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Use case

The primary goal is to decode a large format uncompressed video signal (SMPTE ST 2110-20) along with raw audio (SMPTE ST 2110-30); the stream synchronization being based on PTP (SMPTE ST 2110-10). After the AV content is reconstructed, ffmpeg re-encodes the signal for storage or streaming. The project also aims at evaluating the limitation in terms of bandwidth, especially when additional streams are provided.

+-----------------------+            +-------------------+              +------------+
| Source:               |            | Transcoder:       |              | Monitor:   |
+-----------------------+            +-------------------+              +------------+
| exple: Gstreamer or HW|            | ffmpeg            |              | vlc, ffplay|
+-----------------------+            +-------------------+              +------------+
| generate rtp streams  |-- video -->| depacketize,      |   h264       |            |
|                       |  RFC 4175  | reconstruct,      |-- mpeg-ts -->| playback   |
|                       |-- audio -->| encode and stream |   udp/srt    |            |
+-----------------------+   AES67    +-------------------+              +------------+

Other tools like gstreamer may be used as a transcoder but the following study focuses on ffmpeg.

Install ffmpeg and dependencies

From top directory,

sudo -i
./install.sh common       # gcc, libtool, tar etc.
./install.sh transcoder   # depencies (yasm, nasm, x264, fdkaac, mp3, srt) and ffmpeg

Get a SMPTE 2110 source

If you don't have any source, consider using a sofware-based source for testing.

A SMPTE 2110 streams are described by an individual SDP file for each essence: video, audio, ancillary. However, ffmpeg hardly takes multiple SDP files as input. A hack consists in combining the essence descriptions in one single SDP. See example, where audio is declared before video to ensure that ffmpeg process this lighter stream first.

Simple test

Transcode AV streams to h264 low res file:

ffmpeg -loglevel debug \
    -buffer_size 671088640 -protocol_whitelist file,udp,rtp -i <sdp_file> \
    -fifo_size 1000000000 \
    -c:a libfdk_aac -ac 2 \
    -vf scale=640:480 \
    -c:v libx264 -preset ultrafast -pass 1 \
    output.mp4

Decode interlaced and re-stream to h264 @ 2.5Mbps over mpegts:

ffmpeg -loglevel debug \
    -buffer_size 671088640 -protocol_whitelist file,udp,rtp -i <sdp_file> \
    -fifo_size 1000000000 -passlogfile /tmp/ffmpeg2pass \
    -c:a libfdk_aac -ac 2 -b:a 128k \
    -r 30 -vf yadif=0:-1:0 \
    -s 1280x720 -pix_fmt yuv420p -c:v libx264 -profile:v main -preset fast \
    -level:v 3.1 -b:v 2500k -x264-params b-pyramid=1 -g 30 -keyint_min 16 -pass 1 -refs 6 \
    -f mpegts udp://<monitor_IP>:5000

On a monitoring host:

ffplay udp://@0.0.0.0:5000
vlc --network-caching 4000  udp://@0.0.0.0:5000

If no packets are received, refer to the troubleshoot guide.

RTP packet drops

This is the most common error you'll get. ffmpeg may complain about RTP discontinuity with messages including:

jitter buffer full
RTP: missed ******* packets
Missed previous RTP Marker
RTP: dropping old packet received too late

This is most likely due to the input buffer being too small or the transcode process creating a bottleneck that prevents ffpmeg from reading the incomming packets fast enough. Here are some of the knobs you can try to adjust:

  • downscale the image resolution
  • change CPU, see performance analysis
  • force multithread by applying this patch
  • tune up your network stack, see optimization section below
  • strip the command to bare minimum and start from here to :
ffmpeg -y -loglevel verbose -buffer_size 671088640 -protocol_whitelist 'file,udp,rtp' -i mysdp.sdp  -f null /dev/null

Start as a service

Once the process is stable, you can start it as a background task. Use ./transcode.sh to start the transcoding service in background from one or multiple SDP files, then show logs and, finally stop the service.

$ ./transcoder.sh help
[...]
$ ./transcoder.sh start file.sdp
==================== Start ... ====================
Transcoding from file.sdp
Scaling and encoding: cpu.
Audio codec is aac.
[...]
$ ./transcoder.sh log
[...]
$ ./transcoder.sh stop
==================== Stop ... ====================

Script constants, like destination IP, can be overridden by conf file, i.e. /etc/st2110.conf. See sample ./config/st2110.conf for details.

Optimization

System network stack

You can also check that your NIC ring buffer is the largest as possible:

sudo ethtool -g <iface>
sudo ethtool -G <iface> 8192

And consider increasing Kernel Rx buffer:

sysctl net.core.rmem_max=671088640
sysctl net.core.rmem_default=671088640
echo 10000 > /proc/sys/net/core/netdev_max_backlog

Verify the memory usage of your receiving socket:

ss -uamp | grep -A1 ffmpeg

UNCONN    1119285120    0   225.164.14.100:20000     0.0.0.0:*          users:(("ffmpeg",pid
    #     ^ bytes of data have been received by the kernel but haven’t yet been copied by the process
    skmem:(r1016405760,rb1342177280,t0,tb212992,f256,w0,o112,bl0,d75818)
    #      ^            ^max                                     ^pkt drops
    #      | current, should ideally be 0

Use ./transcoder/transcoder_stats.sh for complete stats.

FFmpeg options

RTP Input:

  • -reorder_queue_size: jitter buffer size, default is 500 pkts, not relevant if no reordering expected
  • -buffer_size: socket memory size in the kernel, overrides /proc/sys/net/core/rmem_default by setsockopt()
  • -max_delay
  • -fifo_size

See usage: ss -uamp | grep ffmpeg -A1

Raw:

  • -thread_queue_size: 8 AVPackets available by default
  • -vf yadif=0:-1:0: for de-interlacing

Output:

-pass 1 (h264): without this option the CPU usage is way higher and the audio breaks after a few seconds, at least for rtmp output. The 1st con of the option is that the output bitrate might less precise. And the generated passlog file is quite large (~10GB/day). The 'monitor' function of the transcoder checks the size of this file and restarts ffmpeg if needed.

Hardware acceleration for transcoding

Nvidia setup.

Transcoding performance

Measurements with CPU vs GPU.

FFmpeg files

The demux is composed of:

  • libavformat/sdp
  • libavformat/udp (multicast join)
  • libavformat/rtsp
  • libavformat/rtpdec
  • libavformat/rtpdec_rfc4175 (dynamic RTP handler)
  • libavcodec/bitpacked_dec

Limitations

Network redundancy

SMPTE 2022-7 is not supported. Eventhough, an SDP with dual stream will be correctly processed, the output would contains 2 separated tracks.

A/V synchro

Audio and video are transcoded as packets come in with no regard to their respective RTP timestamps. If temporal realignement is important, and if RTP timestamps are reliable, consider applying the following patches before re-compile:

  • ffmpeg-avutil-smpte2110-add-helpers-to-compute-PTS.patch
  • ffmpeg-avformat-rtp-compute-smpte2110-timestamps.patch

And activate in command line: -smpte2110_timestamp 1

Trancoding ancillary data (SMPTE ST 2110-40)

There are some limitations in transcoding closed caption.

Here are some guidelines for SCTE-35.