[FFmpeg-user] sytraming AAC+

[Robert Jeffares]

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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