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rtp: aes67: Document sess.ts-direct property and set it to true for AES67
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@ -128,6 +128,11 @@ context.modules = [
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# This property is used if you aren't using ptp4l 4
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sess.ts-refclk = "ptp=traceable"
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sess.ts-offset = 0
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# Directly synchronize output against the PTP-synced driver using the RTP timestamps
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# This can be set to true if the reference clocks are the same; it then makes the
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# synchronization more robust against transport delay variations and can help lower
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# latency
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sess.ts-direct = false
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# You can adjust the latency buffering here. Use integer values only
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sess.latency.msec = 3
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audio.format = "S24BE"
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@ -56,9 +56,19 @@
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* - `sess.latency.msec = <float>`: target network latency in milliseconds, default 100
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* - `sess.ignore-ssrc = <bool>`: ignore SSRC, default false
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* - `sess.media = <string>`: the media type audio|midi|opus, default audio
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* - `sess.ts-direct = <bool>`: directly synchronize output against the current
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* graph driver time, using the RTP timestamps, default false
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* - `stream.may-pause = <bool>`: pause the stream when no data is reveived, default false
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* - `stream.props = {}`: properties to be passed to the stream
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*
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* Set `sess.ts-direct` to true if receivers shall play precisely in sync with the sender even
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* if the transport delay differs. This can be important for use cases like AES67 sessions.
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* The graph driver must then produce time that is in sync with the sender's graph driver.
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* If it is set to false, the RTP timestamps will be used to reproduce the pace of the sender,
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* but not directly for synchronizing when output starts. Note though that this requires that
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* the receivers and senders have synchronized clocks. In PTP, the reference clocks must then
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* be the same. Otherwise, senders and receives will be out of sync.
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*
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* ## General options
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*
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* Options with well-known behavior:
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