTP_PORT = 48000;
public static final int RTCP_PORT = 47999;
public static final int RTP_RECV_BUFFER = 64 * 1024;
public static final int RTP_RECV_BUFFER = 24 * 1024;
private DatagramSocket rtp;

View File

@ -23,7 +23,7 @@ public class VideoDepacketizer {
private ConnectionStatusListener controlListener;
private static final int DU_LIMIT = 15;
private static final int DU_LIMIT = 5;
private LinkedBlockingQueue<DecodeUnit> decodedUnits = new LinkedBlockingQueue<DecodeUnit>(DU_LIMIT);
public VideoDepacketizer(ConnectionStatusListener controlListener)