[asterisk-scf-commits] asterisk-scf/integration/media_rtp_pjmedia.git branch "master" updated.

[Commits to the Asterisk SCF project code repositories]

branch "master" has been updated
       via  c34444cbf9e8d293c5a4053a8a51c8fcf1f6b5bc (commit)
      from  4bf3ca690bee921a65dc38ac61431c696d4d3284 (commit)

Summary of changes:
 src/RTPSink.cpp |    2 +-
 1 files changed, 1 insertions(+), 1 deletions(-)


- Log -----------------------------------------------------------------
commit c34444cbf9e8d293c5a4053a8a51c8fcf1f6b5bc
Author: Joshua Colp <jcolp at digium.com>
Date:   Thu Aug 26 12:17:11 2010 -0300

    Pass in the correct value so the timestamp is incremented by the right amount.

diff --git a/src/RTPSink.cpp b/src/RTPSink.cpp
index e53796b..ae1611b 100644
--- a/src/RTPSink.cpp
+++ b/src/RTPSink.cpp
@@ -96,7 +96,7 @@ void StreamSinkRTPImpl::write(const Hydra::Media::V1::FrameSeq& frames, const Ic
 
 		/* Using the available information construct an RTP header that we can place at the front of our packet */
 		pj_status_t status = pjmedia_rtp_encode_rtp(&mImpl->mOutgoingSession, mImpl->mSession->getPayload((*frame)->mediaformat), 0, (*frame)->payload.size(),
-							    audioformat->frameSize, &header, &header_len);
+							    (*frame)->payload.size(), &header, &header_len);
 
 		if (status != PJ_SUCCESS)
 		{

-----------------------------------------------------------------------


-- 
asterisk-scf/integration/media_rtp_pjmedia.git