Reduce verbosity of streaming

This commit is contained in:
Alexandre Iooss 2020-10-05 11:08:22 +02:00
parent 70296e462e
commit cbe6a00d1a
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GPG Key ID: 6C79278F3FCDCC02
1 changed files with 4 additions and 6 deletions

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@ -2,7 +2,6 @@ package webrtc
import (
"bufio"
"fmt"
"io"
"log"
"net"
@ -43,14 +42,12 @@ func ingestFrom(inputChannel chan srt.Packet) {
}
}()
ffmpeg = exec.Command("ffmpeg", "-re", "-i", "pipe:0",
ffmpeg = exec.Command("ffmpeg", "-hide_banner", "-loglevel", "error", "-re", "-i", "pipe:0",
"-an", "-vcodec", "libvpx", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5004",
"-vn", "-acodec", "libopus", //"-cpu-used", "5", "-deadline", "1", "-g", "10", "-error-resilient", "1", "-auto-alt-ref", "1",
"-f", "rtp", "rtp://127.0.0.1:5005")
fmt.Println("Waiting for RTP Packets, please run GStreamer or ffmpeg now")
input, err := ffmpeg.StdinPipe()
if err != nil {
panic(err)
@ -77,7 +74,6 @@ func ingestFrom(inputChannel chan srt.Packet) {
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
panic(err)
}
log.Printf("[Video] %s", packet)
// Write RTP packet to all video tracks
// Adapt payload and SSRC to match destination
@ -103,7 +99,9 @@ func ingestFrom(inputChannel chan srt.Packet) {
if err := packet.Unmarshal(inboundRTPPacket[:n]); err != nil {
panic(err)
}
log.Printf("[Audio] %s", packet)
// Write RTP packet to all audio tracks
// Adapt payload and SSRC to match destination
for _, audioTrack := range audioTracks {
packet.Header.PayloadType = audioTrack.PayloadType()
packet.Header.SSRC = audioTrack.SSRC()