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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.