commit 1557a0b49326149822612e65b72bd6038e6776a5
author sprang@webrtc.org <sprang@webrtc.org> Tue Mar 17 16:42:49 2015
committer sprang@webrtc.org <sprang@webrtc.org> Tue Mar 17 16:44:54 2015
tree df1a34f5f9c8bbb17d204a0a050b88bad3592c15
parent 9a479723ab76d993c47fc569b4b4ae6054245962

Use ByteReader/ByteWriter instead of rtputility and manual shift/add.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41289004

Cr-Original-Commit-Position: refs/heads/master@{#8761}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 779c3d16b9623f38a72439bc013102aeb0077a62

modules/rtp_rtcp/test/testAPI/test_api_video.cc

diff --git a/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index e8d4320..e28d5ce 100644
--- a/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -18,8 +18,8 @@
 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 
 namespace {
@@ -89,9 +89,9 @@
                          uint32_t sequence_number) {
     dataBuffer[0] = static_cast<uint8_t>(0x80);  // version 2
     dataBuffer[1] = static_cast<uint8_t>(kPayloadType);
-    RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number);
-    RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp);
-    RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234);  // SSRC.
+    ByteWriter<uint16_t>::WriteBigEndian(dataBuffer + 2, sequence_number);
+    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 4, timestamp);
+    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 8, 0x1234);  // SSRC.
     size_t rtpHeaderLength = 12;
     return rtpHeaderLength;
   }