Zac Gaetano
16ae17d2a1
Adds Alloc(), the ephemeral loopback UDP port grabber the subsystem uses to pick the RTP port it will hand to FFmpeg and then re-bind with core/webrtc.NewSourceOn. Covered by a 100x rebind test. Adds BuildArgs(), which emits the -f rtp output fragments (video on the passed port, audio on port+1) with copy codecs by default and an H.264 baseline / libopus re-encode leg when ForceTranscode is set. Covered by three unit tests.
52 lines
1.5 KiB
Go
52 lines
1.5 KiB
Go
package webrtc
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import (
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"fmt"
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appcfg "github.com/datarhei/core/v16/restream/app"
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)
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// BuildArgs emits the FFmpeg output-leg args for the WebRTC side of a
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// process. It produces two separate "outputs" — one for video on
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// videoPort, one for audio on videoPort+1. Each output ends with its
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// UDP address so the slice is structured for consumption by
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// restream.AppendOutput after splitting on the track boundary.
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//
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// Copy vs. re-encode: if ForceTranscode is false, we assume the upstream
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// source is already H.264 + Opus and pass them through (copy). When the
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// source doesn't match, FFmpeg will fail at runtime and the process will
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// restart — the user can flip ForceTranscode on to get a baseline-profile
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// H.264 + Opus re-encode.
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func BuildArgs(cfg appcfg.ConfigWebRTC, videoPort int) []string {
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vcopy := []string{"-c:v", "copy"}
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acopy := []string{"-c:a", "copy"}
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if cfg.ForceTranscode {
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vcopy = []string{
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"-c:v", "libx264",
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"-preset", "veryfast",
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"-profile:v", "baseline",
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"-pix_fmt", "yuv420p",
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"-tune", "zerolatency",
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"-g", "60",
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}
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acopy = []string{"-c:a", "libopus", "-b:a", "96k"}
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}
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args := []string{"-map", "0:v:0"}
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args = append(args, vcopy...)
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args = append(args,
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"-payload_type", fmt.Sprint(cfg.VideoPT),
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"-f", "rtp",
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fmt.Sprintf("udp://127.0.0.1:%d?pkt_size=1316", videoPort),
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)
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args = append(args, "-map", "0:a:0")
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args = append(args, acopy...)
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args = append(args,
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"-payload_type", fmt.Sprint(cfg.AudioPT),
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"-f", "rtp",
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fmt.Sprintf("udp://127.0.0.1:%d?pkt_size=1316", videoPort+1),
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)
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return args
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}
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