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Lossless SRT audio
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Joined: 4/13/2015(UTC) Posts: 4
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Hello All,
Sending SRT streams from jazz music festivals to the studio. At the festival venue professional 48 KHz 24 bit audio provided. When I create an SRT stream in vMix, it encodes the audio to lossy AAC. In the studio this audio stream needs to be decoded, processed and encoded back to AAC again before streaming.
For our audience audio quality is very important, therefore we would like to avoid the unnecessary AAC encoding at the venue and decoding at the studio, but would like to send the audio lossless/untouched to the studio. This could be either PCM, FLAC or anything else lossless... There is 100+ Mbps connection between the venues and the studio therefore there is really no need for lossy audio compression.
I was doing some tests with lossless audio in SRT: - ffmpeg command line & OBS: encoder audio pcm_s24le/pcm_s16le, video libx264; container Matroska - OBS: audio only encoder FLAC; container FLAC
I was able to play these test SRT streams with VLC, ffmpeg/ffplay but not in vMix. Since vMix uses ffmpeg I am pretty sure that the same can be done in vMix as well.
Could you please help me how can I create & receive SRT stream with lossless audio in vMix?
Thank you!
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Rank: Advanced Member
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Joined: 2/23/2019(UTC) Posts: 135 Thanks: 16 times Was thanked: 28 time(s) in 25 post(s)
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They have said SRT decoding is GPU accelerated so I doubt it is using ffmpeg in the way you expect. Though I'd expect PCM audio wouldn't be too complicated to deal with I'm not sure how common your requirement would be.
vMix SRT interface allows you to set a very high audio bitrates. Is there really going to be a noticeable difference between audio that is very lightly compressed to say 1.28mbps AAC and then compressed much more heavily (say 256kbps) for streaming and audio that just has the heavy compression? I doubt it. If you set the AAC bitrate to more than the raw data rate then in theory it wouldn't need to do any compression - but probably would still do the basics.
If you have ffmpeg and OBS working then you could stick to that for the transport then send to vMix with NDI.
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Joined: 1/25/2019(UTC) Posts: 302 Thanks: 17 times Was thanked: 79 time(s) in 60 post(s)
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There is a free SRT Streamer Software available from the company that manufactures SRT Miniserver. I'm not sure what kind of audio is transmitted, but it's worth a try. Input via NDI or Decklink. SRT Streamer
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Rank: Advanced Member
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Joined: 3/20/2014(UTC) Posts: 2,721 Location: Bordeaux, France Thanks: 243 times Was thanked: 794 time(s) in 589 post(s)
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There is something paradoxal in your request: SRT is designed to transport H.264/AAC or TS streams with low bitrates over the internet.
However I think it should still work if you mux H.264 + WAV. Never tested though, to be confirmed. Technically it's easy to test using vMix External out which delivers uncompressed audio and video feeds to any software encoder on the same host. I'd go for FFMPEG using the "-ac copy" parameter for audio (so that it's not compressed).
OTOH it's not clear what your workflow is, if you're live or post, if you're sending multiple cams or just a mix to the studio, etc, so there might be more appropriate solutions to transport uncompressed audio along with video feeds from the venue to the studio. I won't elaborate on this as I miss too much info about your production context but we've been transporting uncompressed audio over the internet for over 20 years, so there might options to consider before going for SRT.
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Rank: Advanced Member
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Joined: 2/23/2019(UTC) Posts: 135 Thanks: 16 times Was thanked: 28 time(s) in 25 post(s)
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Originally Posted by: DWAM SRT is designed to transport H.264/AAC or TS streams with low bitrates over the internet. SRT is a transport protocol. Though MPEG-TS with H.264/AAC is the most common payload, it will transport any type of data you want. It just tries to make sure packets leave the receiver a fixed latency after they were received by the transmitter, recovering lost packets where time allows. But most SRT receivers expect to be handling MPEG-TS and vMix supports H.264 or HEVC and AAC.
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Rank: Newbie
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Joined: 4/13/2015(UTC) Posts: 4
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Thank you All for the answers so far! Originally Posted by: DWAM There is something paradoxal in your request: OTOH it's not clear what your workflow is, if you're live or post, if you're sending multiple cams or just a mix to the studio, etc, so there might be more appropriate solutions to transport uncompressed audio along with video feeds from the venue to the studio. I won't elaborate on this as I miss too much info about your production context but we've been transporting uncompressed audio over the internet for over 20 years, so there might options to consider before going for SRT. Yes, that is the goal to send video with lossless audio and in the studio little bit of post processing happening adding comments etc. and we would like to avoid the multiple audio encoding/decoding cycles. Great to hear that you've been transporting uncompressed audio for such a long time. Could you please share some good techniques? (preferably with similar error tolerance/latency as SRT) Thank you!
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