commit 479a3c0f92b957a9ba1c242e7fe6e4d37b49ce32
author Mirta Dvornicic <mirtad@webrtc.org> Tue Jun 04 13:38:50 2019
committer Commit Bot <commit-bot@chromium.org> Tue Jun 04 14:35:54 2019
tree 127e3a2cf03b3c3568cb3e235c780dc9cd24f93e
parent 961407f5e86fa4429086dc0c532ab9ceebc4623d

Add support for enabling and negotiating raw RTP packetization.

Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.

Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}

pc/webrtc_sdp_unittest.cc

diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 4b7a6d9..5fdf5f6 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -3418,6 +3418,54 @@
   EXPECT_EQ(found->second, "40");
 }
 
+TEST_F(WebRtcSdpTest, DeserializePacketizationAttributeWithIllegalValue) {
+  JsepSessionDescription jdesc_output(kDummyType);
+
+  const char kSdpWithPacketizationString[] =
+      "v=0\r\n"
+      "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
+      "s=-\r\n"
+      "t=0 0\r\n"
+      "m=audio 9 RTP/SAVPF 111\r\n"
+      "a=rtpmap:111 opus/48000/2\r\n"
+      "a=packetization:111 unknownpacketizationattributeforaudio\r\n"
+      "m=video 3457 RTP/SAVPF 120 121 122\r\n"
+      "a=rtpmap:120 VP8/90000\r\n"
+      "a=packetization:120 raw\r\n"
+      "a=rtpmap:121 VP9/90000\r\n"
+      "a=rtpmap:122 H264/90000\r\n"
+      "a=packetization:122 unknownpacketizationattributevalue\r\n";
+
+  SdpParseError error;
+  EXPECT_TRUE(webrtc::SdpDeserialize(kSdpWithPacketizationString, &jdesc_output,
+                                     &error));
+
+  AudioContentDescription* acd =
+      GetFirstAudioContentDescription(jdesc_output.description());
+  ASSERT_TRUE(acd);
+  ASSERT_THAT(acd->codecs(), testing::SizeIs(1));
+  cricket::AudioCodec opus = acd->codecs()[0];
+  EXPECT_EQ(opus.name, "opus");
+  EXPECT_EQ(opus.id, 111);
+
+  const VideoContentDescription* vcd =
+      GetFirstVideoContentDescription(jdesc_output.description());
+  ASSERT_TRUE(vcd);
+  ASSERT_THAT(vcd->codecs(), testing::SizeIs(3));
+  cricket::VideoCodec vp8 = vcd->codecs()[0];
+  EXPECT_EQ(vp8.name, "VP8");
+  EXPECT_EQ(vp8.id, 120);
+  EXPECT_EQ(vp8.packetization, "raw");
+  cricket::VideoCodec vp9 = vcd->codecs()[1];
+  EXPECT_EQ(vp9.name, "VP9");
+  EXPECT_EQ(vp9.id, 121);
+  EXPECT_EQ(vp9.packetization, absl::nullopt);
+  cricket::VideoCodec h264 = vcd->codecs()[2];
+  EXPECT_EQ(h264.name, "H264");
+  EXPECT_EQ(h264.id, 122);
+  EXPECT_EQ(h264.packetization, absl::nullopt);
+}
+
 TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithUnknownParameter) {
   AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
 
@@ -3486,6 +3534,22 @@
   EXPECT_EQ(sdp_with_fmtp, message);
 }
 
+TEST_F(WebRtcSdpTest, SerializeVideoPacketizationAttribute) {
+  VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
+
+  cricket::VideoCodecs codecs = vcd->codecs();
+  codecs[0].packetization = "raw";
+  vcd->set_codecs(codecs);
+
+  ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
+                                jdesc_.session_version()));
+  std::string message = webrtc::SdpSerialize(jdesc_);
+  std::string sdp_with_packetization = kSdpFullString;
+  InjectAfter("a=rtpmap:120 VP8/90000\r\n", "a=packetization:120 raw\r\n",
+              &sdp_with_packetization);
+  EXPECT_EQ(sdp_with_packetization, message);
+}
+
 TEST_F(WebRtcSdpTest, DeserializeAndSerializeSdpWithIceLite) {
   // Deserialize the baseline description, making sure it's ICE full.
   JsepSessionDescription jdesc_with_icelite(kDummyType);