[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  75eb6574fec804f6f5e737fd9b9853a36d7fde43 (commit)
      from  695c4fef0eebb0e0c433f2f6856f81053ce7b86d (commit)

Summary of changes:
 src/RTPSink.cpp |    1 -
 1 files changed, 0 insertions(+), 1 deletions(-)


- Log -----------------------------------------------------------------
commit 75eb6574fec804f6f5e737fd9b9853a36d7fde43
Author: Joshua Colp <jcolp at digium.com>
Date:   Mon Aug 16 15:36:05 2010 -0300

    Remove a comment which is no longer unapplicable.

diff --git a/src/RTPSink.cpp b/src/RTPSink.cpp
index a066d58..06b0ea4 100644
--- a/src/RTPSink.cpp
+++ b/src/RTPSink.cpp
@@ -85,7 +85,6 @@ void StreamSinkRTPImpl::write(const Hydra::Media::V1::FrameSeq& frames, const Ic
 		int header_len;
 
 		/* Using the available information construct an RTP header that we can place at the front of our packet */
-		/* TODO: We need to be able to set the RTP payload value here, from the frame? otherwise? */
 		pj_status_t status = pjmedia_rtp_encode_rtp(&mImpl->mOutgoingSession, mImpl->mSession->getPayload((*frame)->mediaformat), 0, (*frame)->payload.size(),
 							    audioformat->frameSize, &header, &header_len);
 

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-- 
asterisk-scf/integration/media_rtp_pjmedia.git