RTPCodec as functions.

internal/capture/gst/gst.go

@@ -67,11 +67,11 @@ func CreateJPEGPipeline(pipelineDisplay string, pipelineSrc string, rate string,
 }
 
 // CreateAppPipeline creates a GStreamer Pipeline
-func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
+func CreateAppPipeline(codecRTP codec.RTPCodec, pipelineDevice string, pipelineSrc string) (*Pipeline, error) {
 	var pipelineStr string
 
 	switch codecRTP.Name {
-	case codec.VP8:
+	case "vp8":
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp8enc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// vp8enc error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true cpu-used=5 deadline=1
@@ -80,7 +80,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		pipelineStr = fmt.Sprintf(videoSrc + "vp8enc cpu-used=8 threads=2 deadline=1 error-resilient=partitions keyframe-max-dist=10 auto-alt-ref=true" + appSink, pipelineDevice)
-	case codec.VP9:
+	case "vp9":
 		// https://gstreamer.freedesktop.org/documentation/vpx/vp9enc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// vp9enc
@@ -89,7 +89,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		pipelineStr = fmt.Sprintf(videoSrc + "vp9enc" + appSink, pipelineDevice)
-	case codec.H264:
+	case "h264":
 		if err := CheckPlugins([]string{"ximagesrc"}); err != nil {
 			return nil, err
 		}
@@ -112,7 +112,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		return nil, err
-	case codec.Opus:
+	case "opus":
 		// https://gstreamer.freedesktop.org/documentation/opus/opusenc.html
 		// gstreamer1.0-plugins-base
 		// opusenc
@@ -121,7 +121,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		pipelineStr = fmt.Sprintf(audioSrc + "opusenc" + appSink, pipelineDevice)
-	case codec.G722:
+	case "g722":
 		// https://gstreamer.freedesktop.org/documentation/libav/avenc_g722.html?gi-language=c
 		// gstreamer1.0-libav
 		// avenc_g722
@@ -130,7 +130,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		pipelineStr = fmt.Sprintf(audioSrc + "avenc_g722" + appSink, pipelineDevice)
-	case codec.PCMU:
+	case "pcmu":
 		// https://gstreamer.freedesktop.org/documentation/mulaw/mulawenc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// audio/x-raw, rate=8000 ! mulawenc
@@ -139,7 +139,7 @@ func CreateAppPipeline(codecRTP codec.RTP, pipelineDevice string, pipelineSrc st
 		}
 
 		pipelineStr = fmt.Sprintf(audioSrc + "audio/x-raw, rate=8000 ! mulawenc" + appSink, pipelineDevice)
-	case codec.PCMA:
+	case "pcma":
 		// https://gstreamer.freedesktop.org/documentation/alaw/alawenc.html?gi-language=c
 		// gstreamer1.0-plugins-good
 		// audio/x-raw, rate=8000 ! alawenc

internal/capture/manager.go

@@ -126,11 +126,11 @@ func (manager *CaptureManagerCtx) Screencast() types.ScreencastManager {
 	return manager.screencast
 }
 
-func (manager *CaptureManagerCtx) VideoCodec() codec.RTP {
+func (manager *CaptureManagerCtx) VideoCodec() codec.RTPCodec {
 	return manager.config.VideoCodec
 }
 
-func (manager *CaptureManagerCtx) AudioCodec() codec.RTP {
+func (manager *CaptureManagerCtx) AudioCodec() codec.RTPCodec {
 	return manager.config.AudioCodec
 }
 

internal/config/capture.go

@@ -10,9 +10,9 @@ import (
 type Capture struct {
 	Display      string
 	Device       string
-	AudioCodec   codec.RTP
+	AudioCodec   codec.RTPCodec
 	AudioParams  string
-	VideoCodec   codec.RTP
+	VideoCodec   codec.RTPCodec
 	VideoParams  string
 
 	BroadcastPipeline string
@@ -111,30 +111,30 @@ func (Capture) Init(cmd *cobra.Command) error {
 }
 
 func (s *Capture) Set() {
-	var videoCodec codec.RTP
+	var videoCodec codec.RTPCodec
 	if viper.GetBool("vp8") {
-		videoCodec = codec.New(codec.VP8)
+		videoCodec = codec.VP8()
 	} else if viper.GetBool("vp9") {
-		videoCodec = codec.New(codec.VP9)
+		videoCodec = codec.VP9()
 	} else if viper.GetBool("h264") {
-		videoCodec = codec.New(codec.H264)
+		videoCodec = codec.H264()
 	} else {
 		// default
-		videoCodec = codec.New(codec.VP8)
+		videoCodec = codec.VP8()
 	}
 
-	var audioCodec codec.RTP
+	var audioCodec codec.RTPCodec
 	if viper.GetBool("opus") {
-		audioCodec = codec.New(codec.Opus)
+		audioCodec = codec.Opus()
 	} else if viper.GetBool("g722") {
-		audioCodec = codec.New(codec.G722)
+		audioCodec = codec.G722()
 	} else if viper.GetBool("pcmu") {
-		audioCodec = codec.New(codec.PCMU)
+		audioCodec = codec.PCMU()
 	} else if viper.GetBool("pcma") {
-		audioCodec = codec.New(codec.PCMA)
+		audioCodec = codec.PCMA()
 	} else {
 		// default
-		audioCodec = codec.New(codec.Opus)
+		audioCodec = codec.Opus()
 	}
 
 	s.Device = viper.GetString("device")

internal/types/capture.go

@@ -28,8 +28,8 @@ type CaptureManager interface {
 	Broadcast() BroadcastManager
 	Screencast() ScreencastManager
 
-	VideoCodec() codec.RTP
+	VideoCodec() codec.RTPCodec
-	AudioCodec() codec.RTP
+	AudioCodec() codec.RTPCodec
 
 	OnVideoFrame(listener func(sample Sample))
 	OnAudioFrame(listener func(sample Sample))

internal/types/codec/codecs.go

@@ -2,112 +2,123 @@ package codec
 
 import "github.com/pion/webrtc/v3"
 
-const (
+type RTPCodec struct {
-	VP8 = "vp8"
-	VP9 = "vp9"
-	H264 = "h264"
-	Opus = "opus"
-	G722 = "g722"
-	PCMU = "pcmu"
-	PCMA = "pcma"
-)
-
-type RTP struct {
 	Name        string
 	PayloadType webrtc.PayloadType
 	Type        webrtc.RTPCodecType
 	Capability  webrtc.RTPCodecCapability
 }
 
-func New(codecType string) RTP {
+func (codec *RTPCodec) Register(engine *webrtc.MediaEngine) error {
-	codec := RTP{}
-
-	switch codecType {
-	case "vp8":
-		codec.Name = "vp8"
-		codec.PayloadType = 96
-		codec.Type = webrtc.RTPCodecTypeVideo
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypeVP8,
-			ClockRate: 90000,
-			Channels: 0,
-			SDPFmtpLine: "",
-			RTCPFeedback: nil,
-		}
-	case "vp9":
-		codec.Name = "vp9"
-		codec.PayloadType = 98
-		codec.Type = webrtc.RTPCodecTypeVideo
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypeVP9,
-			ClockRate: 90000,
-			Channels: 0,
-			SDPFmtpLine: "profile-id=0",
-			RTCPFeedback: nil,
-		}
-	case "h264":
-		codec.Name = "h264"
-		codec.PayloadType = 102
-		codec.Type = webrtc.RTPCodecTypeVideo
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypeH264,
-			ClockRate: 90000,
-			Channels: 0,
-			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
-			RTCPFeedback: nil,
-		}
-	case "opus":
-		codec.Name = "opus"
-		codec.PayloadType = 111
-		codec.Type = webrtc.RTPCodecTypeAudio
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypeOpus,
-			ClockRate: 48000,
-			Channels: 2,
-			SDPFmtpLine: "",
-			RTCPFeedback: nil,
-		}
-	case "g722":
-		codec.Name = "g722"
-		codec.PayloadType = 9
-		codec.Type = webrtc.RTPCodecTypeAudio
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypeG722,
-			ClockRate: 8000,
-			Channels: 0,
-			SDPFmtpLine: "",
-			RTCPFeedback: nil,
-		}
-	case "pcmu":
-		codec.Name = "pcmu"
-		codec.PayloadType = 0
-		codec.Type = webrtc.RTPCodecTypeAudio
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypePCMU,
-			ClockRate: 8000,
-			Channels: 0,
-			SDPFmtpLine: "",
-			RTCPFeedback: nil,
-		}
-	case "pcma":
-		codec.Name = "pcma"
-		codec.PayloadType = 8
-		codec.Type = webrtc.RTPCodecTypeAudio
-		codec.Capability = webrtc.RTPCodecCapability{
-			MimeType: webrtc.MimeTypePCMA,
-			ClockRate: 8000,
-			Channels: 0,
-			SDPFmtpLine: "",
-			RTCPFeedback: nil,
-		}
-	}
-
-	return codec
-}
-
-func (codec *RTP) Register(engine *webrtc.MediaEngine) error {
 	return engine.RegisterCodec(webrtc.RTPCodecParameters{
 		RTPCodecCapability: codec.Capability,
 		PayloadType:        codec.PayloadType,
 	}, codec.Type)
 }
+
+func VP8() RTPCodec {
+	return RTPCodec{
+		Name: "vp8",
+		PayloadType: 96,
+		Type: webrtc.RTPCodecTypeVideo,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeVP8,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+// TODO: Profile ID.
+func VP9() RTPCodec {
+	return RTPCodec{
+		Name: "vp9",
+		PayloadType: 98,
+		Type: webrtc.RTPCodecTypeVideo,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeVP9,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "profile-id=0",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+// TODO: Profile ID.
+func H264() RTPCodec {
+	return RTPCodec{
+		Name: "h264",
+		PayloadType: 102,
+		Type: webrtc.RTPCodecTypeVideo,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeH264,
+			ClockRate: 90000,
+			Channels: 0,
+			SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+func Opus() RTPCodec {
+	return RTPCodec{
+		Name: "opus",
+		PayloadType: 111,
+		Type: webrtc.RTPCodecTypeAudio,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeOpus,
+			ClockRate: 48000,
+			Channels: 2,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+func G722() RTPCodec {
+	return RTPCodec{
+		Name: "g722",
+		PayloadType: 9,
+		Type: webrtc.RTPCodecTypeAudio,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypeG722,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+func PCMU() RTPCodec {
+	return RTPCodec{
+		Name: "pcmu",
+		PayloadType: 0,
+		Type: webrtc.RTPCodecTypeAudio,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypePCMU,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		},
+	}
+}
+
+func PCMA() RTPCodec {
+	return RTPCodec{
+		Name: "pcma",
+		PayloadType: 8,
+		Type: webrtc.RTPCodecTypeAudio,
+		Capability: webrtc.RTPCodecCapability{
+			MimeType: webrtc.MimeTypePCMA,
+			ClockRate: 8000,
+			Channels: 0,
+			SDPFmtpLine: "",
+			RTCPFeedback: nil,
+		},
+	}
+}

internal/webrtc/manager.go

@@ -28,8 +28,8 @@ type WebRTCManagerCtx struct {
 	logger     zerolog.Logger
 	videoTrack *webrtc.TrackLocalStaticSample
 	audioTrack *webrtc.TrackLocalStaticSample
-	videoCodec codec.RTP
+	videoCodec codec.RTPCodec
-	audioCodec codec.RTP
+	audioCodec codec.RTPCodec
 	desktop    types.DesktopManager
 	capture    types.CaptureManager
 	config     *config.WebRTC