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Commit db2fcee6 authored by Alexandre Savard's avatar Alexandre Savard
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[#[#5168] Must keep the g722 specific RTP rate to avoid incoming packet timeout

parent df4757de
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......@@ -122,23 +122,17 @@ void AudioRtpSession::setSessionMedia (AudioCodec *audioCodec)
_debug ("AudioRtpSession: Codec frame size: %d", frameSize);
_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) {
// _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));
if (payloadType == g722PayloadType) {
_debug ("AudioRtpSession: Setting G722 payload format");
setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
} else {
_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 {
_debug ("AudioRtpSession: Setting static payload format");
setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
}
}
// Set recording sampling rate
......@@ -168,26 +162,17 @@ void AudioRtpSession::updateSessionMedia (AudioCodec *audioCodec)
_debug ("AudioRtpSession: Codec frame size: %d", frameSize);
_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) {
// _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));
if (payloadType == g722PayloadType) {
_debug ("AudioRtpSession: Setting G722 payload format");
setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) payloadType, g722RtpClockRate));
} else {
_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 {
_debug ("AudioRtpSession: Setting static payload format");
setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) payloadType));
}
}
......
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