CODEWITHSUNDEEP
linuxgstreamer
ameeth
ameeth

Reputation: 31

gstreamer audio error on linux

i am using g streamer-0.10 on Ubuntu os for streaming an web cam video on to an rtmp server i am getting an video output but their is a problem in audio . Below command used for streaming

gst-launch-0.10 v4l2src ! videoscale method=0 ! video/x-raw-yuv,width=852,height=480,framerate=(fraction)24/1 ! ffmpegcolorspace ! x264enc pass=pass1 threads=0 bitrate=900 tune=zerolatency ! flvmux name=mux ! rtmpsink location='rtmp://..../live/testing' demux. alsasrc device="hw:0,0" ! audioresample ! audio/x-raw-int,rate=48000,channels=2,depth=16 ! pulseaudiosink Blockquote

by running the above command i got an error

gstbaseaudiosrc.c(840): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Dropped 13920 samples. This is most likely because downstream can't keep up and is consuming samples too slowly. Blockquote

so the audio is not audible.

Help me out to solve this problem. Thanks in advance Ameeth

Upvotes: 2

Views: 3645

Answers (2)

johnboiles
johnboiles

Reputation: 3524

For me (Logitech c920 on Raspberry Pi3 w/ GStreamer 1.4.4) I was able to get rid of the "Dropped samples" warning by using audioresample to set the sampling rate of the alsasrc to something that flvmux liked. From gst-inspect-1.0 flvmux, it looks like flvmux only supports 5512, 11025, 22050, 44100 sample rates for x-raw and 5512, 8000, 11025, 16000, 22050, 44100 for mp4. Here's my working pipeline

gst-launch-1.0 -v -e \
    uvch264src initial-bitrate=800000 average-bitrate=800000 iframe-period=2000 device=/dev/video0 name=src auto-start=true \
    src.vidsrc ! video/x-h264,width=864,height=480,framerate=30/1 ! h264parse ! mux. \
    alsasrc device=hw:1 ! 'audio/x-raw, rate=32000, format=S16LE, channels=2' ! queue ! audioresample ! "audio/x-raw,rate=44100" ! queue ! voaacenc bitrate=96000 ! mux. \
    flvmux name=mux ! rtmpsink location="rtmp://live-sea.twitch.tv/app/MYSTREAMKEY"

I was surprised that flvmux didn't complain about getting an audio source that was at an unsupported sampling rate. Not sure if that's expected behavior.

Upvotes: 0

av501
av501

Reputation: 6729

I don't understand your pipeline. What is "demux." in the middle? The problem you are facing is because you have not seperated your elements with queues. Keep a queue before your sinks and after your sources to give the rest all seperate threads to run. It should allow get rid of the issue.

Since I don't have pulse audio or rtmp reciever in my system i have tested out the following and it works.

gst-launch-0.10 v4l2src ! ffmpegcolorspace ! queue ! x264enc pass=pass1 threads=0 bitrate=900000 tune=zerolatency ! queue ! flvmux name=mux ! fakesink  alsasrc ! queue ! audioresample !  audioconvert !  queue ! autoaudiosink

You can change it accordingly and use it. The only thing I had to do to make it work and remove the error your are facing is to add the queues.

Upvotes: 1

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