Remove RTC prefix from all names

Let's pull off the bandaid! * Reduces studdering: webrtc.RTCTrack -> webrtc.Track * Makes it easier to find types by editor autocomplete * Makes code read more fluently (less repetition) Since we're breaking the API in 2.0, our only chance to do this is now. Relates to #408
pion · Feb 18, 2019 · 0e7086d · 0e7086d
1 parent b711ec2
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diff --git a/README.md b/README.md
@@ -74,7 +74,7 @@ Check out the **[contributing wiki](https://github.com/pions/webrtc/wiki/Contrib
 * [Tobias Fridén](https://github.com/tobiasfriden) *SRTP authentication verification*
 * [Yutaka Takeda](https://github.com/enobufs) *Fix ICE connection timeout*
 * [Hugo Arregui](https://github.com/hugoArregui) *Fix connection timeout*
-* [Rob Deutsch](https://github.com/rob-deutsch) *RTCRtpReceiver graceful shutdown*
+* [Rob Deutsch](https://github.com/rob-deutsch) *RTPReceiver graceful shutdown*
 * [Jin Lei](https://github.com/jinleileiking) - *SFU example use http*
 
 ### License

diff --git a/api.go b/api.go
@@ -77,7 +77,7 @@ func SetConnectionTimeout(connectionTimeout, keepAlive time.Duration) {
 
 // RegisterCodec on the default API.
 // See MediaEngine for details.
-func RegisterCodec(codec *RTCRtpCodec) {
+func RegisterCodec(codec *RTPCodec) {
 	defaultAPI.mediaEngine.RegisterCodec(codec)
 }
 
@@ -89,8 +89,8 @@ func RegisterDefaultCodecs() {
 
 // PeerConnection API
 
-// NewRTCPeerConnection using the default API.
+// NewPeerConnection using the default API.
 // See API.NewRTCPeerConnection for details.
-func NewRTCPeerConnection(configuration RTCConfiguration) (*RTCPeerConnection, error) {
-	return defaultAPI.NewRTCPeerConnection(configuration)
+func NewPeerConnection(configuration Configuration) (*PeerConnection, error) {
+	return defaultAPI.NewPeerConnection(configuration)
 }
diff --git a/errors.go b/errors.go
@@ -35,24 +35,24 @@ var (
 	ErrPrivateKeyType = errors.New("private key type not supported")
 
 	// ErrModifyingPeerIdentity indicates that an attempt to modify
-	// PeerIdentity was made after RTCPeerConnection has been initialized.
+	// PeerIdentity was made after PeerConnection has been initialized.
 	ErrModifyingPeerIdentity = errors.New("peerIdentity cannot be modified")
 
 	// ErrModifyingCertificates indicates that an attempt to modify
-	// Certificates was made after RTCPeerConnection has been initialized.
+	// Certificates was made after PeerConnection has been initialized.
 	ErrModifyingCertificates = errors.New("certificates cannot be modified")
 
 	// ErrModifyingBundlePolicy indicates that an attempt to modify
-	// BundlePolicy was made after RTCPeerConnection has been initialized.
+	// BundlePolicy was made after PeerConnection has been initialized.
 	ErrModifyingBundlePolicy = errors.New("bundle policy cannot be modified")
 
-	// ErrModifyingRtcpMuxPolicy indicates that an attempt to modify
-	// RtcpMuxPolicy was made after RTCPeerConnection has been initialized.
-	ErrModifyingRtcpMuxPolicy = errors.New("rtcp mux policy cannot be modified")
+	// ErrModifyingRTCPMuxPolicy indicates that an attempt to modify
+	// RTCPMuxPolicy was made after PeerConnection has been initialized.
+	ErrModifyingRTCPMuxPolicy = errors.New("rtcp mux policy cannot be modified")
 
-	// ErrModifyingIceCandidatePoolSize indicates that an attempt to modify
-	// IceCandidatePoolSize was made after RTCPeerConnection has been initialized.
-	ErrModifyingIceCandidatePoolSize = errors.New("ice candidate pool size cannot be modified")
+	// ErrModifyingICECandidatePoolSize indicates that an attempt to modify
+	// ICECandidatePoolSize was made after PeerConnection has been initialized.
+	ErrModifyingICECandidatePoolSize = errors.New("ice candidate pool size cannot be modified")
 
 	// ErrStringSizeLimit indicates that the character size limit of string is
 	// exceeded. The limit is hardcoded to 65535 according to specifications.

diff --git a/examples/data-channels-close/main.go b/examples/data-channels-close/main.go
@@ -18,16 +18,16 @@ func main() {
 	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set the handler for ICE connection state
@@ -37,7 +37,7 @@ func main() {
 	})
 
 	// Register data channel creation handling
-	peerConnection.OnDataChannel(func(d *webrtc.RTCDataChannel) {
+	peerConnection.OnDataChannel(func(d *webrtc.DataChannel) {
 		fmt.Printf("New DataChannel %s %d\n", d.Label, d.ID)
 
 		// Register channel opening handling
@@ -83,7 +83,7 @@ func main() {
 	})
 
 	// Wait for the offer to be pasted
-	offer := webrtc.RTCSessionDescription{}
+	offer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &offer)
 
 	// Set the remote SessionDescription

diff --git a/examples/data-channels-create/main.go b/examples/data-channels-create/main.go
@@ -14,16 +14,16 @@ func main() {
 	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Create a datachannel with label 'data'
@@ -73,7 +73,7 @@ func main() {
 	fmt.Println(util.Encode(offer))
 
 	// Wait for the answer to be pasted
-	answer := webrtc.RTCSessionDescription{}
+	answer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &answer)
 
 	// Apply the answer as the remote description

diff --git a/examples/data-channels-detach-create/main.go b/examples/data-channels-detach-create/main.go
@@ -21,16 +21,16 @@ func main() {
 	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Create a datachannel with label 'data'
@@ -70,7 +70,7 @@ func main() {
 	fmt.Println(util.Encode(offer))
 
 	// Wait for the answer to be pasted
-	answer := webrtc.RTCSessionDescription{}
+	answer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), answer)
 
 	// Apply the answer as the remote description

diff --git a/examples/data-channels-detach/main.go b/examples/data-channels-detach/main.go
@@ -21,16 +21,16 @@ func main() {
 	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set the handler for ICE connection state
@@ -40,7 +40,7 @@ func main() {
 	})
 
 	// Register data channel creation handling
-	peerConnection.OnDataChannel(func(d *webrtc.RTCDataChannel) {
+	peerConnection.OnDataChannel(func(d *webrtc.DataChannel) {
 		fmt.Printf("New DataChannel %s %d\n", d.Label, d.ID)
 
 		// Register channel opening handling
@@ -60,7 +60,7 @@ func main() {
 	})
 
 	// Wait for the offer to be pasted
-	offer := webrtc.RTCSessionDescription{}
+	offer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), offer)
 
 	// Set the remote SessionDescription

diff --git a/examples/data-channels/main.go b/examples/data-channels/main.go
@@ -14,16 +14,16 @@ func main() {
 	// Everything below is the pion-WebRTC API! Thanks for using it ❤️.
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set the handler for ICE connection state
@@ -33,7 +33,7 @@ func main() {
 	})
 
 	// Register data channel creation handling
-	peerConnection.OnDataChannel(func(d *webrtc.RTCDataChannel) {
+	peerConnection.OnDataChannel(func(d *webrtc.DataChannel) {
 		fmt.Printf("New DataChannel %s %d\n", d.Label, d.ID)
 
 		// Register channel opening handling
@@ -63,7 +63,7 @@ func main() {
 	})
 
 	// Wait for the offer to be pasted
-	offer := webrtc.RTCSessionDescription{}
+	offer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &offer)
 
 	// Set the remote SessionDescription

diff --git a/examples/gstreamer-receive/main.go b/examples/gstreamer-receive/main.go
@@ -22,21 +22,21 @@ func gstreamerReceiveMain() {
 	webrtc.RegisterDefaultCodecs()
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set a handler for when a new remote track starts, this handler creates a gstreamer pipeline
 	// for the given codec
-	peerConnection.OnTrack(func(track *webrtc.RTCTrack) {
+	peerConnection.OnTrack(func(track *webrtc.Track) {
 		// Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval
 		// This is a temporary fix until we implement incoming RTCP events, then we would push a PLI only when a viewer requests it
 		go func() {
@@ -66,7 +66,7 @@ func gstreamerReceiveMain() {
 	})
 
 	// Wait for the offer to be pasted
-	offer := webrtc.RTCSessionDescription{}
+	offer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &offer)
 
 	// Set the remote SessionDescription

diff --git a/examples/gstreamer-send-offer/main.go b/examples/gstreamer-send-offer/main.go
@@ -17,16 +17,16 @@ func main() {
 	webrtc.RegisterDefaultCodecs()
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set the handler for ICE connection state
@@ -36,13 +36,13 @@ func main() {
 	})
 
 	// Create a audio track
-	opusTrack, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeOpus, "audio", "pion1")
+	opusTrack, err := peerConnection.NewSampleTrack(webrtc.DefaultPayloadTypeOpus, "audio", "pion1")
 	util.Check(err)
 	_, err = peerConnection.AddTrack(opusTrack)
 	util.Check(err)
 
 	// Create a video track
-	vp8Track, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeVP8, "video", "pion2")
+	vp8Track, err := peerConnection.NewSampleTrack(webrtc.DefaultPayloadTypeVP8, "video", "pion2")
 	util.Check(err)
 	_, err = peerConnection.AddTrack(vp8Track)
 	util.Check(err)
@@ -59,7 +59,7 @@ func main() {
 	fmt.Println(util.Encode(offer))
 
 	// Wait for the answer to be pasted
-	answer := webrtc.RTCSessionDescription{}
+	answer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &answer)
 
 	// Set the remote SessionDescription

diff --git a/examples/gstreamer-send/main.go b/examples/gstreamer-send/main.go
@@ -22,16 +22,16 @@ func main() {
 	webrtc.RegisterDefaultCodecs()
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	// Set the handler for ICE connection state
@@ -41,19 +41,19 @@ func main() {
 	})
 
 	// Create a audio track
-	opusTrack, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeOpus, "audio", "pion1")
+	opusTrack, err := peerConnection.NewSampleTrack(webrtc.DefaultPayloadTypeOpus, "audio", "pion1")
 	util.Check(err)
 	_, err = peerConnection.AddTrack(opusTrack)
 	util.Check(err)
 
 	// Create a video track
-	vp8Track, err := peerConnection.NewRTCSampleTrack(webrtc.DefaultPayloadTypeVP8, "video", "pion2")
+	vp8Track, err := peerConnection.NewSampleTrack(webrtc.DefaultPayloadTypeVP8, "video", "pion2")
 	util.Check(err)
 	_, err = peerConnection.AddTrack(vp8Track)
 	util.Check(err)
 
 	// Wait for the offer to be pasted
-	offer := webrtc.RTCSessionDescription{}
+	offer := webrtc.SessionDescription{}
 	util.Decode(util.MustReadStdin(), &offer)
 
 	// Set the remote SessionDescription

diff --git a/examples/janus-gateway/streaming/main.go b/examples/janus-gateway/streaming/main.go
@@ -40,23 +40,23 @@ func main() {
 	webrtc.RegisterDefaultCodecs()
 
 	// Prepare the configuration
-	config := webrtc.RTCConfiguration{
-		IceServers: []webrtc.RTCIceServer{
+	config := webrtc.Configuration{
+		ICEServers: []webrtc.ICEServer{
 			{
 				URLs: []string{"stun:stun.l.google.com:19302"},
 			},
 		},
 	}
 
 	// Create a new RTCPeerConnection
-	peerConnection, err := webrtc.NewRTCPeerConnection(config)
+	peerConnection, err := webrtc.NewPeerConnection(config)
 	util.Check(err)
 
 	peerConnection.OnICEConnectionStateChange(func(connectionState ice.ConnectionState) {
 		fmt.Printf("Connection State has changed %s \n", connectionState.String())
 	})
 
-	peerConnection.OnTrack(func(track *webrtc.RTCTrack) {
+	peerConnection.OnTrack(func(track *webrtc.Track) {
 		if track.Codec.Name == webrtc.Opus {
 			return
 		}
@@ -98,8 +98,8 @@ func main() {
 	util.Check(err)
 
 	if msg.Jsep != nil {
-		err = peerConnection.SetRemoteDescription(webrtc.RTCSessionDescription{
-			Type: webrtc.RTCSdpTypeOffer,
+		err = peerConnection.SetRemoteDescription(webrtc.SessionDescription{
+			Type: webrtc.SDPTypeOffer,
 			Sdp:  msg.Jsep["sdp"].(string),
 		})
 		util.Check(err)