commit 2ec560606be6519dc4e32a1e6855b0f362ca498d
author stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Jul 31 14:59:24 2014 +0000
committer stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Thu Jul 31 14:59:24 2014 +0000
tree 1514bac7209592adb9d466a81135cc964aed3b3f
parent bfe6e081952540bd74b62627a3b910208e33c490

Add H.264 packetization.

This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.

R=niklas.enbom@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/modules.gyp

diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index 2a3ba74..10a1860 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -206,6 +206,7 @@
             'rtp_rtcp/source/rtcp_receiver_unittest.cc',
             'rtp_rtcp/source/rtcp_sender_unittest.cc',
             'rtp_rtcp/source/rtp_fec_unittest.cc',
+            'rtp_rtcp/source/rtp_format_h264_unittest.cc',
             'rtp_rtcp/source/rtp_format_vp8_unittest.cc',
             'rtp_rtcp/source/rtp_format_vp8_test_helper.cc',
             'rtp_rtcp/source/rtp_format_vp8_test_helper.h',