Thankyou Moritz,
my aac input is generated in glasscoder and mounted on Icecast2.
I have been looking at rtp as a possibly more reliable transport
platform for studio transmitter links.
We have barix 100 as receivers and mp3 streams are subject to network
loading.
I have switched to aac with success and had hoped to move from http to
rtp which is supposed to be more robust.
rtp works fine with mp3 coding. I may make that one of the 3 route options.
rtp with aac generates but VLC says SDP is needed
A description in SDP format is required to receive the RTP stream. Note
that rtp:// URIs cannot work with dynamic RTP payload format (97
I tried including a sdp file but clearly the code ignores it.
I will work with what I have and hopefully it will get sorted sometime.
regards
Robert
On 22/03/21 9:51 pm, Moritz Barsnick wrote:
Hi Robert,
On Mon, Mar 22, 2021 at 18:08:23 +1300, Robert Jeffares wrote:
I am getting...
[rtp @ 0x55e52ffe8c00] AAC with no global headers is currently not
supported.
You are usually asked to provide the full, uncut console output of your
command.
ffmpeg -i http://localhost:8000/wireless.aacp -acodec copy -vn -flags
+global_header -f rtp rtp://192.168.1.100:7001
is there support for aac coming?
AAC is supported. Your input seems to be raw AAC, which ffmpeg cannot
(yet) add information about to RTP. Since you are copying, not
reencoding, the option "-flags +global_header", which is encoding
flags, has no effect.
You might try a different RTP format as suggested here:
https://video.stackexchange.com/a/24454
i.e. using
-rtpflags latm
but I don't know whether your RTP target supports this.
Cheers,
Moritz
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