Stop using hardcoded payload types for video codecs

webrtc/api/webrtcsdp_unittest.cc

 /*
  *  Copyright 2011 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.
  */
 
@@ skipping 2065 matching lines @@
                       kCandidateFoundation1);
   candidate.set_tcptype(cricket::TCPTYPE_ACTIVE_STR);
   std::unique_ptr<IceCandidateInterface> jcandidate(
       new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
 
   std::string message = webrtc::SdpSerializeCandidate(*jcandidate);
   EXPECT_EQ(std::string(kSdpTcpActiveCandidate), message);
 }
 
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithH264) {
-  if (!webrtc::H264Encoder::IsSupported())
-    return;
-  for (const auto& codec : cricket::DefaultVideoCodecList()) {
-    video_desc_->AddCodec(codec);
-  }
+  cricket::VideoCodec h264_codec("H264");

[hta-webrtc, 2016/11/15 11:34:59]

Does this compile & run if you're compiling without H.264 support?

[magjed_webrtc, 2016/11/15 17:18:26]

Yes, it doesn't use the actual encoder, it just serializes the
cricket::VideoCodec.

+  h264_codec.SetParam("profile-level-id", "42e01f");

[hta-webrtc, 2016/11/15 11:34:59]

Can you use the constructor and stringifier for profile-level-id, so that it's
easier to see what you're testing?

[magjed_webrtc, 2016/11/15 17:18:26]

I can, but in the test we are checking:
EXPECT_NE(std::string::npos, fmtp_value.find("profile-level-id=42e01f"));
so maybe it's clearer to use the same string here?

[hta-webrtc, 2016/11/15 20:36:24]

Acknowledged.

+  h264_codec.SetParam("level-asymmetry-allowed", "1");
+  h264_codec.SetParam("packetization-mode", "1");
+  video_desc_->AddCodec(h264_codec);

[stefan-webrtc, 2016/11/15 09:00:08]

Could you explain why this change is needed? Do I have to review code in
webrtc/media to understand it?

[magjed_webrtc, 2016/11/15 17:18:26]

This change is needed because cricket::DefaultVideoCodecList() does not exist
anymore. Also, I don't think this unittest should have any dependency to a real
codec implementation. We should only be testing serialization of session
description with H264. So basically, the old 'cricket::DefaultVideoCodecList()'
is faked here with only the H264 codec.

+
   jdesc_.Initialize(desc_.Copy(), kSessionId, kSessionVersion);
 
   std::string message = webrtc::SdpSerialize(jdesc_, false);
   size_t after_pt = message.find(" H264/90000");
   ASSERT_NE(after_pt, std::string::npos);
   size_t before_pt = message.rfind("a=rtpmap:", after_pt);
   ASSERT_NE(before_pt, std::string::npos);
   before_pt += strlen("a=rtpmap:");
   std::string pt = message.substr(before_pt, after_pt - before_pt);
   // TODO(hta): Check if payload type |pt| occurs in the m=video line.
@@ skipping 1092 matching lines @@
       "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
       "s=-\r\n"
       "t=0 0\r\n"
       "m=video 9 UDP/DTLS/SCTP 120\r\n"
       "a=sctp-port 5000\r\n"
       "a=fmtp:108 foo=10\r\n";
 
   ExpectParseFailure(std::string(kSdpWithSctpPortInVideoDescription),
                      "sctp-port");
 }