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author | hubbe@chromium.org <hubbe@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-08-18 20:04:54 +0000 |
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committer | hubbe@chromium.org <hubbe@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98> | 2014-08-18 20:07:11 +0000 |
commit | 8e464358387df04505304ae7b82875f7b6ba79f5 (patch) | |
tree | a4b7c91e8f6a66183b60fb95a29d3efd7fa59c1e /media/cast/sender/video_sender_unittest.cc | |
parent | 69c00da6c228c8bf556b952d704068ab9614af6a (diff) | |
download | chromium_src-8e464358387df04505304ae7b82875f7b6ba79f5.zip chromium_src-8e464358387df04505304ae7b82875f7b6ba79f5.tar.gz chromium_src-8e464358387df04505304ae7b82875f7b6ba79f5.tar.bz2 |
Rewrite of RTCP parser.
Clears out of lot of cruft and data copying.
Makes the parser easier to read.
Parser uses BigEndianReader for everything, which makes security easier to verify.
Removes ~1000 lines of code.
BUG=396207
Review URL: https://codereview.chromium.org/477593003
Cr-Commit-Position: refs/heads/master@{#290348}
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@290348 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'media/cast/sender/video_sender_unittest.cc')
-rw-r--r-- | media/cast/sender/video_sender_unittest.cc | 3 |
1 files changed, 1 insertions, 2 deletions
diff --git a/media/cast/sender/video_sender_unittest.cc b/media/cast/sender/video_sender_unittest.cc index 5522c26..a95f3e5 100644 --- a/media/cast/sender/video_sender_unittest.cc +++ b/media/cast/sender/video_sender_unittest.cc @@ -15,7 +15,6 @@ #include "media/cast/net/cast_transport_config.h" #include "media/cast/net/cast_transport_sender_impl.h" #include "media/cast/net/pacing/paced_sender.h" -#include "media/cast/net/rtcp/rtcp_receiver.h" #include "media/cast/sender/video_sender.h" #include "media/cast/test/fake_single_thread_task_runner.h" #include "media/cast/test/fake_video_encode_accelerator.h" @@ -67,7 +66,7 @@ class TestPacketSender : public PacketSender { callback_ = cb; return false; } - if (RtcpReceiver::IsRtcpPacket(&packet->data[0], packet->data.size())) { + if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { ++number_of_rtcp_packets_; } else { // Check that at least one RTCP packet was sent before the first RTP |