On 4/4/24 10:53, Nagulan Bsc cs wrote:
Dear FFmpeg Team,
I hope this email finds you well. I am writing to seek your assistance with
a project involving audio recording using FFmpeg.
I am currently working on a project that requires capturing both the input
and output audio from mobile devic
Semantics are a little different presumably but the concept
is the same: presumably, an underflow condition exists when
a consuming process may have the expectation of available data
but cannot satisfy that consumption because of (presumably
then unavailable, perhaps by error) lack of producer dat
How can I make it faster?
Thanks!
I don't know the benchmarks, but you might want to look at Lame
for a Wave-to-MP3 solution.
It is certainly a generally available package under most
Linux distros; don't know if it is available for Win
and OSX (latter more likely?)
https://lame.so
On 04/22/2020 02:04 PM, Carl Zwanzig wrote:
> On 4/22/2020 10:39 AM, H wrote:
>
>> I downloaded the latest version for CentOS 7 from the ffmpeg website
>> after your first e-mail and my two previous posts were based on that.
> And that is not the _latest_ ffmpeg (forget abo
On 04/22/2020 12:38 PM, Carl Zwanzig wrote:
> On 4/22/2020 7:40 AM, H wrote:
>> Yes, I was using the latest version of youtube-dl from the CentOS 7
>> repository.
>
> On 4/22/2020 9:15 AM, Edward Park wrote:
>> But your newest version of ffmpeg is still pretty old, yo
On 04/21/2020 10:32 PM, Edward Park wrote:
> Hi,
>> [tls @ 0x22e0d00] The TLS connection was non-properly terminated.
>> [tls @ 0x22e0d00] The specified session has been invalidated for some reason.
>> [tls @ 0x2309820] The TLS connection was non-properly terminated.te=
>> 224.4kbits/s
> It doe
On 04/21/2020 09:03 PM, Tom Sparks wrote:
> Are you sure you are using the latest version of youtube-dl from
> https://youtube-dl.org/ ?
> Are you using the latest version of python?
>
>
> On 22/04/2020, H wrote:
>> On 04/20/2020 02:58 PM, Carl Zwanzig wrote:
>>&g
On 04/20/2020 02:58 PM, Carl Zwanzig wrote:
> On 4/20/2020 11:42 AM, H wrote:
>> I am running ffmpeg 2.8.15 under CentOS 7, the latest version released for
>> that operating system.
>
> First- that version is positively _ancient_ and not supported at all.
> Download
I am running ffmpeg 2.8.15 under CentOS 7, the latest version released for that
operating system.
Yesterday I recorded a YouTube stream for later playback. However, playing it
back in ffmpeg results in the dreaded:
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x1d5ac20] moov atom not found
abb.mp4: Invalid data
On 6/25/19 6:11 PM, Peter B. wrote:
Hi everyone :)
Is it possible somehow to transcode any source audio format type to its
raw, uncompressed format that matches the source?
I'm dealing with collections of mixed input format combinations, and am
trying to find a way to bash-automate something.
Thanks a lot for the help guys! The problem is now solved and it's working
perfectly.
Funny enough I was using a FFMPEG build from 2 days ago! Maybe it was
erroneously compiled?
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Sorry for that, didn't understand it at first...
The command I'm exeucting is the following one...
C:\Users\myuser\myproject>java -jar streamer/strea
mer.jar 6 53122 | ffmpeg/bin/ffmpeg -f s16le -ar 48000 -ac 2 -i - -f ogg
icecast
://hackme:hackme@localhost:8000/streaming.ogg
And the output...
Thanks a lot for the quick reply Mortiz, really appreciate it.
My god the > and | operator... And I wonder if this even worked back then
like this! Thanks a lot for the heads up with that, totally forgot about it.
Now that I've changed the operator, the output shows again but it seems
that it's "
Thanks a lot for the quick reply Mortiz, really appreciate it.
My god the > and | operator... And I wonder if this even worked back then
like this! Thanks a lot for the heads up with that, totally forgot about it.
Now that I've changed the operator, the output shows again but it seems
that it's "
Hey guys,
I'm really struggling with the following problem, a problem I had solved
few months ago and now that I revisit it, it doesn't work anymore...
At the moment I have a Java desktop application which spits audio bytes
through the stdout, bytes that ffmpeg must receive. I've done this using
Hello guys,
I have a Java application which writes into a file without extension all
the bytes that are read from a given mixer in the system. Those raw bytes
should be encoded into a .ogg file so that it can be sent in real time to
Icecast and stream it to the net.
The fact is that I'm facing pr
Hi,
I am trying to run remuxing.c example for remuxing a TS file containing
h264, aac into FLV format. Initially I was getting an error due to ADTS
header in AAC. So I have added aac_adtstoasc filtering to audio stream. Now
I don't get that error any more.
The TS packets that are being fed are no
Hi,
I am a libavformat noob. Please pardon if my question is stupid. I was
trying to modify and run the examples that are provided with ffmpeg. File
avio_reading.c shows how to read data into AVIOContext. As I understand
this uses pull mode, meaning AVIOContext calls read_packet callback
whenever
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