Re: [FFmpeg-user] Hls_list_size
2019-04-19 1:09 GMT+02:00, Benjamin Müller : > I have set up a server and are trying to stream a hls video > to browser. Now I have set the hls_list_size to ten but the > hls files still goes up endless. Command line and complete, uncut console output missing. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Hls_list_size
Dear ladies and gentlemen, I have set up a server and are trying to stream a hls video to browser. Now I have set the hls_list_size to ten but the hls files still goes up endless. What could possibly be the error? Yours faithfully Benjamin Müller ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] AWS Lambda h264 ratecontrol_init: can't open stats file
Moritz,
Thank you oh so much!
Got the general process working. Now i'll try to PIPE the source and output
file from/to S3.
On Tue, Apr 16, 2019 at 12:41 AM Moritz Barsnick wrote:
> Hi Douglas,
>
> On Mon, Apr 15, 2019 at 16:08:15 -0700, Douglas Rhiner wrote:
> > *Pass One Command*
> > os.system('/tmp/ffmpeg -y -i /tmp/789789789/789789789.mp4 -c:v libx264
> > -strict -2 -b:v 7000k -maxrate 7000k -bufsize 5250.0k -vf scale=-2:960
> -an
> > -pass 1 -passlogfile my_passlog_789789789_960_7000 -f mp4
> > /tmp/789789789/temp_789789789_960_7000')
> >
> >- [libx264 @ 0x5d81cc0] ratecontrol_init: can't open stats file
> >- Error initializing output stream 0:0 -- Error while opening encoder
> >for output stream #0:0 - maybe incorrect parameters such as bit_rate,
> rate,
> >width or height
> >- Conversion failed!
>
> libx264 isn't very good at log messages. It obviously has an issue
> accessing the "stats file", which is the two-pass file which you
> specified as "my_passlog_789789789_960_7000".
>
> To see in more detail what is going wrong, you could issue the command
> with strace (with a few extra command line parameters).
>
> My guess is: The command is being launched from a location (current
> directory) where it doesn't have write access. Therefore libx264 and/or
> ffmpeg cannot create "./my_passlog_789789789_960_7000". You should try
> giving the "-passlogfile" a filename with a full path to a writable
> location - "/tmp/my_passlog_789789789_960_7000" comes to mind as the
> most obvious choice.
>
> > *Pass two Command*
>
> You probably shouldn't be running the second command if the first one
> fails. When scripting, try to take return codes into consideration,
> otherwise your scripts may fall over terribly. (Not just in the case of
> using ffmpeg.) You might make assumptions that something worked - e.g.
> creating a directory or removing something - and that has effects on
> what happens afterwards.
>
> Cheers and good luck,
> Moritz
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--
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JD Publishing
CTO
C.530-448-2754
Skype.douglasrhiner
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Incline Village, NV
89451
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[FFmpeg-user] Need info on ATSC3 stream capture going wrong
Hello, this is a really strange one for me that is way outside my understanding. I've been using ffmpeg to stream tv shows and I've run across a TV station I can not capture! The symptoms are after 12:55 or so minutes the audio goes silent in the output, and shortly after that ffmpeg quits. This works with some 40 other tv channels! I've managed to narrow down my problem to -async. Put simply, these commands work: ffmpeg -i "/dev/dvb/adapter5/dvr0" -y out.ts ffmpeg -i "/dev/dvb/adapter5/dvr0" -async 1 -y out.ts and this does not: ffmpeg -i "/dev/dvb/adapter5/dvr0" -async 2000 -y out.ts And I get similar output from my previous report: [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] compensating audio timestamp drift:-0.001021 compensation:-49 in:48000 [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] Using fltp internally between filters [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] adding 16384 audio samples of silence [graph_1_aresample_in_0_0 @ 0x563ce00fa0c0] [SWR @ 0x563ce063a8c0] adding 15312 audio samples of silence /dev/dvb/adapter5/dvr0: Value too large for defined data type [out_0_1 @ 0x563ce00f9d40] EOF on sink link out_0_1:default. /dev/dvb/adapter5/dvr0: Value too large for defined data type [out_0_0 @ 0x563ce05928c0] EOF on sink link out_0_0:default. No more output streams to write to, finishing. Looking for some ideas on the nature of my problem. The log output that talks about "compensating audio timestamp drift" clearly gets it -49 value as number of samples needed to compensate for the drift (-.001021 seconds * 48000 samples/second = -49 samples). So why does it try to add 16384 + 15312 samples? Is that it can only compensate in a positive direction (it should be dropping 49 samples not inserting 49)? Thanks for any help, Deron ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Regarding RTP over TCP without RTSP (for receiving in FFmpeg)
Hi, let me please a basic question that I cannot resolve by reading the documentation. FFmpeg can consume audio/video from a RTP input over UDP protocol. And it can also consume a RTSP stream. At the end, RTSP is a control protocol (over TCP) which negotiates the media by means of a SDP. Audio/video can be transmitted: 1) By using a separate UDP "connection" to send RTP packets. 2) By sharing the TCP connection for both, control protocol and RTP. In this case each RTP packet is encapsulated by prepending it with some bytes as follows: | magic number | channel number | embedded data length | RTP packet | And it seems that FFmpeg supports both methods 1 and 2 above. My specific questions are the following: a) Can FFmpeg receive RTP over TCP **without** RTSP? I mean: just RTP packets over a TCP connection, encapsulated as above in RTSP or by using RFC 4571 "Framing RTP over TCP" [1] (which just requires prepending two bytes indicating the packet length). b) If the answer to a) is "yes", can FFmpeg behave as TCP client (and connect to the remote endpoint from which it will receive RTP over TCP)? or must it behave as TCP server for receiving RTP? Thanks a lot. [1] https://tools.ietf.org/html/rfc4571 -- Iñaki Baz Castillo ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
