Replies: 12 comments 2 replies
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Do you know where is the documentation on how to packetize/depacketize OPUS in MPEG-2 TS? And how did you transmit OPUS over SRT? |
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I compiled ffmpeg with srt and opus codec and used command like this: |
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I can't find any documentation that can send OPUS over MPEG-2 TS. I thought OPUS/MPEG2-TS was not supported. If anyone knows this, I would be very grateful if you could send me a link to the documentation or information. |
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I don't know if it's supported or not. But i wanna to move audio transcoding from Oven to encoding server. |
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You can avoid opus encoding if you use webrtc input. |
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Just tested this with ffmpeg as caller and listener. While some codecs get assigned as Private data streams in ffmpeg opus seems to be muxed just fine and is automatically decoded on the receiving end (ffmpeg) while other codecs that get muxed as private data have to be manually extracted. |
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@getroot just +1'ing this request we could really use this in our workflow as well. there's an OPUS IRC and SRT Alliance Slack i can post in and see if i can get some info passed along. |
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@getroot OPUS IRC for the win! ( https://kiwiirc.com/nextclient/#ircs://irc.libera.chat/opus ) Some links they provided. MPEG TS Encap spec that they are working on. Tips on checking how ffmpeg is handling the OPUS codec And apparently GStreamer is able to handle the OPUS codec if you wanted to look into that some. Hope this helps! Happy to test anything you can come up with. |
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@mithrand1rr As I have seen so far, FFMPEG still handles OPUS as a private stream of MPEG-TS, so it can be used only when the receiving end is aware that the private stream is OPUS in advance. This is not an acceptable method for streaming servers. I will work on the OPUS/MPEG-TS draft you sent me when it becomes a formal specification. (Even before the document becomes the official version, a little bit in advance if I'm not busy) |
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@getroot sounds good and totally understand! They sent an updated Draft docs last night here is the link I'll let you know if i hear any other updates. And feel free to let me know if there's something in the future to test out. Happy to lend a hand. |
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This issue has been automatically marked as stale because it has not had recent activity. It will be closed if no further activity occurs. Thank you for your contributions. |
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Still need this feature. Probably, may it implemented as ffmpeg's developers does? This is great for reducing online transcode and CPU using. |
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I don't like implementing non-standards in OME. |
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I agree, WHIP is the good choose but there are no simple tool like ffmpeg that allow to use WHIP directly. Is there any WHIP CLI utilites or examples of using gstreater with this interface? Googling still no luck. |
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Is your feature request related to a problem? Please describe.
SRT not support OPUS codec.
[2021-06-15 12:57:47.681] E [SPSRT:81] MPEGTS_DEPACKETIZER | mpegts_depacketizer.cpp:555 | Unsupported codec has been received. (pid : 257 stream_type : 6)Describe the solution you'd like
Add OPUS codec for incoming SRT stream. Then audio will be prepared to WebRTC in source, not transcoded with OvenMediaEngine.
https://airensoft.gitbook.io/ovenmediaengine/live-source/srt-beta
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