Add SdpHelper for more reliable codec configuration in WebrtcTransport.

remoting/protocol/sdp_message_unittest.cc

+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "remoting/protocol/sdp_message.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace remoting {
+namespace protocol {
+
+// Verify that SDP is normalized by removing empty lines and normalizing
+// line-endings to \n.
+TEST(SdpMessages, Normalize) {
+  SdpMessage sdp_message("a=foo\n\r\nb=bar");
+  EXPECT_EQ("a=foo\nb=bar\n", sdp_message.ToString());
+}
+
+TEST(SdpMessages, AddCodecParameter) {
+  std::string kSourceSdp =
+      "a=group:BUNDLE video\n"
+      "m=video 9 UDP/TLS/RTP/SAVPF 96\n"
+      "a=rtpmap:96 VP8/90000\n"
+      "a=rtcp-fb:96 transport-cc\n";
+  SdpMessage sdp_message(kSourceSdp);
+  EXPECT_TRUE(sdp_message.AddCodecParameter("VP8", "test_param=1"));
+  EXPECT_EQ(
+      "a=group:BUNDLE video\n"
+      "m=video 9 UDP/TLS/RTP/SAVPF 96\n"
+      "a=rtpmap:96 VP8/90000\n"
+      "a=fmtp:96 test_param=1\n"
+      "a=rtcp-fb:96 transport-cc\n",
+      sdp_message.ToString());
+}
+
+TEST(SdpMessages, AddCodecParameterMissingCodec) {
+  std::string kSourceSdp =
+      "a=group:BUNDLE audio video\n"
+      "m=audio 9 UDP/TLS/RTP/SAVPF 111\n"
+      "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\n"
+      "a=rtpmap:111 opus/48000/2\n"
+      "a=rtcp-fb:111 transport-cc\n"
+      "m=video 9 UDP/TLS/RTP/SAVPF 96\n"
+      "a=rtpmap:96 VP9/90000\n"
+      "a=rtcp-fb:96 transport-cc\n"
+      "m=application 9 DTLS/SCTP 5000\n";
+  SdpMessage sdp_message(kSourceSdp);
+  EXPECT_FALSE(sdp_message.AddCodecParameter("VP8", "test_param=1"));
+  EXPECT_EQ(kSourceSdp, sdp_message.ToString());
+}
+
+}  // namespace protocol
+}  // namespace remoting