Remove RTP data implementation
Bug: webrtc:6625
Change-Id: Ie68d7a938d8b7be95a01cca74a176104e4e44e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215321
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33759}
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 5e7c225..a4f75e2 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -65,7 +65,6 @@
using cricket::RELAY_PORT_TYPE;
using cricket::RidDescription;
using cricket::RidDirection;
-using cricket::RtpDataContentDescription;
using cricket::SctpDataContentDescription;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
@@ -263,22 +262,6 @@
"a=ssrc:3 mslabel:local_stream_1\r\n"
"a=ssrc:3 label:video_track_id_1\r\n";
-static const char kSdpRtpDataChannelString[] =
- "m=application 9 RTP/SAVPF 101\r\n"
- "c=IN IP4 0.0.0.0\r\n"
- "a=rtcp:9 IN IP4 0.0.0.0\r\n"
- "a=ice-ufrag:ufrag_data\r\n"
- "a=ice-pwd:pwd_data\r\n"
- "a=mid:data_content_name\r\n"
- "a=sendrecv\r\n"
- "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
- "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5\r\n"
- "a=rtpmap:101 google-data/90000\r\n"
- "a=ssrc:10 cname:data_channel_cname\r\n"
- "a=ssrc:10 msid:data_channel data_channeld0\r\n"
- "a=ssrc:10 mslabel:data_channel\r\n"
- "a=ssrc:10 label:data_channeld0\r\n";
-
// draft-ietf-mmusic-sctp-sdp-03
static const char kSdpSctpDataChannelString[] =
"m=application 9 UDP/DTLS/SCTP 5000\r\n"
@@ -906,12 +889,6 @@
static const char kAudioTrackId3[] = "audio_track_id_3";
static const uint32_t kAudioTrack3Ssrc = 7;
-// DataChannel
-static const char kDataChannelLabel[] = "data_channel";
-static const char kDataChannelMsid[] = "data_channeld0";
-static const char kDataChannelCname[] = "data_channel_cname";
-static const uint32_t kDataChannelSsrc = 10;
-
// Candidate
static const char kDummyMid[] = "dummy_mid";
static const int kDummyIndex = 123;
@@ -1466,11 +1443,6 @@
simulcast2.receive_layers().size());
}
- void CompareRtpDataContentDescription(const RtpDataContentDescription* dcd1,
- const RtpDataContentDescription* dcd2) {
- CompareMediaContentDescription<RtpDataContentDescription>(dcd1, dcd2);
- }
-
void CompareSctpDataContentDescription(
const SctpDataContentDescription* dcd1,
const SctpDataContentDescription* dcd2) {
@@ -1521,14 +1493,6 @@
const SctpDataContentDescription* scd2 =
c2.media_description()->as_sctp();
CompareSctpDataContentDescription(scd1, scd2);
- } else {
- if (IsDataContent(&c1)) {
- const RtpDataContentDescription* dcd1 =
- c1.media_description()->as_rtp_data();
- const RtpDataContentDescription* dcd2 =
- c2.media_description()->as_rtp_data();
- CompareRtpDataContentDescription(dcd1, dcd2);
- }
}
CompareSimulcastDescription(
@@ -1816,28 +1780,6 @@
kDataContentName, TransportDescription(kUfragData, kPwdData)));
}
- void AddRtpDataChannel() {
- std::unique_ptr<RtpDataContentDescription> data(
- new RtpDataContentDescription());
- data_desc_ = data.get();
-
- data_desc_->AddCodec(DataCodec(101, "google-data"));
- StreamParams data_stream;
- data_stream.id = kDataChannelMsid;
- data_stream.cname = kDataChannelCname;
- data_stream.set_stream_ids({kDataChannelLabel});
- data_stream.ssrcs.push_back(kDataChannelSsrc);
- data_desc_->AddStream(data_stream);
- data_desc_->AddCrypto(
- CryptoParams(1, "AES_CM_128_HMAC_SHA1_80",
- "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", ""));
- data_desc_->set_protocol(cricket::kMediaProtocolSavpf);
- desc_.AddContent(kDataContentName, MediaProtocolType::kRtp,
- std::move(data));
- desc_.AddTransportInfo(TransportInfo(
- kDataContentName, TransportDescription(kUfragData, kPwdData)));
- }
-
bool TestDeserializeDirection(RtpTransceiverDirection direction) {
std::string new_sdp = kSdpFullString;
ReplaceDirection(direction, &new_sdp);
@@ -2103,7 +2045,6 @@
SessionDescription desc_;
AudioContentDescription* audio_desc_;
VideoContentDescription* video_desc_;
- RtpDataContentDescription* data_desc_;
SctpDataContentDescription* sctp_desc_;
Candidates candidates_;
std::unique_ptr<IceCandidateInterface> jcandidate_;
@@ -2269,18 +2210,6 @@
EXPECT_TRUE(TestSerializeRejected(true, true));
}
-TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithRtpDataChannel) {
- AddRtpDataChannel();
- JsepSessionDescription jsep_desc(kDummyType);
-
- MakeDescriptionWithoutCandidates(&jsep_desc);
- std::string message = webrtc::SdpSerialize(jsep_desc);
-
- std::string expected_sdp = kSdpString;
- expected_sdp.append(kSdpRtpDataChannelString);
- EXPECT_EQ(expected_sdp, message);
-}
-
TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithSctpDataChannel) {
bool use_sctpmap = true;
AddSctpDataChannel(use_sctpmap);
@@ -2327,22 +2256,6 @@
EXPECT_EQ(expected_sdp, message);
}
-TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
- JsepSessionDescription jsep_desc(kDummyType);
- AddRtpDataChannel();
- data_desc_->set_bandwidth(100 * 1000);
- data_desc_->set_bandwidth_type("AS");
- MakeDescriptionWithoutCandidates(&jsep_desc);
- std::string message = webrtc::SdpSerialize(jsep_desc);
-
- std::string expected_sdp = kSdpString;
- expected_sdp.append(kSdpRtpDataChannelString);
- // Serializing data content shouldn't ignore bandwidth settings.
- InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
- "b=AS:100\r\n", &expected_sdp);
- EXPECT_EQ(expected_sdp, message);
-}
-
TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapAllowMixed) {
jdesc_.description()->set_extmap_allow_mixed(true);
TestSerialize(jdesc_);
@@ -2913,21 +2826,6 @@
EXPECT_FALSE(SdpDeserializeCandidate(kSdpTcpInvalidCandidate, &jcandidate));
}
-TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannels) {
- AddRtpDataChannel();
- JsepSessionDescription jdesc(kDummyType);
- ASSERT_TRUE(jdesc.Initialize(desc_.Clone(), kSessionId, kSessionVersion));
-
- std::string sdp_with_data = kSdpString;
- sdp_with_data.append(kSdpRtpDataChannelString);
- JsepSessionDescription jdesc_output(kDummyType);
-
- // Deserialize
- EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
- // Verify
- EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
-}
-
TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannels) {
bool use_sctpmap = true;
AddSctpDataChannel(use_sctpmap);
@@ -3088,8 +2986,9 @@
}
TEST_F(WebRtcSdpTest, DeserializeSdpWithStrangeApplicationProtocolNames) {
- static const char* bad_strings[] = {"DTLS/SCTPRTP/", "obviously-bogus",
- "UDP/TL/RTSP/SAVPF", "UDP/TL/RTSP/S"};
+ static const char* bad_strings[] = {
+ "DTLS/SCTPRTP/", "obviously-bogus", "UDP/TL/RTSP/SAVPF",
+ "UDP/TL/RTSP/S", "DTLS/SCTP/RTP/FOO", "obviously-bogus/RTP/"};
for (auto proto : bad_strings) {
std::string sdp_with_data = kSdpString;
sdp_with_data.append("m=application 9 ");
@@ -3099,18 +2998,6 @@
EXPECT_FALSE(SdpDeserialize(sdp_with_data, &jdesc_output))
<< "Parsing should have failed on " << proto;
}
- // The following strings are strange, but acceptable as RTP.
- static const char* weird_strings[] = {"DTLS/SCTP/RTP/FOO",
- "obviously-bogus/RTP/"};
- for (auto proto : weird_strings) {
- std::string sdp_with_data = kSdpString;
- sdp_with_data.append("m=application 9 ");
- sdp_with_data.append(proto);
- sdp_with_data.append(" 47\r\n");
- JsepSessionDescription jdesc_output(kDummyType);
- EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output))
- << "Parsing should have succeeded on " << proto;
- }
}
// For crbug/344475.
@@ -3168,21 +3055,6 @@
EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
}
-TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
- // We want to test that deserializing data content limits bandwidth
- // settings (it should never be greater than the default).
- // This should prevent someone from using unlimited data bandwidth through
- // JS and "breaking the Internet".
- // See: https://code.google.com/p/chromium/issues/detail?id=280726
- std::string sdp_with_bandwidth = kSdpString;
- sdp_with_bandwidth.append(kSdpRtpDataChannelString);
- InjectAfter("a=mid:data_content_name\r\n", "b=AS:100\r\n",
- &sdp_with_bandwidth);
- JsepSessionDescription jdesc_with_bandwidth(kDummyType);
-
- EXPECT_FALSE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
-}
-
TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
bool use_sctpmap = true;
AddSctpDataChannel(use_sctpmap);
