Streaming

GStreamer has elements that allow for network streaming to occur. For example, the Yocto/gstreamer is an example application that uses the gstreamer-rtsp-plugin to create a rtsp stream.

However, creating a GStreamer application is not the only way to create a network stream. Simple GStreamer pipelines to accomplish this as well which is often used for testing purposes. The following examples are based on GStreamer-1.0 using gstreamer-imx plugins.

There are several ways to accomplish networked streaming over Internet Protocol (IP):

• Raw UDP/IP
• Raw TCP/IP
• Real-time Transport Protocol (RTP)
• Real Time Streaming Protocol (RTSP) (recommended)
• Adaptive Bitrate Streaming

To see older/deprecated information, please see this older revision page.

Raw UDP

Using UDP/IP is the simplest mechanism for streaming and utilizes the least amount of bandwidth. Because UDP does not provide any error detection, packet ordering, or error correction the bitrate is deterministic and simply the bitrate of the media you are streaming.

The limitations of raw UDP is:

• requires codec that can handle missing/corrupt data (most do these days)
• does not use headers containing payload type or timestamp info on stream (making it suitable for only a single type of media, or a pre-muxed type of media)
• does not fragment packets - will try to send a raw udp packet for whatever size buffer the udpsink is passed (which can lead to pipeline errors). To fragment packets use RTP

The only benefit of using raw UDP is that it is the simplest pipeline you can create for streaming and requires the least amount of dependencies (albeit you might run into one or all of the above problems).

Note that it is recommended that you use RTP or RTSP unless you know exactly what you are doing to overcome the limitations listed above

The udpsrc element can be used to render/save a stream originated from a udpsink pipeline.

Examples:

• encode and send H264 video from Ventana:
  1. Start decoder first:

     ifconfig eth0 192.168.1.1
     gst-launch-1.0 udpsrc port=9001 ! h264parse ! imxvpudec ! imxipuvideosink sync=false
     

  2. Start encoder second:

     ifconfig eth0 192.168.1.2
     gst-launch-1.0 videotestsrc is-live=true ! imxvpuenc_h264 bitrate=1000 ! udpsink host=192.168.1.1 port=9001
     

Notes:

• On the client (stream receiver and renderer) you must use the sync=false property to render frames as they are received otherwise the stream will stall because their is no headers containing timestamps
• the decoder (udpsrc) needs to be started first because udpsink will fail if nothing is listening to the socket

TCP

Using TCP/IP brings error detection, packet re-ordering, and error correction to the network stream. This however causes the bitrate to be non-deterministic because as the error rate increases so does the bitrate and latency.

The limitations of using TCP:

• non-deterministic bitrate
• added latency
• does not use headers containing payload type or timestamp info on stream (making it suitable for only a single type of media, or a pre-muxed type of media)

Note that it is recommended that you use RTP or RTSP unless you know exactly what you are doing to overcome the limitations listed above

TCP/IP introduces the concept of a socket connection therefore there must exist a server and a client in which case the server must be started first to listen for a connection. You can use a server sink or a server source. The tcpserversrc source can be used to create a TCP server that waits for a connection from a tcpclientsink to render/save. Alternatively the tcpserversink sink can be used to create a TCP server that waits for a connection from a tcpclientsrc that will send data.

Examples:

• encode and send H264 video from Ventana with decoder as server:
  1. Start decoder (server) first:

     ifconfig eth0 192.168.1.1
     gst-launch-1.0 tcpserversrc host=192.168.1.1 port=9001 ! decodebin ! autovideosink sync=false
     

  2. Start encoder (client) second:

     ifconfig eth0 192.168.1.2
     gst-launch-1.0 videotestsrc is-live=true ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! tcpclientsink host=192.168.1.1 port=9001
     

Notes:

• TCP is connection oriented therefore the TCP 'server' must be started first. You can choose your elements such that the stream originator is the server or the stream renderer is the server however doing so can be problematic for certain codecs because the client decoding the stream may pick up the stream somewhere in the middle and not know how to parse it.

RTP (raw/session-less)

The ​Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

The RTP packet type encapsulates multimedia data with a payload type and time-stamp and therefore can be used to compensate for jitter, out of sequence packets, and time synchronization between streams of different types (ie audio/video lip-sync).

RTP is typically used in conjunction with other protocols such as RTP Control Protocol (RTCP) and Real Time Streaming Protocol (RTSP) to manage stream sessions however can be used on its own in a raw session-less fashion using udpsink and udpsrc elements.

The limitations of using raw/session-less RTP:

• session management needs to be handled manually (capsfilter is needed to specify stream format)

Note that it is recommended that you use RTSP unless you know exactly what you are doing to overcome the limitations listed above

Examples:

• encode and send H264 video from Ventana:
  1. Start decoder first:

     ifconfig eth0 192.168.1.1
     gst-launch-1.0 udpsrc port=9001 \
      caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264" \
      ! decodebin ! autovideosink
     

  2. Start encoder second:

     gst-launch-1.0 videotestsrc is-live=true \
      ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay ! udpsink host=192.168.1.1 port=9001
     

Notes:

• when using RTP a capsfilter must be used to specify the payload as application/x-rtp as above. You can determine the capsfilter required by starting the encoder with a verbose flag -v and looking for caps = "application/x-rtp"

RTSP (Real Time Streaming Protocol) (recommended)

The ​Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between end points. Clients of media servers issue VCR-style commands, such as play and pause, to facilitate real-time control of playback of media files from the server. This protocol uses the Real-time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for media stream delivery.

The limitations of using RTSP are:

• gst-inspect has no way of using a simple pipeline to create an RTSP server - you must create or use an existing gstreamer based application (keep reading below).

Creating an RTSP server is not possible via a simple pipeline to gst-launch however GStreamer libraries do exist to make writing an RTSP server trivial. The source for gst-rtsp-server contains an example application ​test-launch.c which provides a simple example that can take a GStreamer 'bin' element consisting of everything but the sink element and serves it via RTSP.

An extension of the gst-rtsp-server test-launch application ​gst-variable-rtsp-server is included on our Yocto BSP images which will setup a RTSP server, encode the video stream to h264, and allow multiple clients to connect to it. The enhancements made to gst-variable-rtsp-server includes a mechanism for auto-adjusting the encoding bitrate depending on the number of clients connected in addition to serving as a fairly simple example of how to write a gstreamer application.

Notes:

• refer here for more info on gst-variable-rtsp-server.
• refer to the gstreamer/video and gstreamer/audio pages to understand how to first capture video and audio sources.
• refer to ventana/audio and Yocto/Video_In for more info on Ventana Audio input and Video input devices.
• RTP streams must use a payloader element appropriate for the media type for that stream. Additionally the payloader name property must be defined with the first stream starting at 0 (ie 'pay0') and the pt property must be set to a value according to ​RFC3551. Use gst-inspect-1.0 | grep rtp.*pay to see a full list of available payloaders.
• you can use playbin (ie gst-launch-1.0 playbin uri=rtsp://<server>:<port>/<stream>) as well as an RTSP client if you do not want to specify the various element details such as jitterbuffer latency.

Video only

encode and send H264 video from Ventana:

1. Start server (encoder) first:

   ifconfig eth0 192.168.1.2
   

   # video test source:
   gst-variable-rtsp-server -p 9001 -u \
    "videotestsrc ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96"
   

   # or alternatively live captured video
   gst-variable-rtsp-server -p 9001 -u \
    "imxv4l2videosrc device=/dev/video0 ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96"
   

   • see Yocto/Video_In for details on Ventana video capture devices (typically the first video capture device is HDMI if available, and otherwise analog CVBS)

2. Connect decoder client(s) second:

   ifconfig eth0 192.168.1.1
   

   # view with gstreamer rtspsrc
   gst-launch-1.0 rtspsrc location=rtsp://192.168.1.2:9001/stream latency=10 ! decodebin ! autovideosink
   

   # or with vlc
   vlc rtsp://192.168.1.2:9001/stream --rtsp-caching=10
   

Audio only

encode and send Audio only from Ventana:

1. Start server (encoder) first:

   ifconfig eth0 192.168.1.2
   

   # audio test source (tone generator) and AC3 audio encoding
   gst-variable-rtsp-server -p 9001 -u \
     "audiotestsrc ! avenc_ac3 ! rtpac3pay name=pay0 pt=97"
   

   # or audio test soruce (tone generator) and alaw G711 audio encoding
   gst-variable-rtsp-server -p 9001 -u \
     "audiotestsrc ! alawenc ! rtppcmapay name=pay0 pt=97"
   

   # or live audio input (of the first audio capture device) and alaw G711 audio encoding
   gst-variable-rtsp-server -p 9001 -u \
     "alsasrc device=hw:0,0 ! alawenc ! rtppcmapay name=pay0 pt=97"
   

   # or live audio input of HDMI audio (specified by card name) and alaw G711 audio encoding
   gst-variable-rtsp-server -p 9001 -u \
     "alsasrc device=sysdefault:CARD=tda1997xaudio ! alawenc ! rtppcmapay name=pay0 pt=97"
   

   • use arecord -L to list available audio input devices by name and arecord -l to list by number - see ventana/audio for more details

2. Connect decoder client(s) second:

   ifconfig eth0 192.168.1.1
   

   # via playbin
   gst-launch-1.0 -v playbin uri=rtsp://192.168.1.2:9001/stream
   

   # or via playbin calling out a specific audio output device (HDMI out in this case)
   gst-launch-1.0 playbin uri=rtsp://192.168.1.2:9001/stream audio-sink="alsasink device=sysdefault:CARD=imxhdmisoc"
   

   # or with vlc
   vlc rtsp://192.168.1.2:9001/stream --rtsp-caching=10
   

   • use aplay -L to list available audio output devices by name and aplay -l to list by number - see ventana/audio for more details

Codec Notes:

• We have seen issues decoding AC3 with GStreamer on Ventana - you may want to use alaw for compatibility.

Audio + Video

encode and stream H264 video and encoded audio from Ventana:

1. Start server (encoder) first:

   ifconfig eth0 192.168.1.2
   

   # test video (colorbars) and test audio (tone) via AC3 encoding:
   gst-variable-rtsp-server -p 9001 -u \
    "videotestsrc ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     audiotestsrc ! audioconvert ! avenc_ac3 ! rtpac3pay name=pay1 pt=97"
   

   # or test video (colorbars) and test audio (tone) via ulaw G711 encoding:
   gst-variable-rtsp-server -p 9001 -u \
    "videotestsrc ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     audiotestsrc ! alawenc ! rtppcmapay name=pay1 pt=97"
   

   # or live captured HDMI alaw audio and H264 video on a GW540x:
   gst-variable-rtsp-server -p 9001 -u \
    "imxv4l2videosrc device=/dev/video0 ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     alsasrc device=sysdefault:CARD=tda1997xaudio ! alawenc ! rtppcmapay name=pay1 pt=97"
   

   # or live captured HDMI AC3 audio and video H264 video on a GW540x:
   gst-variable-rtsp-server -p 9001 -u  "imxv4l2videosrc device=/dev/video0 ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     alsasrc device=sysdefault:CARD=tda1997xaudio ! audioconvert ! avenc_ac3 ! rtpac3pay name=pay1 pt=97"
   

   # or live captured CVBS audio and video on a GW540x:
   gst-variable-rtsp-server -p 9001 -u \
    "imxv4l2videosrc device=/dev/video1 ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     alsasrc device=sysdefault:CARD=sgtl5000audio ! audioconvert ! avenc_ac3 ! rtpac3pay name=pay1 pt=97"
   

   # or live captured CVBS audio and video on a GW510x:
   gst-variable-rtsp-server -p 9001 -u \
    "imxv4l2videosrc device=/dev/video0 ! imxipuvideotransform ! imxvpuenc_h264 bitrate=1000 ! rtph264pay name=pay0 pt=96 ! \
     alsasrc device=sysdefault:CARD=sgtl5000audio ! audioconvert ! avenc_ac3 ! rtpac3pay name=pay1 pt=97"
   

   • use arecord -L to list available audio input devices by name and arecord -l to list by number - see ventana/audio for more details

2. Connect decoder client(s) second:

   ifconfig eth0 192.168.1.1
   

   # via playbin
   gst-launch-1.0 -v playbin uri=rtsp://192.168.1.2:9001/stream
   

   # or via playbin calling out a specific audio output device (HDMI out in this case)
   gst-launch-1.0 playbin uri=rtsp://192.168.1.2:9001/stream audio-sink="alsasink device=sysdefault:CARD=imxhdmisoc"
   

   # or manually specifying sinks
   gst-launch-1.0 rtspsrc location=rtsp://192.168.1.2:9001/stream latency=10 name=demux \
    demux. ! decodebin ! autovideosink sync=true \
    demux. ! decodebin ! autoaudiosink sync=true
   

   # or with vlc
   vlc rtsp://192.168.1.2:9001/stream --rtsp-caching=10
   

   • use aplay -L to list available audio output devices by name and aplay -l to list by number - see ventana/audio for more details

Codec Notes:

• We have seen issues decoding AC3 both with GStreamer on Ventana and with VLC when using audio+video - you may want to use alaw for compatibility.

File based Audio + Video

stream file based audio+video:

1. Start server first:

   ifconfig eth0 192.168.1.2
   

   gst-variable-rtsp-server -p 9001 -u \
    "filesrc location=/mnt/usb/open-media/tears_of_steel_1080p.webm typefind=true do-timestamp=true ! \
     matroskademux name=demux \
     demux. ! queue2 ! rtpvorbispay name=pay0 \
     demux. ! queue2 ! rtpvp8pay name=pay1"
   

2. Connect decoder client(s) second:

   ifconfig eth0 192.168.1.1
   

   gst-launch-1.0 rtspsrc location=rtsp://192.168.1.2:9001/stream latency=2000 name=demux \
    demux. ! decodebin ! queue2 ! autovideosink sync=true \
    demux. ! decodebin ! queue2 ! autoaudiosink sync=true
   

   • the larger latency can help account for audio/video timestamp discontinuities in the encoded file source

Adaptive Bitrate Streaming

Adaptive bitrate streaming is the concept of a video lowering its' image quality based on its' network quality. This is often seen in situations of online media streaming from services such as YouTube and Netflix where a lower quality connection will receive SD quality video, which a higher quality connection will receive HD.

Some common protocols that exist are: ​HLS (Created by Apple Inc.), ​MPEG DASH, ​SmoothStreaming (Created by Microsoft). Please note that these protocols are not provided on any BSPs by Gateworks.

Gateworks has decided to create a sample application that features our implementation of adaptive bitrate live video streaming for our customers. Please see the below section for more details.

Gateworks Adaptive Bitrate solution (RTSP)

For low latency live video streaming, RTSP might be a good choice. Taking the data found on our Latency page, we see that live streaming with RTSP had a low end-to-end latency of just 98ms when capturing with an analog CVBS camera (this is including latency in the camera itself).

The reason we are including this information under the "Adaptive Bitrate" section is due to the fact that our gst-variable-rtsp-server has the ability to change bitrate on the fly. Our implementation relies on the number of clients currently connected. The quality of the stream will decrease as more users join the stream and increase with less users. This simple GStreamer application is fully open-sourced so you may reference how to do something similar, maybe utilizing other information to determine stream quality. Please visit the ​GitHub page ​here to get started.

For more detail on this application, please visit our gst-variable-rtsp-server wiki page on the topic.

References

• ​HTTP Adaptive Streaming with GStreamer
• ​Live streaming web audio and video by Mozilla

Troubleshooting

If you're having issues with network streaming:

• Verify that both sides can ping one another
• If the message There may be a timestamping problem, or this computer is too slow appears and the video display appears choppy, try the following:
  • Lower the bitrate from the server
  • Place a sync=false on the sink side of the server and client.
• If video appears choppy, try using UDP over TCP.
• Verify that the network is not congested.
• Verify your gstreamer pipeline is correct. The best way to find the element that causes a negotiation failure is to end your pipeline in a fakesink and one-by-one eliminate elements leading up to it until it negotiates successfully.
• When encoding streams from video input devices, you will need an imxipuvideotransform if using HDMI capture in the yuv422smp mode. It doens't hurt to add one regardless as it will be skipped if not needed.