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savoirfairelinux
jami-daemon
Commits
1ea4a1a1
Commit
1ea4a1a1
authored
14 years ago
by
Alexandre Savard
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[#5168] Fix static/dynamic payload rtp session update
parent
7ad64053
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sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+27
-10
27 additions, 10 deletions
sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
with
27 additions
and
10 deletions
sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+
27
−
10
View file @
1ea4a1a1
...
...
@@ -123,13 +123,20 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
_debug
(
"AudioRtpSession: RTP timestamp increment: %d"
,
_timestampIncrement
);
// Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
if
(
payloadType
==
g722PayloadType
)
{
_debug
(
"AudioRtpSession: Setting G722 payload format"
);
setPayloadFormat
(
ost
::
DynamicPayloadFormat
(
(
ost
::
PayloadType
)
payloadType
,
g722RtpClockRate
));
}
else
if
(
dynamic
)
{
// if (payloadType == g722PayloadType) {
// _debug ("AudioRtpSession: Setting G722 payload format");
// setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
// } else if (dynamic) {
// _debug ("AudioRtpSession: Setting dynamic payload format");
// setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
// } else if (dynamic && payloadType != g722PayloadType) {
// _debug ("AudioRtpSession: Setting static payload format");
// setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
// }
if
(
dynamic
)
{
_debug
(
"AudioRtpSession: Setting dynamic payload format"
);
setPayloadFormat
(
ost
::
DynamicPayloadFormat
(
(
ost
::
PayloadType
)
payloadType
,
smplRate
));
}
else
if
(
dynamic
&&
payloadType
!=
g722PayloadType
)
{
}
else
{
_debug
(
"AudioRtpSession: Setting static payload format"
);
setPayloadFormat
(
ost
::
StaticPayloadFormat
(
(
ost
::
StaticPayloadType
)
payloadType
));
}
...
...
@@ -162,13 +169,23 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
_debug
(
"AudioRtpSession: RTP timestamp increment: %d"
,
_timestampIncrement
);
// Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
if
(
payloadType
==
g722PayloadType
)
{
_debug
(
"AudioRtpSession: Setting G722 payload format"
);
setPayloadFormat
(
ost
::
DynamicPayloadFormat
(
(
ost
::
PayloadType
)
payloadType
,
g722RtpClockRate
));
}
else
if
(
dynamic
)
{
// if (payloadType == g722PayloadType) {
// _debug ("AudioRtpSession: Setting G722 payload format");
// setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
// } else if (dynamic) {
// _debug ("AudioRtpSession: Setting dynamic payload format");
// setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, smplRate));
// } else if (dynamic && payloadType != g722PayloadType) {
// _debug ("AudioRtpSession: Setting static payload format");
// setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
// } else {
// _debug ("Did not enter any of above case");
// }
if
(
dynamic
)
{
_debug
(
"AudioRtpSession: Setting dynamic payload format"
);
setPayloadFormat
(
ost
::
DynamicPayloadFormat
(
(
ost
::
PayloadType
)
payloadType
,
smplRate
));
}
else
if
(
dynamic
&&
payloadType
!=
g722PayloadType
)
{
}
else
{
_debug
(
"AudioRtpSession: Setting static payload format"
);
setPayloadFormat
(
ost
::
StaticPayloadFormat
(
(
ost
::
StaticPayloadType
)
payloadType
));
}
...
...
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