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https://github.com/aler9/rtsp-simple-server
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move constant (#5190)
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@@ -27,6 +27,15 @@ const (
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webrtcPayloadMaxSize = 1188 // 1200 - 12 (RTP header)
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)
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var multichannelOpusSDP = map[int]string{
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3: "channel_mapping=0,2,1;num_streams=2;coupled_streams=1",
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4: "channel_mapping=0,1,2,3;num_streams=2;coupled_streams=2",
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5: "channel_mapping=0,4,1,2,3;num_streams=3;coupled_streams=2",
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6: "channel_mapping=0,4,1,2,3,5;num_streams=4;coupled_streams=2",
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7: "channel_mapping=0,4,1,2,3,5,6;num_streams=4;coupled_streams=4",
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8: "channel_mapping=0,6,1,4,5,2,3,7;num_streams=5;coupled_streams=4",
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}
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var errNoSupportedCodecsFrom = errors.New(
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"the stream doesn't contain any supported codec, which are currently " +
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"AV1, VP9, VP8, H265, H264, Opus, G722, G711, LPCM")
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@@ -11,15 +11,6 @@ import (
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"github.com/pion/webrtc/v4"
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)
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var multichannelOpusSDP = map[int]string{
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3: "channel_mapping=0,2,1;num_streams=2;coupled_streams=1",
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4: "channel_mapping=0,1,2,3;num_streams=2;coupled_streams=2",
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5: "channel_mapping=0,4,1,2,3;num_streams=3;coupled_streams=2",
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6: "channel_mapping=0,4,1,2,3,5;num_streams=4;coupled_streams=2",
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7: "channel_mapping=0,4,1,2,3,5,6;num_streams=4;coupled_streams=4",
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8: "channel_mapping=0,6,1,4,5,2,3,7;num_streams=5;coupled_streams=4",
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}
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// OutgoingTrack is a WebRTC outgoing track
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type OutgoingTrack struct {
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Caps webrtc.RTPCodecCapability
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