CODEWITHSUNDEEP
gstreamer-1.0gst-launch-1.0
Jerremy Koot
Jerremy Koot

Reputation: 11

gst-launch-1.0 - rtspsrc audio/video issue

I'm trying to combine two RTSP streams using gst-launch-1.0 but I'm already stuck already at trying to record/play one RTSP stream. The stream contains both audio and video.

The command line I'm using is:

gst-launch-1.0 ^
               rtspsrc location=<location-omitted> protocols=GST_RTSP_LOWER_TRANS_TCP latency=5000 name=s_0 ^
               s_0. ! application/x-rtp,media=audio ! rtpjitterbuffer ! decodebin ! audioconvert ! autoaudiosink ^
               s_0. ! application/x-rtp,media=video ! rtpjitterbuffer ! decodebin ! videoconvert ! autovideosink

If I change the autoaudiosink to fakesink, the video plays. If I remove the video sink (didn't test with fakesink), the audio plays. But if I add both (like above), the video shows 1 frame and then freezes. [not sure if it matters, but I am on Windows]

I actually suspect that (for some reason) the pipeline goes on pause, but I'm at a loss on how to debug this issue.

I've tried with/without rtpjitterbuffer, I've tried with/without queue's at various places. I've tried with combinations of rtp...depay/parse. Although I've tried so much in the past few hours (ohoh) that I'm not sure if I overlooked anything.

Also tried with various options for rtspsrc (sync/etc).

But the end result is pretty much always the same, playing either audio or video works fine, playing both at the same time (or muxing them into a single file) fails.

Writing each to their own file works fine, for example:

gst-launch-1.0 ^
               rtspsrc location=<location-omitted> protocols=GST_RTSP_LOWER_TRANS_TCP latency=5000 name=s_0 ^
               s_0. ! application/x-rtp,media=audio ! rtpjitterbuffer ! decodebin ! audioconvert ! avenc_aac ! flvmux ! filesink location=audio.flv ^
               s_0. ! application/x-rtp,media=video ! rtpjitterbuffer ! decodebin ! videoconvert ! x264enc ! flvmux ! filesink location=video.flv

If I mux the above into one file the same thing happens as with trying to play the file (using the auto-sink), the output file stays at 0-bytes.

The output of gst-launch-1.0 when trying to play is the same as when writing to two files:

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to <location-omitted>
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
Progress: (request) SETUP stream 0
Progress: (request) SETUP stream 1
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending PLAY request
Progress: (request) Sending PLAY request
Progress: (request) Sent PLAY request
Redistribute latency...
Redistribute latency...

Somehow I think it's some type of sync-issue, but I can't seem to figure out how to solve it.

Upvotes: 1

Views: 2614

Answers (1)

Hans Wong
Hans Wong

Reputation: 11

Since this question is more than 1 year ago, I hope someone with same problem still need the help (as I do, I used many hours to figure this out, and thank you to the poster he gives me the hints)

You need to edit as this

gst-launch-1.0 ^
rtspsrc location=<location-omitted> protocols=GST_RTSP_LOWER_TRANS_TCP latency=5000 name=s_0 ^
s_0. ! application/x-rtp,media=audio ! rtpjitterbuffer ! decodebin ! audioconvert ! avenc_aac ! flvmux name=mux ^
s_0. ! application/x-rtp,media=video ! rtpjitterbuffer ! decodebin ! videoconvert ! x264enc ! mux. ^
mux. ! filesink location=video.flv

Upvotes: 1

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