Expose setPayloadType on the rtp_sender. Thus allowing other users of this module
to set the payload type to be used without having to call SendOutgoingData.
BUG=3694
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index d5bffa9..f544db2 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -283,6 +283,11 @@
return 0;
}
+void RTPSender::SetSendPayloadType(int8_t payload_type) {
+ CriticalSectionScoped cs(send_critsect_);
+ payload_type_ = payload_type;
+}
+
int8_t RTPSender::SendPayloadType() const {
CriticalSectionScoped cs(send_critsect_);
return payload_type_;
@@ -385,7 +390,7 @@
LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
return -1;
}
- payload_type_ = payload_type;
+ SetSendPayloadType(payload_type);
RtpUtility::Payload* payload = it->second;
assert(payload);
if (!payload->audio && !audio_configured_) {
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 39bcb0c..e4d4fca 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -100,6 +100,8 @@
int32_t DeRegisterSendPayload(const int8_t payload_type);
+ void SetSendPayloadType(int8_t payload_type);
+
int8_t SendPayloadType() const;
int SendPayloadFrequency() const;
