Rename RTP-extension constants.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/config.cc

@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/config.h"
+
+namespace webrtc {
+const char* RtpExtension::kTOffset = "urn:ietf:params:rtp-hdrext:toffset";
+const char* RtpExtension::kAbsSendTime =
+    "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
+}  // namespace webrtc

webrtc/config.h

@@ -16,6 +16,8 @@
 #include <string>
 #include <vector>
 
+#include "webrtc/typedefs.h"
+
 namespace webrtc {
 
 struct RtpStatistics {
@@ -67,6 +69,8 @@ struct RtxConfig {
 
 // RTP header extension to use for the video stream, see RFC 5285.
 struct RtpExtension {
+  static const char* kTOffset;
+  static const char* kAbsSendTime;
   RtpExtension(const char* name, int id) : name(name), id(id) {}
   // TODO(mflodman) Add API to query supported extensions.
   std::string name;

webrtc/video/rampup_tests.cc

@@ -155,7 +155,7 @@ TEST_P(RampUpTest, RampUpWithPadding) {
   send_config.codec.plType = 125;
   send_config.pacing = GetParam();
   send_config.rtp.extensions.push_back(
-      RtpExtension("toffset", kTOffsetExtensionId));
+      RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
 
   test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
 

webrtc/video/video_send_stream.cc

@@ -123,10 +123,10 @@ VideoSendStream::VideoSendStream(newapi::Transport* transport,
   for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) {
     const std::string& extension = config_.rtp.extensions[i].name;
     int id = config_.rtp.extensions[i].id;
-    if (extension == "toffset") {
+    if (extension == RtpExtension::kTOffset) {
       if (rtp_rtcp_->SetSendTimestampOffsetStatus(channel_, true, id) != 0)
         abort();
-    } else if (extension == "abs-send-time") {
+    } else if (extension == RtpExtension::kAbsSendTime) {
       if (rtp_rtcp_->SetSendAbsoluteSendTimeStatus(channel_, true, id) != 0)
         abort();
     } else {

webrtc/video/video_send_stream_tests.cc

@@ -170,7 +170,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
 
   VideoSendStream::Config send_config = GetSendTestConfig(call.get());
   send_config.rtp.extensions.push_back(
-      RtpExtension("abs-send-time", kAbsSendTimeExtensionId));
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
 
   RunSendTest(call.get(), send_config, &observer);
 }
@@ -216,7 +216,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
   VideoSendStream::Config send_config = GetSendTestConfig(call.get());
   send_config.encoder = &encoder;
   send_config.rtp.extensions.push_back(
-      RtpExtension("toffset", kTOffsetExtensionId));
+      RtpExtension(RtpExtension::kTOffset, kTOffsetExtensionId));
 
   RunSendTest(call.get(), send_config, &observer);
 }

webrtc/webrtc.gyp

@@ -61,6 +61,7 @@
       'sources': [
         'call.cc',
         'call.h',
+        'config.cc',
         'config.h',
         'frame_callback.h',
         'transport.h',