RTP Audio System
2.0.0
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#include "audiowriterinterface.h"
#include "spectrumanalyzer.h"
#include "audiomixer.h"
#include "tools.h"
#include "strings.h"
#include "audioclient.h"
#include <qapplication.h>
#include <qlayout.h>
#include <qpushbutton.h>
#include <qscrollbar.h>
#include <qlineedit.h>
#include <qbuttongroup.h>
#include <qcheckbox.h>
#include <qcombobox.h>
#include <qradiobutton.h>
#include <qframe.h>
#include <qgroupbox.h>
#include <qlabel.h>
#include <qlcdnumber.h>
#include <qwhatsthis.h>
#include <qmainwindow.h>
#include <qlist.h>
#include "qspectrumanalyzer.h"
#include "qaudiomixer.h"
#include "qinfotabwidget.h"
Go to the source code of this file.
Classes | |
class | QClient |
QClient. More... | |
Variables | |
const InfoEntry | InfoEntries1 [] |
const InfoTable | InfoTable1 |
const InfoEntry | InfoEntries2 [] |
const InfoTable | InfoTable2 |
const InfoEntry InfoEntries1[] |
{ {"SA", "Server Address", "This are the IPv4 or IPv6 address and port number of the audio server."}, {"TF", "TOS/Flow Label", "This are the TOS/traffic class values for each layer and the flow label " "(IPv6 only) of the received packets."}, {"SSSRC", "Server SSRC", "This is the audio server's RTP SSRC. It is a 32-bit random number."}, {"CA", "Client Address", "This is the IPv4 or IPv6 address and port number of the audio client."}, {"CSSRC", "Client SSRC", "This is the audio clients's RTP SSRC. It is a 32-bit random number."}, {"BR", "Bytes Received", "This is a counter for the number of bytes received from server " "(IP/UDP/RTP/RTP Audio headers + payload)."}, {"PR", "Packets Received", "This is a counter for the number of packets received from server. " "The bytes per second value is the value for the quality received " "from server (IP/UDP/RTP/RTP Audio headers + payload)."}, {"PL", "Packets Lost", "This is a counter for the number of packets lost during transmission." "The loss fraction shows the fraction of packets lost during the last " "RTCP report interval in each layer."}, {"IJ", "Interarrival Jitter", "This is the interarrival jitter: An estimate of the statistical variance of " "the RTP data packet interarrival time, measured in milliseconds.\n\n" "Definition:\n" "Let Si, Sj be the RTP timestamps of packets i, j.\n" "Let Ri, Rj be the arrival timestamps.\n" "Dij := (Rj - Sj) - (Ri - Si).\n" "Jitter := Jitter + (1.0/16.0) * abs(Dij).\n\n" "See RFC 1889, Page 25-26 for more details."}, {"Q", "Quality", "This is the audio quality received from server: Sampling rate, bits and channels."}, {"E", "Encoding", "This is the name of the audio encoding format received from server."}, }
Transmission status info table #1 entries.
const InfoEntry InfoEntries2[] |
{ {"LSA", "Source", "This is the current layer's source address and port number."}, {"LTF", "TOS/Flow Label", "This are the current layer's traffic class and flow label (IPv6 only)."}, {"CA", "Destination", "This is the current layer's destination address and port number."}, {"LPR", "Packets Received", "This is the number of packets received in this layer."}, {"LPL", "Packets Lost", "This is the number of packets lost in this layer."}, {"LFL", "Fraction Lost", "This is the fraction of packets lost in this layer."}, {"LBR", "Bytes Received", "This is the sum of bytes received in this layer."}, {"LIJ", "Interarrival Jitter", "This is the interarrival jitter of this layer."}, {"Q", "Quality", "This is the audio quality received from server: Sampling rate, bits and channels."}, {"E", "Encoding", "This is the name of the audio encoding format received from server."}, }
Transmission status info table #1 entries.
const InfoTable InfoTable1 |
{ sizeof(InfoEntries1) / sizeof(InfoEntry), (const InfoEntry*)&InfoEntries1 }
Transmission status info table #1.
const InfoTable InfoTable2 |
{ sizeof(InfoEntries2) / sizeof(InfoEntry), (const InfoEntry*)&InfoEntries2 }
Transmission status info table #2.