mirror of
https://github.com/pion/webrtc.git
synced 2025-10-05 07:06:51 +08:00
Upgrade golangci-lint, more linters
Introduces new linters, upgrade golangci-lint to version (v1.63.4)
This commit is contained in:
@@ -31,8 +31,7 @@ const (
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oggPageDuration = time.Millisecond * 20
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)
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// nolint:gocognit
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func main() {
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func main() { //nolint:gocognit,cyclop,gocyclo,maintidx
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// Assert that we have an audio or video file
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_, err := os.Stat(videoFileName)
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haveVideoFile := !os.IsNotExist(err)
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@@ -63,7 +62,7 @@ func main() {
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iceConnectedCtx, iceConnectedCtxCancel := context.WithCancel(context.Background())
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if haveVideoFile {
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if haveVideoFile { //nolint:nestif
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file, openErr := os.Open(videoFileName)
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if openErr != nil {
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panic(openErr)
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@@ -88,7 +87,9 @@ func main() {
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}
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// Create a video track
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videoTrack, videoTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: trackCodec}, "video", "pion")
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videoTrack, videoTrackErr := webrtc.NewTrackLocalStaticSample(
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webrtc.RTPCodecCapability{MimeType: trackCodec}, "video", "pion",
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)
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if videoTrackErr != nil {
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panic(videoTrackErr)
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}
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@@ -131,7 +132,9 @@ func main() {
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// It is important to use a time.Ticker instead of time.Sleep because
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// * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
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// * works around latency issues with Sleep (see https://github.com/golang/go/issues/44343)
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ticker := time.NewTicker(time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000))
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ticker := time.NewTicker(
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time.Millisecond * time.Duration((float32(header.TimebaseNumerator)/float32(header.TimebaseDenominator))*1000),
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)
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defer ticker.Stop()
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for ; true; <-ticker.C {
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frame, _, ivfErr := ivf.ParseNextFrame()
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@@ -151,9 +154,11 @@ func main() {
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}()
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}
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if haveAudioFile {
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if haveAudioFile { //nolint:nestif
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// Create a audio track
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audioTrack, audioTrackErr := webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "pion")
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audioTrack, audioTrackErr := webrtc.NewTrackLocalStaticSample(
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webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "pion",
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)
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if audioTrackErr != nil {
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panic(audioTrackErr)
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}
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@@ -233,18 +238,19 @@ func main() {
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// Set the handler for Peer connection state
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// This will notify you when the peer has connected/disconnected
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peerConnection.OnConnectionStateChange(func(s webrtc.PeerConnectionState) {
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fmt.Printf("Peer Connection State has changed: %s\n", s.String())
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peerConnection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
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fmt.Printf("Peer Connection State has changed: %s\n", state.String())
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if s == webrtc.PeerConnectionStateFailed {
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// Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
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if state == webrtc.PeerConnectionStateFailed {
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// Wait until PeerConnection has had no network activity for 30 seconds or another failure.
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// It may be reconnected using an ICE Restart.
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// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
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// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
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fmt.Println("Peer Connection has gone to failed exiting")
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os.Exit(0)
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}
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if s == webrtc.PeerConnectionStateClosed {
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if state == webrtc.PeerConnectionStateClosed {
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// PeerConnection was explicitly closed. This usually happens from a DTLS CloseNotify
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fmt.Println("Peer Connection has gone to closed exiting")
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os.Exit(0)
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@@ -286,7 +292,7 @@ func main() {
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select {}
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}
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// Read from stdin until we get a newline
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// Read from stdin until we get a newline.
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func readUntilNewline() (in string) {
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var err error
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@@ -303,10 +309,11 @@ func readUntilNewline() (in string) {
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}
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fmt.Println("")
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return
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}
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// JSON encode + base64 a SessionDescription
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// JSON encode + base64 a SessionDescription.
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func encode(obj *webrtc.SessionDescription) string {
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b, err := json.Marshal(obj)
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if err != nil {
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@@ -316,7 +323,7 @@ func encode(obj *webrtc.SessionDescription) string {
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return base64.StdEncoding.EncodeToString(b)
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}
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// Decode a base64 and unmarshal JSON into a SessionDescription
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// Decode a base64 and unmarshal JSON into a SessionDescription.
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func decode(in string, obj *webrtc.SessionDescription) {
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b, err := base64.StdEncoding.DecodeString(in)
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if err != nil {
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