Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Ronak


> On Jun 22, 2018, at 7:59 PM, Ronak Patel  wrote:
> 
> 
>> On Jun 22, 2018, at 5:03 PM, Carl Zwanzig  wrote:
>> 
>>> On 6/22/2018 1:37 PM, Ronak wrote:
>>> We have audio files that are more than 100 hours long, and we need them
>>> to be fragmented quickly. It totally seems like there's an I/O problem
>>> here because fragmentation is faster the longer we set our fragment size
>>> to.
>> Sounds like it's time to break out iostat or sar (or even dtrace) and do 
>> some snooping.
>> 
>> Are you reading and writing to different drives?  (The usual metric for a 
>> spinning drive is 150 IOPS for random access, RAID doesn't necessarily speed 
>> that up.)
>> 
> 
> Nope it’s the same disk that we’re reading and writing to. I’m really curious 
> why we can’t just buffer fragments in ram and write them out in larger 
> bursts. Why not make this an option we can set?
> 
> This would simulate the performance for a 10 sec fragment size.
> 
> I’d imagine you have the same reading buffer regardless of the fragment size.
> 
> I’m thinking of running this on even more powerful machines to see if I can 
> get the times down. But is it possible to get an option to tweak the output 
> buffer size?
> 

The problem is probably because of this:

[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.m4s' for reading
[dash @ 0x7f92cd005600] Opening 'test_60_hr_10sec.mpd.tmp' for writing
[dash @ 0x7f92cd005600] Opening 'media_0.m3u8.tmp' for writing

Ffmpeg constantly makes this tmp files, and seems to be opening and closing the 
file handles constantly.
Is there a way to just keep the main files open and just keep appending to it? 
Why make temp files at all?

>> Later,
>> 
>> z!
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[FFmpeg-user] personalize show freqs (bar)

2018-06-22 Thread fal856
Hi, im new to ffmpeg, and i found the showfreqs filter (
https://ffmpeg.org/ffmpeg-filters.html#showfreqs
  ).

Im using it to make a video out of a song, (ffmpeg -i input.mp3
-filter_complex
"[0:a]showfreqs=s=1920x1080:mode=bar:fscale=log:colors=white,format=yuv420p[v]"
-map "[v]" -map 0:a showfreqs.mp4) but i can find in the docs any way to
"personalize" the bars.

For example, i would like to reduce the ammount of bars, and put some
spacing between each other, and if its posible round the corners of the
bars.

Can someone help me ? 

thnks



--
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[FFmpeg-user] ffmpeg goes 100% cpu

2018-06-22 Thread Octavio Alvarez
Hi,

I created this 5-sec sample video:

ffmpeg -y -video_size 1366x768 -framerate 25 -f x11grab -i :0.0+0,0 -f
pulse -ac 2 -i default -c:v libx264 -qp 0 -preset ultrafast -pix_fmt
yuv420p -t 5 desk5.mkv

And I want to have it tailpadded with black if the video is too short
compared to the -t option. It works fine if I use -t 5.1 (or 6, or
whatever number larger than 5) like this:

ffmpeg -y -i desk5.mkv -f lavfi -i color=black:s=1366x768:r=25
-filter_complex
'[0:v]setpts=PTS-STARTPTS[timed];[timed][1:v]concat=n=2:v=1[vout]' -map
"[vout]" -t 5.1 -preset ultrafast -intra output.mkv

But if I use -t 4.9 (the result should just be the first 4.9 seconds of
the original video) ffmpeg hangs at the end using 100% of one cpu core:

ffmpeg -y -i desk5.mkv -f lavfi -i color=black:s=1366x768:r=25
-filter_complex
'[0:v]setpts=PTS-STARTPTS[timed];[timed][1:v]concat=n=2:v=1[vout]' -map
"[vout]" -t 4.9 -preset ultrafast -intra output.mkv

Here I add -loglevel trace for both and past the last output lines.
First for -t 5.1 (the one that works) and then for -t 4.9 (the one that
doesn't):

For -t 5.1:

[matroska @ 0x5604d38d6540] Writing block at offset 10, size 102160, pts
4720, dts 4720, duration 40, keyframe 1
[rawvideo @ 0x5604d38d6d00] PACKET SIZE: 1573632, STRIDE: 2049
[libx264 @ 0x5604d38d2a00] frame= 111 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=102160 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 11791814
bytes, pts 4760dts 4760
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 102160, pts
4760, dts 4760, duration 40, keyframe 1
[rawvideo @ 0x5604d38d6d00] PACKET SIZE: 1573632, STRIDE: 2049
[libx264 @ 0x5604d38d2a00] frame= 112 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=102160 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 11894004
bytes, pts 4800dts 4800
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 102160, pts
4800, dts 4800, duration 40, keyframe 1
[rawvideo @ 0x5604d38d6d00] PACKET SIZE: 1573632, STRIDE: 2049
[out_0_0 @ 0x5604d3911380] EOF on sink link out_0_0:default.
No more output streams to write to, finishing.
[Parsed_concat_1 @ 0x5604d38d4780] EOF on in1:v0, 0 streams left in segment.
[Parsed_concat_1 @ 0x5604d38d4780] Segment finished at pts=5162759
[libx264 @ 0x5604d38d2a00] frame= 113 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=102160 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 11996194
bytes, pts 4840dts 4840
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 102160, pts
4840, dts 4840, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 114 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=102551 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 12098384
bytes, pts 4880dts 4880
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 102551, pts
4880, dts 4880, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 115 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=99733 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 12200965
bytes, pts 4920dts 4920
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 99733, pts
4920, dts 4920, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 116 QP=28.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=99732 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 12300728
bytes, pts 4960dts 4960
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 99732, pts
4960, dts 4960, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 117 QP=25.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=3131 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 12400490
bytes, pts 5000dts 5000
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 3131, pts
5000, dts 5000, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 118 QP=23.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=3130 bytes
[matroska @ 0x5604d38d6540] Writing block at offset 3148, size 3130, pts
5040, dts 5040, duration 40, keyframe 1
[libx264 @ 0x5604d38d2a00] frame= 119 QP=21.00 NAL=3 Slice:I Poc:0
I:4128 P:0SKIP:0size=3131 bytes
[matroska @ 0x5604d38d6540] Starting new cluster at offset 12406787
bytes, pts 5080dts 5080
[matroska @ 0x5604d38d6540] Writing block at offset 10, size 3131, pts
5080, dts 5080, duration 40, keyframe 1
[matroska @ 0x5604d38d6540] end duration = 5120
[matroska @ 0x5604d38d6540] stream 0 end duration = 5120
frame=  120 fps= 37 q=-1.0 Lsize=   12121kB time=00:00:05.08
bitrate=19543.0kbits/s speed=1.56x
video:12115kB audio:0kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 0.053097%
Input file #0 (desk5.mkv):
  Input stream #0:0 (video): 117 packets read (6242725 bytes); 117
frames decoded;
  Input stream #0:1 (audio): 52 packets read (7290 bytes);
  Total: 169 packets (6250015 bytes) demuxed
Input file #1 (color=black:s=1366x768:r=25):
  Input stream #1:0 (video): 4 packets read (6294528 bytes); 

Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Ronak Patel


Sent from my iPhone

> On Jun 22, 2018, at 5:03 PM, Carl Zwanzig  wrote:
> 
>> On 6/22/2018 1:37 PM, Ronak wrote:
>> We have audio files that are more than 100 hours long, and we need them
>> to be fragmented quickly. It totally seems like there's an I/O problem
>> here because fragmentation is faster the longer we set our fragment size
>> to.
> Sounds like it's time to break out iostat or sar (or even dtrace) and do some 
> snooping.
> 
> Are you reading and writing to different drives?  (The usual metric for a 
> spinning drive is 150 IOPS for random access, RAID doesn't necessarily speed 
> that up.)
> 

Nope it’s the same disk that we’re reading and writing to. I’m really curious 
why we can’t just buffer fragments in ram and write them out in larger bursts. 
Why not make this an option we can set?

This would simulate the performance for a 10 sec fragment size.

I’d imagine you have the same reading buffer regardless of the fragment size.

I’m thinking of running this on even more powerful machines to see if I can get 
the times down. But is it possible to get an option to tweak the output buffer 
size?

> Later,
> 
> z!
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Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Carl Zwanzig

On 6/22/2018 1:37 PM, Ronak wrote:

We have audio files that are more than 100 hours long, and we need them
to be fragmented quickly. It totally seems like there's an I/O problem
here because fragmentation is faster the longer we set our fragment size
to.
Sounds like it's time to break out iostat or sar (or even dtrace) and do 
some snooping.


Are you reading and writing to different drives?  (The usual metric for a 
spinning drive is 150 IOPS for random access, RAID doesn't necessarily speed 
that up.)


Later,

z!
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Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Ronak


> On Jun 22, 2018, at 4:31 PM, Ronak  wrote:
> 
> 
> 
>> On Jun 22, 2018, at 3:59 PM, David Favor  wrote:
>> 
>> I'd say you're looking at a very long transcode time,
>> as your video dump shows...
>> 
>>  Duration: 52:15:29.84, start: 0.00, bitrate: 63 kb/s
>> 
>> So... a 52 hour video... well... better fire up the popcorn
>> maker + settle in for a very long night...
> 
> 
> This is not video, only AAC audio. And there shouldn't be any transcoding 
> here. I'm using -codec copy. We're going from MP4 -> fMP4 here.

We have audio files that are more than 100 hours long, and we need them to be 
fragmented quickly. It totally seems like there's an I/O problem here because 
fragmentation is faster the longer we set our fragment size to.
I don't imagine the cost of creating more fMP4 boxes would tax the CPU and slow 
everything down.

> 
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Re: [FFmpeg-user] Adaptive Streaming between heaac and lcaac

2018-06-22 Thread Ronak


>> Hi,
>> 
>> We are trying to setup an adaptive stream across two heaac streams and 
>> higher order lcaac ones.
>> 
>> So we’d have something like so:
>> 
>> HEAACv2 44khz/32kbps
>> HEAACv2 44/64
>> LCAAC 44/128
>> LCAAC 44/256
>> LCAAC 44/320
>> 
>> I setup a test stream this way and I’m seeing some problems.
>> 
>> Exoplayer is seeing this stream as not being able to support seamless 
>> adaptive switching because the defaultSampleDurations in the HE-AACv2 fMP4's 
>> moof boxes is 2048 while the LC-AAC file has 1024.
>> 
>> Why does HE-AACv2 have a higher sample duration when the sampling rate is 
>> supposed to be 44100, equivalent to the sampling rate of the LC-AAC audio?
>> 
> 
> 
> Has anyone ever tried to do this before? Is this a problem with how ffmpeg is 
> generating the HEAACv2 fmp4 files?
> 
> 


Could anyone please shed any light on why the defaultSampleDurations in moof 
are different between LC & HE-AAC at the same sampling rates?



>> Thanks,
>> 
>> Ronak
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Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Ronak


> On Jun 22, 2018, at 3:59 PM, David Favor  wrote:
> 
> I'd say you're looking at a very long transcode time,
> as your video dump shows...
> 
>   Duration: 52:15:29.84, start: 0.00, bitrate: 63 kb/s
> 
> So... a 52 hour video... well... better fire up the popcorn
> maker + settle in for a very long night...


This is not video, only AAC audio. And there shouldn't be any transcoding here. 
I'm using -codec copy. We're going from MP4 -> fMP4 here.

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Re: [FFmpeg-user] Getting started: how to enable filters

2018-06-22 Thread Lou Logan
On Thu, Jun 21, 2018, at 5:21 PM, Todd Van Horne wrote:
>
> I see, thank you. Can you tell me where I can find instructions for
> installing the filters?
> -Todd

Lazy method is to download a ffmpeg with libvidstab already included:

Linux: https://johnvansickle.com/ffmpeg/
Windows& macOS: https://ffmpeg.zeranoe.com/builds/
macOS: https://evermeet.cx/ffmpeg/
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Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread David Favor

I'd say you're looking at a very long transcode time,
as your video dump shows...

   Duration: 52:15:29.84, start: 0.00, bitrate: 63 kb/s

So... a 52 hour video... well... better fire up the popcorn
maker + settle in for a very long night...
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Re: [FFmpeg-user] fMP4 generation speed

2018-06-22 Thread Ronak


> On Jun 20, 2018, at 4:48 PM, Carl Eugen Hoyos  wrote:
> 
> 2018-06-20 18:37 GMT+02:00, Ronak :
> 
>> That command took the following time:
>> 
>> real 3m19.316s
>> user 3m14.730s
>> sys  0m1.830s
>> 
>> I'm curious about your thoughts on this. With -codec copy, the decoder
>> should not be involved, so it shouldn't be a bottleneck.
> 
> Yes.
> 
>> I'm wondering if the extra file I/O is the culprit here?
> 
> Yes.
> 
> You can now specify "-loglevel verbose" to get some useful
> I/O info at the end of the console output.
> 
> Please remember not to top-post here, Carl Eugen


Hi Carl,

I got more logs, here's what I got:

[dash @ 0x22c3f80] Opening 'bk_potr_06_22_64_1sec_test.m4s' for reading
[file @ 0x2382c80] Setting default whitelist 'file,crypto'
[AVIOContext @ 0x23b9140] Statistics: 4236 bytes read, 1 seeks
[dash @ 0x22c3f80] Representation 0 media segment 69894 written to: 
bk_potr_06_22_64_1sec_test.m4s
[dash @ 0x22c3f80] Opening 'bk_potr_06_22_64_1sec_test.mpd.tmp' for writing
[file @ 0x2382c80] Setting default whitelist 'file,crypto'
[AVIOContext @ 0x23b9140] Statistics: 0 seeks, 15 writeouts
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467754, dts 68162362630
size=N/A time=18:56:02.36 bitrate=N/A speed=9.37x
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467755, dts 68162409070
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467756, dts 68162455510
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467757, dts 68162501950
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467758, dts 68162548390
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467759, dts 68162594830
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x22be580] stream 0, sample 1467760, dts 68162641270
No more output streams to write to, finishing.
[dash @ 0x22c3f80] Opening 'bk_potr_06_22_64_1sec_test.m4s' for reading
[file @ 0x23540c0] Setting default whitelist 'file,crypto'
[AVIOContext @ 0x23b9140] Statistics: 0 bytes read, 1 seeks
[dash @ 0x22c3f80] Representation 0 media segment 69895 written to: 
bk_potr_06_22_64_1sec_test.m4s
[dash @ 0x22c3f80] Opening 'bk_potr_06_22_64_1sec_test.mpd.tmp' for writing
[file @ 0x2382c80] Setting default whitelist 'file,crypto'
[AVIOContext @ 0x23b9140] Statistics: 0 seeks, 15 writeouts
[AVIOContext @ 0x2310e80] Statistics: 0 seeks, 2107 writeouts
size=N/A time=18:56:02.64 bitrate=N/A speed=9.37x
video:0kB audio:522488kB subtitle:0kB other streams:0kB global headers:0kB 
muxing overhead: unknown
Input file #0 (bk_potr_06_22_64.mp4):
  Input stream #0:0 (audio): 1467761 packets read (535027204 bytes); 
  Total: 1467761 packets (535027204 bytes) demuxed
Output file #0 (bk_potr_06_22_64_1sec_test.mpd):
  Output stream #0:0 (audio): 1467761 packets muxed (535027204 bytes); 
  Total: 1467761 packets (535027204 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x22c72c0] Statistics: 541488832 bytes read, 2 seeks

This file took about 6 hours of clock time to get through on a 4 Core Computer 
running at 2.4 Ghz with 16 GB of RAM.

Are there any options for parallelization that we can use so the load is spread 
across the cores? Or does ffmpeg does this already?
If we're strictly I/O bound for this, is it possible to buffer more individual 
fragments in RAM before we write them out to disk?

Ronak

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[FFmpeg-user] Too many packets buffered for output stream 0:0

2018-06-22 Thread Nick Bright
I'm attempting to use ffmpeg to stream from an IP Video Camera to 
YouTube Live, using ffmpeg version N-46347-g649d7ca47-static on CentOS 7 
with x264-0.148-11.20160614gita5e06b9.el7.x86_64.


So far, all of my searches regarding this error have resulted in a 
suggested work around of -max_muxing_queue_size, but I have tried many 
size parameters and the error persists.


My command is:

VBR="1000k"# Bitrate de la vidéo en sortie
FPS="30"   # FPS de la vidéo en sortie
QUAL="medium"  # Preset de qualité FFMPEG
YOUTUBE_URL="rtmp://a.rtmp.youtube.com/live2"  # URL de base RTMP youtube

SOURCE="rtsp://X/videoMain"  # Source UDP (voir les 
annonces SAP)
KEY="X"
ffmpeg \
    -i "$SOURCE" -deinterlace \
    -vcodec libx264 -pix_fmt yuv420p -preset $QUAL -r $FPS -g $(($FPS * 2)) 
-b:v $VBR \
    -acodec libmp3lame -ar 8000 -threads 6 -crf 23 -b:a 128k -bufsize 512k \
    -max_muxing_queue_size 2147483647 \
-f flv "$YOUTUBE_URL/$KEY"

I've attached the full output of the command when attempting to run the 
stream.


--
---
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-  Vice President of Technology   -
-  Valnet -=- We Connect You -=-  -
-  Tel 888-332-1616 x 315 / Fax 620-331-0789  -
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---
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- http://www.valnet.net/services/valnet-vault -
---

ffmpeg version N-46347-g649d7ca47-static https://johnvansickle.com/ffmpeg/  
Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
  configuration: --enable-gpl --enable-version3 --enable-static --disable-debug 
--disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 
--enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray 
--enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf 
--enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband 
--enable-libsoxr --enable-libspeex --enable-libvorbis --enable-libopus 
--enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libxvid --enable-libzimg
  libavutil  56. 18.102 / 56. 18.102
  libavcodec 58. 20.103 / 58. 20.103
  libavformat58. 17.100 / 58. 17.100
  libavdevice58.  4.101 / 58.  4.101
  libavfilter 7. 25.100 /  7. 25.100
  libswscale  5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc55.  2.100 / 55.  2.100
[rtsp @ 0x5804d80] UDP timeout, retrying with TCP
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://X/videoMain':
  Metadata:
title   : IP Camera Video
comment : videoMain
  Duration: N/A, start: 0.044878, bitrate: N/A
Stream #0:0: Video: h264 (Baseline), yuv420p(progressive), 1280x720, 90k 
tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (pcm_mulaw (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
[libx264 @ 0x58635c0] VBV bufsize set but maxrate unspecified, ignored
[libx264 @ 0x58635c0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
[libx264 @ 0x58635c0] profile High, level 3.1
[libx264 @ 0x58635c0] 264 - core 155 r2901 7d0ff22 - H.264/MPEG-4 AVC codec - 
Copyleft 2003-2018 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 
fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 
nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 
b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 
keyint=60 keyint_min=6 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf 
mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Too many packets buffered for output stream 0:0.577014:32:22.77 bitrate=  
-0.0kbits/s dup=0 drop=4 speed=N/A
[libx264 @ 0x58635c0] frame I:1 Avg QP:18.82  size:122122
[libx264 @ 0x58635c0] frame P:2 Avg QP:21.45  size:  5356
[libx264 @ 0x58635c0] frame B:6 Avg QP:30.13  size:  1175
[libx264 @ 0x58635c0] consecutive B-frames: 11.1%  0.0%  0.0% 88.9%
[libx264 @ 0x58635c0] mb I  I16..4:  6.1% 24.4% 69.4%
[libx264 @ 0x58635c0] mb P  I16..4:  0.1%  1.2%  0.6%  P16..4: 17.7%  3.0%  
1.3%  0.0%  0.0%skip:76.1%
[libx264 @ 0x58635c0] mb B  I16..4: 

Re: [FFmpeg-user] x11grab segfault when display is closed

2018-06-22 Thread robertlazarski
On Fri, Jun 22, 2018 at 9:03 AM, Matt Garman 
wrote:

> I'm using a static git build, version N-46353-g40b7e6071-static[1] on
> CentOS 7.3.  This server is used as a remote login launch point (or "jump
> box").  What we'd like to do, for security/auditing purposes, is create
> videos of user sessions.
>
>
I cannot answer the ffmpeg specific question though as a CentOS 7.x user I
use Xvfb a lot. I have it doing important high volume prod stuff at my day
job and its been a workhorse that has rarely been a problem.

Seems like its possible to make videos via Xvfb. I just do screenshots
myself.

https://github.com/lightsofapollo/x-recorder

Regards,
Robert
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[FFmpeg-user] x11grab segfault when display is closed

2018-06-22 Thread Matt Garman
I'm using a static git build, version N-46353-g40b7e6071-static[1] on
CentOS 7.3.  This server is used as a remote login launch point (or "jump
box").  What we'd like to do, for security/auditing purposes, is create
videos of user sessions.

I'm using a command like this:

# ./ffmpeg -an -f x11grab -framerate 10 -video_size 1280x1024 -i :13
/tmp/cap.mp4

If I stop ffmpeg before that display is closed, I get a nice video, exactly
as expected.

However, if ffmpeg is still running when the display (:13 in the above
command) is closed, ffmpeg segfaults.  This results in an unplayable video
file (presumably, ffmpeg doesn't get a chance to clean up and properly
close out the file).

Would this be considered a bug?  Not sure how hard it would be to recognize
that the display is no longer valid, and cleanly exit...

Is there any kind of workaround possible?

Thanks!

[1] # ./ffmpeg -version

ffmpeg version N-46353-g40b7e6071-static https://johnvansickle.com/ffmpeg/
Copyright (c) 2000-2018 the FFmpeg developers

built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516

configuration: --enable-gpl --enable-version3 --enable-static
--disable-debug --disable-ffplay --disable-indev=sndio
--disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-gray --enable-libaom --enable-libfribidi
--enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
--enable-librubberband --enable-libsoxr --enable-libspeex
--enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab
--enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg

libavutil  56. 18.102 / 56. 18.102

libavcodec 58. 20.103 / 58. 20.103

libavformat58. 17.101 / 58. 17.101

libavdevice58.  4.101 / 58.  4.101

libavfilter 7. 25.100 /  7. 25.100

libswscale  5.  2.100 /  5.  2.100

libswresample   3.  2.100 /  3.  2.100

libpostproc55.  2.100 / 55.  2.100
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Re: [FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

2018-06-22 Thread Dragan Randjelovic
Here is an interesting article:

https://avpres.net/FFmpeg/SD_HD.html

as far as scaling and re-sampling algo goes I would invoke avisynth and try
thin raster  with

Spline64Resize(1440,1080)

as it usually gives better Y-PSNR metric.

Try also sinc re-sampler and see if it suits your case. Once material is
with a lot of fractal details I get crisper results with it.

I would also consider colorspace matrix compensation for luma coefficients
like

colormatrix=bt601:bt709

and then your padding formula

pad=1920:1080:240:0

Re-interlacing  might not be necessary after all, even if it goes to
play-out.


Kind Regards.
.

On Fri, Jun 22, 2018 at 3:19 PM, Cley Faye  wrote:

> Le ven. 22 juin 2018 à 15:10, Christoph Gerstbauer <
> christophgerstba...@gmail.com> a écrit :
>
> > >
> > >
> > > ​The "-s" option is incorrect in your case. From the documentation:
> > > > As an output option, this inserts the scale video filter to the end
> of
> > > the corresponding filtergraph. Please use the scale filter directly to
> > > insert it at the beginning or some other place.​
> > >
> > > The scale video filter will take whatever ends up from your filter, and
> > > scale it without respect to aspect ratio.
> > > To do what you want, you probable have to use a complex filter, where
> you
> > > create a secondary black source and paste the output of your current
> > filter
> > > into it using the "overlay" filter.
> > >
> >
> > I understand
> > So i need to generate (after the cropping and deinterlacing to 720x576
> 50p)
> > a filter syntax (within the videofilter) which transforms the 720x576 4:3
> > content to a pillarbox 1050x576 quare pixel video (with black bars at
> > sides) -> and THIS one I can rescale to HD.
> > Correct?
> >
> > if yes, has anyone more experience with that?
> >
>
> ​Roughly. Look at the other answer; it basically does the pillarbox *after*
> the initlal rescale, which is probably a better idea.​
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Re: [FFmpeg-user] ffplay 3.4 errors with Win10

2018-06-22 Thread Jeff Elkins
[h264 @ 01f5df14e5e0] cbp too large (51) at 43 7=0B f=0/0
[h264 @ 01f5df14e5e0] error while decoding MB 43 7
[h264 @ 01f5df14e5e0] concealing 1556 DC, 1556 AC, 1556 MV errors in P
frame

[repeated until closed...]

>Complete, uncut console output missing.
>Carl Eugen

As noted, the three lines quoted scroll forever until the job is closed via
cntrl-c.  I'm not sure what more information you need.

On Fri, Jun 22, 2018 at 9:24 AM Carl Eugen Hoyos  wrote:

> 2018-06-22 5:54 GMT+02:00, Jeff Elkins :
> > I'm using Android's screenrecord to stream the device screen via adb and
> > pipe the
> > output to ffplay 3.4.  It works great using OS X via the command:
> >
> > adb shell screenrecord --bit-rate=16m --output-format=h264 --size
> 800x600 -
> > | ffplay -framerate 60 -framedrop -bufsize 60M -
>
> Complete, uncut console output missing.
>
> Carl Eugen
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Re: [FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

2018-06-22 Thread Cley Faye
Le ven. 22 juin 2018 à 15:10, Christoph Gerstbauer <
christophgerstba...@gmail.com> a écrit :

> >
> >
> > ​The "-s" option is incorrect in your case. From the documentation:
> > > As an output option, this inserts the scale video filter to the end of
> > the corresponding filtergraph. Please use the scale filter directly to
> > insert it at the beginning or some other place.​
> >
> > The scale video filter will take whatever ends up from your filter, and
> > scale it without respect to aspect ratio.
> > To do what you want, you probable have to use a complex filter, where you
> > create a secondary black source and paste the output of your current
> filter
> > into it using the "overlay" filter.
> >
>
> I understand
> So i need to generate (after the cropping and deinterlacing to 720x576 50p)
> a filter syntax (within the videofilter) which transforms the 720x576 4:3
> content to a pillarbox 1050x576 quare pixel video (with black bars at
> sides) -> and THIS one I can rescale to HD.
> Correct?
>
> if yes, has anyone more experience with that?
>

​Roughly. Look at the other answer; it basically does the pillarbox *after*
the initlal rescale, which is probably a better idea.​
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Re: [FFmpeg-user] ffplay 3.4 errors with Win10

2018-06-22 Thread Carl Eugen Hoyos
2018-06-22 5:54 GMT+02:00, Jeff Elkins :
> I'm using Android's screenrecord to stream the device screen via adb and
> pipe the
> output to ffplay 3.4.  It works great using OS X via the command:
>
> adb shell screenrecord --bit-rate=16m --output-format=h264 --size 800x600 -
> | ffplay -framerate 60 -framedrop -bufsize 60M -

Complete, uncut console output missing.

Carl Eugen
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Re: [FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

2018-06-22 Thread Christoph Gerstbauer
>
>
> ​The "-s" option is incorrect in your case. From the documentation:
> > As an output option, this inserts the scale video filter to the end of
> the corresponding filtergraph. Please use the scale filter directly to
> insert it at the beginning or some other place.​
>
> The scale video filter will take whatever ends up from your filter, and
> scale it without respect to aspect ratio.
> To do what you want, you probable have to use a complex filter, where you
> create a secondary black source and paste the output of your current filter
> into it using the "overlay" filter.
>

I understand
So i need to generate (after the cropping and deinterlacing to 720x576 50p)
a filter syntax (within the videofilter) which transforms the 720x576 4:3
content to a pillarbox 1050x576 quare pixel video (with black bars at
sides) -> and THIS one I can rescale to HD.
Correct?

if yes, has anyone more experience with that?

br
Chris
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Re: [FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

2018-06-22 Thread Gyan Doshi



On 22-06-2018 06:13 PM, Christoph Gerstbauer wrote:



CommandLineOutput:

ffmpeg -i "F:\scaler test\_5sec.avi" -vf "crop=720:576:0:32, setfield=tff,
yadif=1:0:0, scale=-1:1080, tinterlace=interleave_top, fieldorder=tff"
-vcodec ffvhuff -an -s 1920x1080 -aspect 16:9 "F:\scaler


Since you're scaling proportionally ("-1:1080"), you need to pad out the 
frame to FHD.


ffmpeg -i in -vf "crop=720:576:0:32,setfield=tff,yadif=1:0:0,
  scale=-1:1080,pad=1920:1080:(ow-iw)/2:(oh-ih)/2,
  tinterlace=interleave_top, fieldorder=tff,
  setsar=1"
   -c:v ffvhuff -an -aspect 16:9   out

Additionally, since your input isn't square-pixels, the following scale 
may be more accurate,


scale=oh*4/3:1080

Regards,
Gyan
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[FFmpeg-user] Upscaling SD 4x3 to FullHD 16x9 - Aspect Ratio problem

2018-06-22 Thread Christoph Gerstbauer
Hello I am trying to upscale a 720x608 (incl VBI) 25i 4:3 source to
1920x1080 25i 16:9.

My deinterlacing and reinterlacing is working, scaling and cropping too.
But I have understanding problem how to use the videofilter regarding
aspect ratios.

My source is 4:3 and the target is 16:9.
For the target I am expecting that I get a PillarBox FullHD image (black
bars left and right) where the original 4:3 aspect ratio of the source is
kept.
PROBLEM: But I get only a stretched 16:9 HD version of the 4:3 original
image. (no black bars at left and right)
See files:
https://www.dropbox.com/s/mgwmiitcpgta0ym/scalertest.zip?dl=0

CommandLineOutput:

ffmpeg -i "F:\scaler test\_5sec.avi" -vf "crop=720:576:0:32, setfield=tff,
yadif=1:0:0, scale=-1:1080, tinterlace=interleave_top, fieldorder=tff"
-vcodec ffvhuff -an -s 1920x1080 -aspect 16:9 "F:\scaler
test\_5sec_fullHDinterlaced.avi"
ffmpeg version 3.4 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 7.2.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-sdl2
--enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv
--enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame
--enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy
--enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp
--enable-libvidstab --enable-libvorbis --enable-cuda --enable-cuvid
--enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
--enable-libmfx
  libavutil  55. 78.100 / 55. 78.100
  libavcodec 57.107.100 / 57.107.100
  libavformat57. 83.100 / 57. 83.100
  libavdevice57. 10.100 / 57. 10.100
  libavfilter 6.107.100 /  6.107.100
  libswscale  4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc54.  7.100 / 54.  7.100
Input #0, avi, from 'F:\scaler test\_5sec.avi':
  Metadata:
encoder : Lavf58.12.100
  Duration: 00:00:05.00, start: 0.00, bitrate: 116115 kb/s
Stream #0:0: Video: ffvhuff (FFVH / 0x48564646), yuv422p, 720x608,
107684 kb/s, SAR 152:135 DAR 4:3, 25 fps, 25 tbr, 25 tbn, 25 tbc
Stream #0:1: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1,
s32 (24 bit), 9216 kb/s
File 'F:\scaler test\_5sec_fullHDinterlaced.avi' already exists. Overwrite
? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (ffvhuff (native) -> ffvhuff (native))
Press [q] to stop, [?] for help
Output #0, avi, to 'F:\scaler test\_5sec_fullHDinterlaced.avi':
  Metadata:
ISFT: Lavf57.83.100
Stream #0:0: Video: ffvhuff (FFVH / 0x48564646), yuv422p, 1920x1080
[SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc
Metadata:
  encoder : Lavc57.107.100 ffvhuff
frame=  125 fps= 53 q=-0.0 Lsize=  200261kB time=00:00:05.00
bitrate=328108.1kbits/s speed=2.14x
video:200253kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.004325%


Best Regards
Christoph Gerstbauer
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Re: [FFmpeg-user] Adaptive Streaming between heaac and lcaac

2018-06-22 Thread Ronak Patel


> On Jun 21, 2018, at 9:02 AM, Ronak Patel  
> wrote:
> 
> Hi,
> 
> We are trying to setup an adaptive stream across two heaac streams and higher 
> order lcaac ones.
> 
> So we’d have something like so:
> 
> HEAACv2 44khz/32kbps
> HEAACv2 44/64
> LCAAC 44/128
> LCAAC 44/256
> LCAAC 44/320
> 
> I setup a test stream this way and I’m seeing some problems.
> 
> Exoplayer is seeing this stream as not being able to support seamless 
> adaptive switching because the defaultSampleDurations in the HE-AACv2 fMP4's 
> moof boxes is 2048 while the LC-AAC file has 1024.
> 
> Why does HE-AACv2 have a higher sample duration when the sampling rate is 
> supposed to be 44100, equivalent to the sampling rate of the LC-AAC audio?
> 


Has anyone ever tried to do this before? Is this a problem with how ffmpeg is 
generating the HEAACv2 fmp4 files?


> Thanks,
> 
> Ronak
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