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Remove RTP data implementation

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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}
This commit is contained in:
Harald Alvestrand
2021-04-16 11:12:14 +00:00
committed by Commit Bot
parent f981cb3d2e
commit 7af57c6e48
41 changed files with 87 additions and 3487 deletions
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134
pc/webrtc_sdp_unittest.cc
View File

@ -65,7 +65,6 @@ using cricket::MediaProtocolType;
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 @@ static const char kSdpString[] =
"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 uint32_t kVideoTrack3Ssrc = 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 @@ class WebRtcSdpTest : public ::testing::Test {
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 @@ class WebRtcSdpTest : public ::testing::Test {
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 @@ class WebRtcSdpTest : public ::testing::Test {
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 @@ class WebRtcSdpTest : public ::testing::Test {
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 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithAudioVideoRejected) {
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 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
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 @@ TEST_F(WebRtcSdpTest, DeserializeInvalidCandidiate) {
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, DeserializeSdpWithRtpmapAttribute) {
}
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 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithStrangeApplicationProtocolNames) {
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 @@ TEST_F(WebRtcSdpTest,
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);