Implement RTP extension support in WebRtcVideoEngine2.
BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index d6d1354..716c5a8 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -278,6 +278,13 @@
video_codecs_ = DefaultVideoCodecs();
default_codec_format_ = VideoFormat(kDefaultVideoFormat);
+
+ rtp_header_extensions_.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
+ kRtpTimestampOffsetHeaderExtensionDefaultId));
+ rtp_header_extensions_.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+ kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@@ -774,6 +781,20 @@
return true;
}
+static std::string RtpExtensionsToString(
+ const std::vector<RtpHeaderExtension>& extensions) {
+ std::stringstream out;
+ out << '{';
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
+ if (i != extensions.size() - 1) {
+ out << ", ";
+ }
+ }
+ out << '}';
+ return out.str();
+}
+
} // namespace
bool WebRtcVideoChannel2::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
@@ -967,6 +988,8 @@
config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
}
+ config.rtp.extensions = send_rtp_extensions_;
+
if (IsNackEnabled(codec_settings.codec)) {
config.rtp.nack.rtp_history_ms = kNackHistoryMs;
}
@@ -1047,6 +1070,7 @@
config.rtp.nack.rtp_history_ms = kNackHistoryMs;
}
config.rtp.remb = true;
+ config.rtp.extensions = recv_rtp_extensions_;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
// corresponding sender SSRC. Figure out what to do when we don't have
@@ -1280,15 +1304,31 @@
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
- // TODO(pbos): Implement.
- LOG(LS_VERBOSE) << "SetRecvRtpHeaderExtensions()";
+ LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
+ << RtpExtensionsToString(extensions);
+ std::vector<webrtc::RtpExtension> webrtc_extensions;
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ // TODO(pbos): Make sure we don't pass unsupported extensions!
+ webrtc::RtpExtension webrtc_extension(extensions[i].uri.c_str(),
+ extensions[i].id);
+ webrtc_extensions.push_back(webrtc_extension);
+ }
+ recv_rtp_extensions_ = webrtc_extensions;
return true;
}
bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
- // TODO(pbos): Implement.
- LOG(LS_VERBOSE) << "SetSendRtpHeaderExtensions()";
+ LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
+ << RtpExtensionsToString(extensions);
+ std::vector<webrtc::RtpExtension> webrtc_extensions;
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ // TODO(pbos): Make sure we don't pass unsupported extensions!
+ webrtc::RtpExtension webrtc_extension(extensions[i].uri.c_str(),
+ extensions[i].id);
+ webrtc_extensions.push_back(webrtc_extension);
+ }
+ send_rtp_extensions_ = webrtc_extensions;
return true;
}
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index d1a784d..81466eb 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -236,6 +236,9 @@
OVERRIDE;
virtual void OnReadyToSend(bool ready) OVERRIDE;
virtual bool MuteStream(uint32 ssrc, bool mute) OVERRIDE;
+
+ // Set send/receive RTP header extensions. This must be done before creating
+ // streams as it only has effect on future streams.
virtual bool SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) OVERRIDE;
virtual bool SetSendRtpHeaderExtensions(
@@ -351,8 +354,11 @@
std::map<uint32, webrtc::VideoReceiveStream*> receive_streams_;
Settable<VideoCodecSettings> send_codec_;
+ std::vector<webrtc::RtpExtension> send_rtp_extensions_;
+
WebRtcVideoEncoderFactory2* const encoder_factory_;
std::vector<VideoCodecSettings> recv_codecs_;
+ std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
VideoOptions options_;
};
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index c9ff182..6886300 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -396,6 +396,31 @@
FAIL() << "No RTX codec found among default codecs.";
}
+TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
+ std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
+ ASSERT_FALSE(extensions.empty());
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ if (extensions[i].uri == kRtpTimestampOffsetHeaderExtension) {
+ EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extensions[i].id);
+ return;
+ }
+ }
+ FAIL() << "Timestamp offset extension not in header-extension list.";
+}
+
+TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
+ std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
+ ASSERT_FALSE(extensions.empty());
+ for (size_t i = 0; i < extensions.size(); ++i) {
+ if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
+ EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
+ extensions[i].id);
+ return;
+ }
+ }
+ FAIL() << "Absolute Sender Time extension not in header-extension list.";
+}
+
class WebRtcVideoChannel2BaseTest
: public VideoMediaChannelTest<WebRtcVideoEngine2, WebRtcVideoChannel2> {
protected:
@@ -598,6 +623,67 @@
EXPECT_EQ(video_codec.height, webrtc_codec.height);
EXPECT_EQ(video_codec.framerate, webrtc_codec.maxFramerate);
}
+
+ void TestSetSendRtpHeaderExtensions(const std::string& cricket_ext,
+ const std::string& webrtc_ext) {
+ // Enable extension.
+ const int id = 1;
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ extensions.push_back(cricket::RtpHeaderExtension(cricket_ext, id));
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+
+ FakeVideoSendStream* send_stream =
+ AddSendStream(cricket::StreamParams::CreateLegacy(123));
+
+ // Verify the send extension id.
+ ASSERT_EQ(1u, send_stream->GetConfig().rtp.extensions.size());
+ EXPECT_EQ(id, send_stream->GetConfig().rtp.extensions[0].id);
+ EXPECT_EQ(webrtc_ext, send_stream->GetConfig().rtp.extensions[0].name);
+ // Verify call with same set of extensions returns true.
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(extensions));
+ // Verify that SetSendRtpHeaderExtensions doesn't implicitly add them for
+ // receivers.
+ EXPECT_TRUE(AddRecvStream(cricket::StreamParams::CreateLegacy(123))
+ ->GetConfig()
+ .rtp.extensions.empty());
+
+ // Remove the extension id, verify that this doesn't reset extensions as
+ // they should be set before creating channels.
+ std::vector<cricket::RtpHeaderExtension> empty_extensions;
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
+ EXPECT_FALSE(send_stream->GetConfig().rtp.extensions.empty());
+ }
+
+ void TestSetRecvRtpHeaderExtensions(const std::string& cricket_ext,
+ const std::string& webrtc_ext) {
+ // Enable extension.
+ const int id = 1;
+ std::vector<cricket::RtpHeaderExtension> extensions;
+ extensions.push_back(cricket::RtpHeaderExtension(cricket_ext, id));
+ EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+
+ FakeVideoReceiveStream* recv_stream =
+ AddRecvStream(cricket::StreamParams::CreateLegacy(123));
+
+ // Verify the recv extension id.
+ ASSERT_EQ(1u, recv_stream->GetConfig().rtp.extensions.size());
+ EXPECT_EQ(id, recv_stream->GetConfig().rtp.extensions[0].id);
+ EXPECT_EQ(webrtc_ext, recv_stream->GetConfig().rtp.extensions[0].name);
+ // Verify call with same set of extensions returns true.
+ EXPECT_TRUE(channel_->SetRecvRtpHeaderExtensions(extensions));
+ // Verify that SetRecvRtpHeaderExtensions doesn't implicitly add them for
+ // senders.
+ EXPECT_TRUE(AddSendStream(cricket::StreamParams::CreateLegacy(123))
+ ->GetConfig()
+ .rtp.extensions.empty());
+
+ // Remove the extension id, verify that this doesn't reset extensions as
+ // they should be set before creating channels.
+ std::vector<cricket::RtpHeaderExtension> empty_extensions;
+ EXPECT_TRUE(channel_->SetSendRtpHeaderExtensions(empty_extensions));
+ EXPECT_FALSE(recv_stream->GetConfig().rtp.extensions.empty());
+ }
+
talk_base::scoped_ptr<VideoMediaChannel> channel_;
FakeWebRtcVideoChannel2* fake_channel_;
uint32 last_ssrc_;
@@ -723,12 +809,34 @@
ASSERT_TRUE(recv_stream->GetConfig().rtp.rtx.empty());
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_RtpTimestampOffsetHeaderExtensions) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+TEST_F(WebRtcVideoChannel2Test, NoHeaderExtesionsByDefault) {
+ FakeVideoSendStream* send_stream =
+ AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
+ ASSERT_TRUE(send_stream->GetConfig().rtp.extensions.empty());
+
+ FakeVideoReceiveStream* recv_stream =
+ AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrcs1[0]));
+ ASSERT_TRUE(recv_stream->GetConfig().rtp.extensions.empty());
}
-TEST_F(WebRtcVideoChannel2Test, DISABLED_AbsoluteSendTimeHeaderExtensions) {
- FAIL() << "Not implemented."; // TODO(pbos): Implement.
+// Test support for RTP timestamp offset header extension.
+TEST_F(WebRtcVideoChannel2Test, SendRtpTimestampOffsetHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension,
+ webrtc::RtpExtension::kTOffset);
+}
+TEST_F(WebRtcVideoChannel2Test, RecvRtpTimestampOffsetHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpTimestampOffsetHeaderExtension,
+ webrtc::RtpExtension::kTOffset);
+}
+
+// Test support for absolute send time header extension.
+TEST_F(WebRtcVideoChannel2Test, SendAbsoluteSendTimeHeaderExtensions) {
+ TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
+ webrtc::RtpExtension::kAbsSendTime);
+}
+TEST_F(WebRtcVideoChannel2Test, RecvAbsoluteSendTimeHeaderExtensions) {
+ TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
+ webrtc::RtpExtension::kAbsSendTime);
}
TEST_F(WebRtcVideoChannel2Test, DISABLED_LeakyBucketTest) {