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Unbescholtener Bürger back with another FFmpeg question. My goal is to have an instance of FFmpeg running that receives a mpeg transport stream via RTP, executes arbitrary operations on the received stream like transcoding or filtering, then passes the altered transport stream on via RTP.
However, I don't get it to work in even the most basic configuration: I have a .ts file that contains a single program consisting of one video and one audio stream. I use an instance of FFmpeg to stream this file to localhost:
ffmpeg -re -i 1.ts -c copy -f rtp_mpegts rtp://127.0.0.1:5003
With ffplay, I validate that this works correctly:
ffplay -i rtp://127.0.0.1:5003
Result looks good, sounds good, doesn't cause error messages on the console.
Now I try to put in another FFmpeg instance in between and things go downhill. I just want FFmpeg to copy the stream from port 5003 to port 5005 without any further processing or transcoding:
ffmpeg -re -probesize 50M -analyzeduration 50M -i rtp://127.0.0.1:5003?fifo_size=10000 -c copy -f rtp_mpegts rtp://127.0.0.1:5005
I get a lot of recurring error messages on that console, and when I monitor the stream at port 5005 with ffplay, lots of artifacts, glitches and dropouts appear. The console output looks like this for the most part:
...
Input #0, rtp, from 'rtp://127.0.0.1:5003?fifo_size=10000':
Duration: N/A, start: 1.400022, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(progressive), 352x240 [SAR 1:1 DAR 22:15], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1: Audio: ac3 ([129][0][0][0] / 0x0081), 44100 Hz, stereo, fltp, 192 kb/s
Output #0, rtp_mpegts, to 'rtp://127.0.0.1:5005':
Metadata:
encoder : Lavf58.2.102
Stream #0:0: Video: h264 (Constrained Baseline) ([27][0][0][0] / 0x001B), yuv420p(progressive), 352x240 [SAR 1:1 DAR 22:15], q=2-31, 29.97 fps, 29.97 tbr, 90k tbn, 29.97 tbc
Stream #0:1: Audio: ac3 ([129][0][0][0] / 0x0081), 44100 Hz, stereo, fltp, 192 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=-1.0 size= 0kB time=00:00:00.48 bitrate= 0.0kbits/s speed=0.957x
frame= 0 fps=0.0 q=-1.0 size= 0kB time=00:00:01.01 bitrate= 0.0kbits/s speed=0.991x
frame= 0 fps=0.0 q=-1.0 size= 0kB time=00:00:01.53 bitrate= 0.0kbits/s speed=1.01x
frame= 1 fps=0.5 q=-1.0 size= 57kB time=00:00:02.05 bitrate= 227.6kbits/s speed=1.01x
frame= 17 fps=6.7 q=-1.0 size= 83kB time=00:00:02.54 bitrate= 267.6kbits/s speed= 1x
frame= 32 fps= 11 q=-1.0 size= 121kB time=00:00:03.04 bitrate= 324.9kbits/s speed= 1x
[rtp @ 0x3db64c0] max delay reached. need to consume packet
[rtp @ 0x3db64c0] RTP: missed 50 packets
[rtp @ 0x3db64c0] PES packet size mismatch
frame= 37 fps= 10 q=-1.0 size= 132kB time=00:00:04.07 bitrate= 266.0kbits/s speed=1.15x frame= 37 fps=9.1 q=-1.0 size= 132kB time=00:00:04.07 bitrate= 266.0kbits/s speed=1.01x
[rtp @ 0x3db64c0] max delay reached. need to consume packet
[rtp @ 0x3db64c0] RTP: missed 1 packets
[rtp @ 0x3db64c0] PES packet size mismatch
[rtp @ 0x3db64c0] max delay reached. need to consume packet
[rtp @ 0x3db64c0] RTP: missed 1 packets
frame= 47 fps= 10 q=-1.0 size= 150kB time=00:00:04.57 bitrate= 269.3kbits/s speed=1.01x
...
So... any idea what went wrong and how to fix it? A glance at the resource monitor doesn't indicate any high cpu, memory or network load.
Can you provide an example .ts file? – None – 2018-01-05T13:58:01.917
I could, but it wouldn't be legal I think. I downloaded Bronski Beat - Smalltown Boy from YT and used FFmpeg to convert it to .ts. Anyways, that doesn't matter because I actually found the culprit. – UnbescholtenerBuerger – 2018-01-05T14:27:09.047