[FFmpeg-trac] #6625(undetermined:new): "Freezes" transcoding RTP g.711 stream to mp3
FFmpeg
trac at avcodec.org
Wed Aug 30 16:08:42 EEST 2017
#6625: "Freezes" transcoding RTP g.711 stream to mp3
-------------------------------------+-------------------------------------
Reporter: sagonzal | Owner:
Type: defect | Status: new
Priority: normal | Component:
Version: unspecified | undetermined
Keywords: rtp | Resolution:
Blocking: | Blocked By:
Analyzed by developer: 0 | Reproduced by developer: 0
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Comment (by sagonzal):
Truthfully, I’m not entirely certain. My command used to just be:
ffmpeg -re -f mulaw -i rtp://10.200.1.14:32760 -acodec libmp3lame live-
rtp-g711-toMP3.mp3 -report
only it didn't work with live intercom audio (0 packets read/encoded,
etc.), and with the Wireshark audio resulted in an mp3 that sounded very
high-pitched and fast. After adding sample_rate and asetrate of 9000, the
mp3 sounded "normal". I figured the same could be applied to the intercom
audio.
I should note that when I spoke with the company that makes the intercoms,
I was told that the g.711 audio stream would send as either '''7kHz''' or,
more likely, as '''3.4kHz'''. I haven’t gotten those rates to work for me
with either rtp or .raw input.
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Ticket URL: <https://trac.ffmpeg.org/ticket/6625#comment:2>
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