I am trying to stream audio from my Pi Zero and my I2s MEMS mic. I would like to stream using FFMPEG and ALSA (which I have already compiled) but I'm running into stuttering issues.


    ~/special/ffmpeg/ffmpeg -report -f alsa -ar 48000 -ac 2 -acodec pcm_s32le -i mic_sv -f lavfi -i testsrc -c:v h264_omx -c:a aac -ab 32k -bufsize 32k -f flv rtmp://

This results in constant stuttering and choppiness.

  1. Arecord piped directly to FFMPEG

    arecord -Dmic_sv -c2 -r48000 -fS32_LE | ~/special/ffmpeg/ffmpeg -report -acodec pcm_s32le -i - -f lavfi -i testsrc -c:v h264_omx -acodec aac -ab 32k -bufsize 32k -f flv rtmp://

This results in a coherent audio stream, but with skipping every 5 seconds or so.

  1. Arecord recorded to a wav file, piped into FFMPEG

    arecord -Dmic_sv -c2 -r48000 -fS32_LE -twav temp.v & ~/special/ffmpeg/ffmpeg -report -re -i temp.v -f lavfi -i testsrc -c:v h264_omx -ac 2 -acodec aac -ab 32k -bufsize 32k -async 2 -f flv rtmp://

This results in a perfect audio stream.

I don't know why #3 works but #2 and #1 cause problems. Any suggestions?

  • Is there a difference in latency? As a very casual guess I'd say #3 exploits the OS the most, buffering wise (but probably comes at the cost of greater latency). Also if you aren't already, use a tmpfs based location for the file instead of regular storage (on stock Raspbian there's just /run, but you can create your own). This will be a little more robust.
    – goldilocks
    Feb 7, 2017 at 19:52
  • You have an extra option in #3: -async 2. What happens if you add that option to #1 and #2? Do #1 and #2 still stutter?
    – Hydraxan14
    Apr 20, 2017 at 19:57

1 Answer 1


The main priority is to determine where the bottleneck applies. It could be with computational load or with excessive total bitstream rate. It would first be a good idea to eliminate the video, especially as you're doing nothing obvious to set its bitrate or framerate. You only have -f lavfi -i testsrc -c:v h264_omx for the video options. You don't say anything about testsrc.

All the video data in the RTMP multiplexed stream needs to be sent over your internet connection in a timely manner and data rate will have a big effect on the quality of your stream playback so you probably need to tune your video encoder. You can find the full list of video encoding options using ffmpeg -h encoder=h264_omx.

Try running ffmpeg without the audio, and then without the video while monitoring the CPU usage and final bitrate to see if either is excessive. Since you want an RTMP stream you should do this by replacing video or audio with a small black frame or with silence so you always have both audio and video in the multiplexed stream.

For example to reduce video load, use options like:

-f lavfi -i color=c=black:s=320x240:rate=10 -pix_fmt yuv420p -c:v h264_omx

This will encode a black 320x240 image at 10fps. To encode silent audio you can add a graph input like this:

-f lavfi -i anullsrc=channel_layout=stereo:sample_rate=44100

Also use the -loglevel debug to see more of what is happening and to check if any slow software format conversion filters are being auto inserted.

To be honest it sounds like a CPU bottleneck because the option that works best for you is to encode the audio and then after that encode the video and stream with the pre-processed audio. Your first two scenarios have the CPU doing both things at the same time. To reduce video encoding load you should look towards reduced frame rate, reduced frame size, lower bitrate, and any other speed setting optimizations on the encoder.

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