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Fix typos in multiple comments
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@@ -6,7 +6,7 @@
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package webrtc
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// API bundles the global funcions of the WebRTC and ORTC API.
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// API bundles the global functions of the WebRTC and ORTC API.
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type API struct {
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settingEngine *SettingEngine
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}
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@@ -236,7 +236,7 @@ func (d *DataChannel) onOpen() {
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}
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// OnDial sets an event handler which is invoked when the
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// peer has been dialed, but before said peer has responsed
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// peer has been dialed, but before said peer has responded
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func (d *DataChannel) OnDial(f func()) {
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d.mu.Lock()
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d.dialHandlerOnce = sync.Once{}
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@@ -115,7 +115,7 @@ func (d *DataChannel) SendText(s string) (err error) {
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// Before calling Detach you have to enable this behavior by calling
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// webrtc.DetachDataChannels(). Combining detached and normal data channels
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// is not supported.
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// Please reffer to the data-channels-detach example and the
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// Please refer to the data-channels-detach example and the
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// pion/datachannel documentation for the correct way to handle the
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// resulting DataChannel object.
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func (d *DataChannel) Detach() (datachannel.ReadWriteCloser, error) {
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@@ -4,7 +4,7 @@ This example modifies the video with a single-byte XOR cipher before sending, an
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decrypts in Javascript.
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insertable-streams allows the browser to process encoded video. You could implement
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E2E encyption, add metadata or insert a completely different video feed!
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E2E encryption, add metadata or insert a completely different video feed!
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## Instructions
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### Create IVF named `output.ivf` that contains a VP8 track
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@@ -127,7 +127,7 @@ func ConfigureCongestionControlFeedback(mediaEngine *MediaEngine, interceptorReg
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return nil
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}
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// ConfigureSimulcastExtensionHeaders enables the RTP Extenison Headers needed for Simulcast
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// ConfigureSimulcastExtensionHeaders enables the RTP Extension Headers needed for Simulcast
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func ConfigureSimulcastExtensionHeaders(mediaEngine *MediaEngine) error {
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if err := mediaEngine.RegisterHeaderExtension(RTPHeaderExtensionCapability{URI: sdp.SDESMidURI}, RTPCodecTypeVideo); err != nil {
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return err
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@@ -18,12 +18,12 @@ const (
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// Use global random generator to properly seed by crypto grade random.
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var globalMathRandomGenerator = randutil.NewMathRandomGenerator() // nolint:gochecknoglobals
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// MathRandAlpha generates a mathmatical random alphabet sequence of the requested length.
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// MathRandAlpha generates a mathematical random alphabet sequence of the requested length.
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func MathRandAlpha(n int) string {
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return globalMathRandomGenerator.GenerateString(n, runesAlpha)
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}
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// RandUint32 generates a mathmatical random uint32.
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// RandUint32 generates a mathematical random uint32.
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func RandUint32() uint32 {
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return globalMathRandomGenerator.Uint32()
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}
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@@ -1326,7 +1326,7 @@ func runIfNewReceiver(
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return false
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}
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// configurepRTPReceivers opens knows inbound SRTP streams from the RemoteDescription
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// configureRTPReceivers opens knows inbound SRTP streams from the RemoteDescription
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func (pc *PeerConnection) configureRTPReceivers(isRenegotiation bool, remoteDesc *SessionDescription, currentTransceivers []*RTPTransceiver) { //nolint:gocognit
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incomingTracks := trackDetailsFromSDP(pc.log, remoteDesc.parsed)
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@@ -1606,7 +1606,7 @@ func TestPeerConnection_Simulcast_RTX(t *testing.T) {
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assert.Greater(t, rtxPacketRead.Load(), int32(0), "no rtx packet read")
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}
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// Everytime we receieve a new SSRC we probe it and try to determine the proper way to handle it.
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// Everytime we receive a new SSRC we probe it and try to determine the proper way to handle it.
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// In most cases a Track explicitly declares a SSRC and a OnTrack is fired. In two cases we don't
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// know the SSRC ahead of time
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// * Undeclared SSRC in a single media section (https://github.com/pion/webrtc/issues/880)
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2
sdp.go
2
sdp.go
@@ -85,7 +85,7 @@ func trackDetailsFromSDP(log logging.LeveledLogger, s *sdp.SessionDescription) (
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tracksInMediaSection := []trackDetails{}
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rtxRepairFlows := map[uint64]uint64{}
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// Plan B can have multiple tracks in a signle media section
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// Plan B can have multiple tracks in a single media section
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streamID := ""
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trackID := ""
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