ffmpeg not honoring sample rate in opus output
<p>I am capturing a live audio stream to Opus, and no matter what I choose for the audio sample rate, I get 48khz output.</p>

<p>This is my command line</p>

<blockquote>
  <p>./ffmpeg -f alsa -ar 16000 -i sysdefault:CARD=CODEC -f
  alsa -ar 16000 -i sysdefault:CARD=CODEC_1 -filter_complex
  join=inputs=2:channel_layout=stereo:map=0.1-FR\|1.0-FL,asetpts=expr=N/SR/TB
  -ar 16000 -ab 64k -c:a opus -vbr off -compression_level 5 output.ogg</p>
</blockquote>

<p>And this is what ffmpeg responds with:</p>

<blockquote>
  <p>Output #0, ogg, to 'output.ogg':    Metadata:
       encoder         : Lavf57.48.100
       Stream #0:0: Audio: opus (libopus), 16000 Hz, stereo, s16, delay 104, padding 0, 64 kb/s (default)
       Metadata:
         encoder         : Lavc57.54.100 libopus</p>
</blockquote>

<p>However, it appears that ffmpeg has lied, because when analysing the file again, I get:</p>

<blockquote>
  <p>Input #0, ogg, from 'output.ogg':   Duration: 00:00:03.21, start:
  0.000000, bitrate: 89 kb/s
      Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, delay 156, padding 0
      Metadata:
        ENCODER         : Lavc57.54.100 libopus</p>
</blockquote>

<p>I have tried so many permutations of sample rate, simplifying down to a single audio input etc etc - always with the same result.</p>

<p>Any ideas?</p>