Re: [FFmpeg-user] Unable to capture audio and video synced

2017-05-23 Thread Leonardo Soares Müller 
Em 23/05/2017 17:16, Carl Eugen Hoyos escreveu:
> 2017-05-23 19:11 GMT+02:00 Leonardo Soares Müller :
>> To record speakers and microphone, I use pulse.
> (If there really is an issue, it is unclear for me from your report:)
> Please also test alsa if possible.
While I can record the microphone without (apparent) problems, if I try 
to record the speakers sound with alsa I hear nothingwhile it is being 
recorded.

The pulse recording has more problems: after I started the recording, if 
I open an application with a sound stream (then visible on pavucontrol), 
then all the sound gets delayed. This is not observed with parec and 
kazam (which, apparently, uses GStreamer).This is very noticeable with 
applications that use SDL, for example, mednafen. Not only the delay 
increases, but the sound gets distorted too, as if started to play 
faster due to the delay.

I have just downloaded and built ffmpeg snapshot. Then I recorded the 
screen using the following command:

env PULSE_LATENCY_MSEC=20 ffmpeg -vaapi_device /dev/dri/renderD128 
-hwaccel vaapi -hwaccel_output_format yuv420p -thread_queue_size 16384 
-f pulse -sample_rate 44100 -channels 2 -i 
alsa_output.pci-_00_1f.3.analog-stereo.monitor -thread_queue_size 
16384 -f x11grab -s 1366x768 -framerate 30 -i :0.0 -acodec libfdk_aac 
-b:a 160k -vf format=nv12,hwupload,scale_vaapi=w=1280:h=720 -vcodec 
h264_vaapi -qp 20 -f flv -shortest SDL000.flv

Soon after I started the recording, I opened mednafen and then some 
messages started to appear. The output had messages like "Past duration 
0.880180 too large". Here is it complete:

ffmpeg version N-86241-gfb75ad7 Copyright (c) 2000-2017 the FFmpeg 
developers
   built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
   configuration: --enable-shared --enable-avresample --enable-avisynth 
--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray 
--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite 
--enable-libfontconfig --enable-libfreetype --enable-libfribidi 
--enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp 
--enable-libschroedinger --enable-libshine --enable-libsnappy 
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora 
--enable-libtwolame --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid 
--enable-libzvbi --enable-openal --enable-opengl --enable-libdc1394 
--enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 
--enable-libopencv --enable-libfdk-aac --enable-libmfx --enable-vaapi 
--enable-nonfree --enable-gpl --enable-libxcb --enable-libxcb-shm 
--enable-libxcb-xfixes --enable-libxcb-shape --enable-ffplay
   libavutil  55. 63.100 / 55. 63.100
   libavcodec 57. 96.101 / 57. 96.101
   libavformat57. 72.101 / 57. 72.101
   libavdevice57.  7.100 / 57.  7.100
   libavfilter 6. 90.100 /  6. 90.100
   libavresample   3.  6.  0 /  3.  6.  0
   libswscale  4.  7.101 /  4.  7.101
   libswresample   2.  8.100 /  2.  8.100
   libpostproc54.  6.100 / 54.  6.100
libva info: VA-API version 0.39.0
libva info: va_getDriverName() returns 0
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/i965_drv_video.so
libva info: Found init function __vaDriverInit_0_39
libva info: va_openDriver() returns 0
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from 'alsa_output.pci-_00_1f.3.analog-stereo.monitor':
   Duration: N/A, start: 1495580627.717112, bitrate: 1411 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
[x11grab @ 0xa07240] Stream #0: not enough frames to estimate rate; 
consider increasing probesize
Input #1, x11grab, from ':0.0':
   Duration: N/A, start: 1495580628.116939, bitrate: N/A
 Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 
30 fps, 1000k tbr, 1000k tbn, 1000k tbc
Stream mapping:
   Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (h264_vaapi))
   Stream #0:0 -> #0:1 (pcm_s16le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
[swscaler @ 0xa21100] Warning: data is not aligned! This can lead to a 
speedloss
Output #0, flv, to 'SDL000.flv':
   Metadata:
 encoder : Lavf57.72.101
 Stream #0:0: Video: h264 (h264_vaapi) (High) ([7][0][0][0] / 
0x0007), vaapi_vld(progressive), 1280x720, q=0-31, 30 fps, 1k tbn, 30 tbc
 Metadata:
   encoder : Lavc57.96.101 h264_vaapi
 Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 
44100 Hz, stereo, s16, 160 kb/s
 Metadata:
   encoder : Lavc57.96.101 libfdk_aac
frame=   17 fps=0.0 q=-0.0 size= 591kB time=00:00:00.44 
bitrate=10924.6kbits/s speed=0.87frame=   32 fps= 31 q=-0.0 size= 
687kB time=00:00:00.94 bitrate=5975.8kbits/s speed=0.915frame=   48 fps= 
31 q=-0.0 size= 796kB time=00:00:01.47 bitrate=4432.3kbits/s 
speed=0.958frame=   63 fps= 31

Re: [FFmpeg-user] Transmitting Sliced-I frames with ffmpeg

2017-05-23 Thread Carl Eugen Hoyos 
2017-05-23 13:50 GMT+02:00 Simon Brown :

>  ~/ffmpeg/ffmpeg -i udp://127.0.0.1:65111

If the issue is reproducible with current FFmpeg git head,
please dump the udp stream to a file (with tcpdump or
similar) and provide a sample for testing.

Carl Eugen
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Re: [FFmpeg-user] Unable to capture audio and video synced

2017-05-23 Thread Carl Eugen Hoyos 
2017-05-23 19:11 GMT+02:00 Leonardo Soares Müller :
> To record speakers and microphone, I use pulse.

(If there really is an issue, it is unclear for me from your report:)
Please also test alsa if possible.

> The command I used to record was:
>
> env PULSE_LATENCY_MSEC=20 ffmpeg -vaapi_device /dev/dri/renderD128
> -hwaccel vaapi -hwaccel_output_format yuv420p -thread_queue_size 16384
> -f pulse -sample_rate 44100 -channels 2 -i
> alsa_output.pci-_00_1f.3.analog-stereo.monitor -thread_queue_size

> 16384 -f x11grab -s 1366x768 -r 30 -i :0.0 -acodec libfdk_aac -b:a 160k

Remove -r 30, the correct option is "framerate" iirc.

> -vf format=nv12,hwupload,scale_vaapi=w=960:h=540 -vcodec h264_vaapi
> -qp 20 -f flv -shortest TEST001.flv

[...]

> After I have recorded it, I have separated the video and audio from the
> resulting file in two files

Do you mean you did not test A/V sync on the file that was recorded
with FFmpeg?
You (generally) cannot "separate" a media file and expect A/V sync
to be preserved, for screen recording, a (very large) length difference
between audio and video is expected.

Carl Eugen
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Re: [FFmpeg-user] Capturing without break

2017-05-23 Thread Carl Eugen Hoyos 
2017-05-23 9:24 GMT+02:00 LaHu :

> I'd like to know if there's a way to go on recording even when
> the source will disappear for a while?

FFmpeg is (a library and) a transcoding tool, the specific
recording format "data" is currently broken.

Use mplayer -dumpstream or tcpdump or tools/aviocat
to record streamed media.
(You did not give enough information, so I cannot know
which tool would be the best for your usecase, there
likely are others.)

Carl Eugen
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Re: [FFmpeg-user] ffplay after 20161003 not support subtitle

2017-05-23 Thread Carl Eugen Hoyos 
2017-05-23 21:10 GMT+02:00 Giannis Micros :
> Win32 Version:
>
> ffmpeg-20161006-0212867-win32-static version and before works with ass
> subtitle , can show video window.
>
> After ffmpeg-20161007-c45ba26-win32-static version , can't show video
> window .

Did you compile yourself?
If not, this is likely more a question for the Zeranoe forum.
(FFplay works fine for us, but we compile ourselves.)

Carl Eugen
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Re: [FFmpeg-user] How to download streaming video with FFmpeg?

2017-05-23 Thread Moritz Barsnick 
On Mon, May 22, 2017 at 07:25:14 +0200, Mirco Buttari wrote:
> fail and suggest me: "Found streaming video, download it with FFmpeg
> [1]".

Is that really all it says? Nothing else? No URL?

Please give an example.

Moritz
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Re: [FFmpeg-user] ffplay after 20161003 not support subtitle

2017-05-23 Thread Moritz Barsnick 
On Tue, May 23, 2017 at 22:10:41 +0300, Giannis Micros wrote:
> After ffmpeg-20161007-c45ba26-win32-static version , can't show video
> window .
> 
> Why?

No error messages, no console output, nothing? In other words: Could
you kindly show us your command line and the complete, uncut console
output.

Have you tried the latest versions as well?

Moritz
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Re: [FFmpeg-user] why doesn't fast seeking work in live hls

2017-05-23 Thread Louis Letourneau 
As a follow up, if anyone wants to enable and use fast (index) seeking
in HLS videos while they are being transcoded, the secret is to use
the

-hls_playlist_type event

parameter as mentioned in the doc here:
https://ffmpeg.org/ffmpeg-all.html#toc-Options-38

This feature was added decembre 2015 so it's been in the 3.2 branch
for awhile now.

Louis

On Fri, May 19, 2017 at 5:49 PM, Louis Letourneau  wrote:
> Just a question before I start diving in the code, why doesn't fast
> seeking work on live hls feeds/streams/files?
>
> Context:
> If you create a dummy video:
> mkdir -p a;ffmpeg -y -s 640x480 -f rawvideo -pix_fmt rgb24 -r 25 -i
> /dev/zero -vf 
> "drawtext=fontfile=/usr/share/fonts/truetype/freefont/FreeMono.ttf:
> text=%{n}: x=(w-tw)/2: y=h-(2*lh): fontcolor=white: box=1:
> boxcolor=0x00FF" -an -vcodec libx264 -preset medium -tune
> stillimage -crf 24 -pix_fmt yuv420p -shortest -force_key_frames
> "expr:gte(t,n_forced*5)" -bf 0 -hls_time 5 -hls_list_size 0 -hls_wrap
> 0 -hls_allow_cache 1 -hls_segment_filename "a/a_%04d.ts" -t 
> a/a.m3u8
>
> let it encode
> and in another terminal try to seek (once the encoding passed the time
> you want to seek to)
>
> ffmpeg -v info -ss 00:00:10 -y -live_start_index 0 -i a/a.m3u8 -f
> image2 -vframes 1 a.png ; feh a.png
>
> You'll get in the logs
> a/a.m3u8: could not seek to position 11.400
>
> and then ffmpeg will decode all frames until the asked frame.
>
> If you quit the encoder and seek again, it fast seeks immediately
> (jumps to the right segment and decodes te right frame). It doesn't
> need to fo through all the segments.
>
> My question is, if the segments (.ts) are already there and the
> manifest, m3u8 already shows that they are there, why doesn't fast
> seeking work?
>
> Thanks
> Louis
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[FFmpeg-user] How to download streaming video with FFmpeg?

2017-05-23 Thread Mirco Buttari 
  Hi, traying to download streraming video with http://pasty.link/
it
fail and suggest me: "Found streaming video, download it with FFmpeg
[1]".

I download FFmpeg files but now I need instruction, what can i do
now?
Please let me know
Tnx Mirco
  


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[FFmpeg-user] ffplay after 20161003 not support subtitle

2017-05-23 Thread Giannis Micros 
Win32 Version:

ffmpeg-20161006-0212867-win32-static version and before works with ass
subtitle , can show video window.

After ffmpeg-20161007-c45ba26-win32-static version , can't show video
window .

Why?
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[FFmpeg-user] Unable to capture audio and video synced

2017-05-23 Thread Leonardo Soares Müller 
Hello

I am trying to record my computer's screen together with the sound from 
speakers and (occasionally) microphone. While the record is made 
successfully, it is problematic: when watching on VLC it is good, but on 
Kdenlive or YouTube audio and video are out of sync.

I am using Xubuntu 16.04.2 and I have downloaded ffmpeg source and built 
it. To record the screen, I use x11grab. To record speakers and 
microphone, I use pulse. The command I used to record was:

env PULSE_LATENCY_MSEC=20 ffmpeg -vaapi_device /dev/dri/renderD128 
-hwaccel vaapi -hwaccel_output_format yuv420p -thread_queue_size 16384 
-f pulse -sample_rate 44100 -channels 2 -i 
alsa_output.pci-_00_1f.3.analog-stereo.monitor -thread_queue_size 
16384 -f x11grab -s 1366x768 -r 30 -i :0.0 -acodec libfdk_aac -b:a 160k 
-vf format=nv12,hwupload,scale_vaapi=w=960:h=540 -vcodec h264_vaapi -qp 
20 -f flv -shortest TEST001.flv

The output of the command was:

ffmpeg version N-86126-ge434840 Copyright (c) 2000-2017 the FFmpeg 
developers
   built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
   configuration: --enable-shared --enable-avresample --enable-avisynth 
--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray 
--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite 
--enable-libfontconfig --enable-libfreetype --enable-libfribidi 
--enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame 
--enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp 
--enable-libschroedinger --enable-libshine --enable-libsnappy 
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora 
--enable-libtwolame --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid 
--enable-libzvbi --enable-openal --enable-opengl --enable-libdc1394 
--enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 
--enable-libopencv --enable-libfdk-aac --enable-libmfx --enable-vaapi 
--enable-nonfree --enable-gpl --enable-libxcb --enable-libxcb-shm 
--enable-libxcb-xfixes --enable-libxcb-shape
   libavutil  55. 63.100 / 55. 63.100
   libavcodec 57. 96.101 / 57. 96.101
   libavformat57. 72.101 / 57. 72.101
   libavdevice57.  7.100 / 57.  7.100
   libavfilter 6. 90.100 /  6. 90.100
   libavresample   3.  6.  0 /  3.  6.  0
   libswscale  4.  7.101 /  4.  7.101
   libswresample   2.  8.100 /  2.  8.100
   libpostproc54.  6.100 / 54.  6.100
libva info: VA-API version 0.39.0
libva info: va_getDriverName() returns 0
libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/i965_drv_video.so
libva info: Found init function __vaDriverInit_0_39
libva info: va_openDriver() returns 0
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from 'alsa_output.pci-_00_1f.3.analog-stereo.monitor':
   Duration: N/A, start: 149903.842161, bitrate: 1411 kb/s
 Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
[x11grab @ 0x18612e0] Stream #0: not enough frames to estimate rate; 
consider increasing probesize
Input #1, x11grab, from ':0.0':
   Duration: N/A, start: 149904.112324, bitrate: N/A
 Stream #1:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 1366x768, 
30 fps, 1000k tbr, 1000k tbn, 1000k tbc
Stream mapping:
   Stream #1:0 -> #0:0 (rawvideo (native) -> h264 (h264_vaapi))
   Stream #0:0 -> #0:1 (pcm_s16le (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
[swscaler @ 0x187b200] Warning: data is not aligned! This can lead to a 
speedloss
Output #0, flv, to 'TEST001.flv':
   Metadata:
 encoder : Lavf57.72.101
 Stream #0:0: Video: h264 (h264_vaapi) (High) ([7][0][0][0] / 
0x0007), vaapi_vld(progressive), 960x540, q=0-31, 30 fps, 1k tbn, 30 tbc
 Metadata:
   encoder : Lavc57.96.101 h264_vaapi
 Stream #0:1: Audio: aac (libfdk_aac) ([10][0][0][0] / 0x000A), 
44100 Hz, stereo, s16, 160 kb/s
 Metadata:
   encoder : Lavc57.96.101 libfdk_aac
frame=  681 fps= 30 q=-0.0 Lsize=2607kB time=00:00:22.64 bitrate= 
943.1kbits/s speed=0.999x
video:2134kB audio:443kB subtitle:0kB other streams:0kB global 
headers:0kB muxing overhead: 1.161332%

After I have recorded it, I have separated the video and audio from the 
resulting file in two files (the input being the file, using -an -vcodec 
copy and -vn acodec copy, for the respective outputs). The video only 
file was opened on Kdenlive and the audio file was opened on Audacity.

Looking at the command, it would be expected a difference of 20 ms (0,02 
s) between audio and video at maximum, but the length's difference was 
460 ms (0,46 s). There are three different lengths for this file and, 
possibly, different multimedia implementations see the file with a 
different lengths.

ffmpeg (from command line): 22,64 s
Kdenlive: 22,22 s
Audacity: 22,686 s

The millisecond is only visible using Audacity, so I don't know the 
milliseconds from ffmpeg and Kdenlive.

I have done this test some times,

Re: [FFmpeg-user] IP camera stream video with audio

2017-05-23 Thread Moritz Barsnick 
On Tue, May 23, 2017 at 12:17:33 +0200, Gabor Alsecz wrote:
> I can grab video but not audio? What should i do here?
> I opened stream - of Bosch - with ffplay like this:
> ffplay rtsp://192.168.1.105/rtsp_tunnel

Peculiar. This site does say audio is "possible":
https://www.ispyconnect.com/man.aspx?n=Bosch
Are you sure it's also enabled for output (I don't know anything about
the Bosch controls)?

Have you tried
rtsp://192.168.1.105/h264
?

Have you tried using VLC?

Cheers,
Moritz
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[FFmpeg-user] Transmitting Sliced-I frames with ffmpeg

2017-05-23 Thread Simon Brown 
I have an encoder which gives me the option of generating sliced I frames.
These frames have a slice of I frame inserted in each frame, but there is
never a full I frame.  If I use FFMpeg to rebroadcast the stream to an HLS
stream, then it works well when I'm not using sliced I frames, but only
transmits audio when there are sliced I frames in the incoming stream.  Is
there any way of persuading it to accept sliced I frames?

I know I'm using an old version of ffmpeg, but it's built for an embedded
processor, and I've used different options and modified some examples for a
different application.  The result is that I REALLY don't want to have to
rebuild with the latest code, but if someone can assure me that the latest
build DOES support sliced I frame I will do that - I don't want to take a
gamble though.

 ~/ffmpeg/ffmpeg -i udp://127.0.0.1:65111 -bsf:v h264_mp4toannexb -hls_time
0.4 -hls_list_size 20 -hls_flags delete_segments+split_by_time -codec copy
browser.m3u8
ffmpeg version N-81696-gd38dff8e Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 4.9.3 (Linaro GCC 4.9-2014.11) 20141031 (prerelease)
  configuration: --disable-decoders --enable-decoder=h264
--enable-decoder=vc1 --enable-decoder=aac --disable-ffplay
--disable-ffprobe --disable-ffserver --enable-neon
  libavutil  55. 29.100 / 55. 29.100
  libavcodec 57. 57.100 / 57. 57.100
  libavformat57. 49.100 / 57. 49.100
  libavdevice57.  0.102 / 57.  0.102
  libavfilter 6. 62.100 /  6. 62.100
  libswscale  4.  1.100 /  4.  1.100
  libswresample   2.  1.100 /  2.  1.100
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
   Last message repeated 1 times
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] no frame!
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
Last message repeated 1 times
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] non-existing PPS 0 referenced
[h264 @ 0xbc4a30] decode_slice_header error
[h264 @ 0xbc4a30] no frame!
Input #0, mpegts, from 'udp://127.0.0.1:65111':
  Duration: N/A, start: 10874.683144, bitrate: N/A
  Program 1
Metadata:
  service_name: PROGRAM 001
  service_provider: SVP NETWORK
Stream #0:0[0x1100]: Video: h264 (Main) ([27][0][0][0] / 0x001B),
yuv420p, 1280x720 [SAR 1:1 DAR 16:9], 60 fps, 60 tbr, 90k tbn, 120 tbc
Stream #0:1[0x1110]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 109 kb/s
[hls @ 0xc2f540] Using AVStream.codec to pass codec parameters to muxers is
deprecated, use AVStream.codecpar instead.
Last message repeated 1 times
Output #0, hls, to '/home/root/studio_web/myapp/public/images/browser.m3u8':
  Metadata:
encoder : Lavf57.49.100
Stream #0:0: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p,
1280x720 [SAR 1:1 DAR 16:9], q=2-31, 60 fps, 60 tbr, 90k tbn, 60 tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz,
stereo, 109 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)

So it clearly sees the H264 stream, but never copies any frames.

Cheers,
Simon
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Re: [FFmpeg-user] Images to video under Windows

2017-05-23 Thread Gyan 
On Tue, May 23, 2017 at 4:27 PM, Wolfgang Hugemann  wrote:

>
> In regard to the other comment: I knew that Cygwin is an option, but my
> main intent is to completely automise standard operations, such that all my
> colleagues can perform them without thinking about what happens exactly.
> Installing Cygwin on each client computer in our network would make things
> more complicated.
>

The GOW project at https://github.com/bmatzelle/gow installs native win32
ports of the common *nix utilities, including cat. Far lighter than Cygwin.
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Re: [FFmpeg-user] Images to video under Windows

2017-05-23 Thread Wolfgang Hugemann 
> This works for me in the cmd shell on Win7:
> type *.jpg | ffmpeg -f image2pipe -i - {encoding options) output.ext
> In any case, dir /b produces a listing. cat/type emit the data of the
> operands.

Thanks. This command line does the job. My problem was that I did not really 
know what exactly 'cat' does.

In regard to the other comment: I knew that Cygwin is an option, but my main 
intent is to completely automise standard operations, such that all my 
colleagues can perform them without thinking about what happens exactly. 
Installing Cygwin on each client computer in our network would make things more 
complicated.

Thanks for your hints!
Wolfgang
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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

2017-05-23 Thread Christian Ebert 
* Mark Burton on Monday, May 22, 2017 at 15:22:34 +0100
> On 15 Apr 2017, at 09:22, Christian Ebert  wrote:
>> Somewhat counterintuitive, but you never know:
>> 
>> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+1024
>> 
>> combined with the -t incantation.
> 
> It seems this issue is not going to garner much attention which is a little 
> disheartening. I can show how to reproduce it and would love to be able to 
> help.
> 
> So I looked back at your above -af and realised that the 1024 should actually 
> be 2112 which is Apple’s chosen fixed encoding delay.
> https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
>  
> 
> 
> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+2112

Strange, probably all depends on the demuxing application.

ffprobe -show_entries stream=codec_type,start_time,start_pts,duration result

for me gives 0 for all start values.

> ...brings the native aac encoder almost perfectly into sync when played by a 
> Quicktime based decoder. There is a tiny discrepancy, but its 99.9% better 
> than without the -af line.
> 
> Further to this, using the AudioToolbox AAC encoder (aac_at) available in 
> ffmpeg on macOS only, with the above -af line, this discrepancy is gone and 
> the encoded file is a perfect sync match for the original source file.
> 
> The outstanding issue is the remaining samples for the file which are not 
> being trimmed, so the clip runs past the end of the picture and we get a 
> black frame. Perhaps the remaining samples are not being flagged in a way 
> that the decoder would expect, I’m really not sure.
> 
> I can use the '-t' command with the value for ('total duration of source' - 
> ‘0.041') to trim the end, but its has issues too.
> In my case of 24fps source, 0.041 is the duration of 1 frame. Doing this 
> shortens the overlong audio stream, but it removes the last frame of audio 
> and in some cases does strange things with the last two frames of audio. So 
> its not really usable in a production environment. Without subtracting 0.041, 
> the audio is still overlong.

Did you look at the atrim filter? It offers end, end_sample, and
end_pts parameters. You may have to to some calculations.

Re-reading the article you referenced, it may even a better idea
to use atrim instead of asetpts for the start as well.

Maybe:

-filter:a aresample=async=1:first_pts=0,atrim=start_pts=2112:end={-t value}

-- 
\black\trash movie   _SAME  TIME  SAME  PLACE_
   New York, in the summer of 2001

--->> https://blacktrash.org/underdogma/stsp.php
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Re: [FFmpeg-user] upscaling a single DASH segment

2017-05-23 Thread Qian Li 

Hello again,

The problem is now solved. The previous init.mp4 was corrupted. I tried 
to re-download it and re-cat the files, then the upscaling returned no 
errors.


Best regards,
Qian

On 2017-05-23 12:12, Qian Li wrote:

On 2017-05-23 04:09, Steven Liu wrote:

2017-05-22 17:26 GMT+08:00 Qian Li :


Hello,


I am a newbie of FFmpeg. I am trying to upscale a single video 
segment
(downloaded by a DASH client from a DASH server) to a higher 
resolution. I

tried the copy and scale filters, but both of them gave the following
errors:


[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] could not find 
corresponding

track id 1
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] could not find 
corresponding

trex
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] error reading header
BigBuckBunny_4s1.m4s: Invalid data found when processing input

The following is the meta data of BigBuckBunny_4s1.m4s generated by 
mp4dump


?[styp] size=8+16
[sidx] size=12+32
  reference_ID = 1
  timescale = 96
  earliest_presentation_time = 0
  first_offset = 0
[moof] size=8+464
  [mfhd] size=12+4
sequence number = 1
  [traf] size=8+440
[tfhd] size=12+4, flags=2
  track ID = 1
[tfdt] size=12+4
  base media decode time = 0
[trun] size=12+396, flags=205
  sample count = 96
  data offset = 480
  first sample flags = 0
[mdat] size=8+21112

The following is the meta data of the init.mp4 file
[ftyp] size=8+20
  major_brand = iso5
  minor_version = 1
  compatible_brand = avc1
  compatible_brand = dash
  compatible_brand = iso5
[free] size=8+50
[moov] size=8+783
...

Could you give me a hint on what is wrong and how to upscale a single 
DASH

segment?



which module do you using? ffmpeg dash muxer? or other?


Hi,

Thank you very much for the reply.

I installed ffmpeg 3.3-1~16.04.york1 for Ubuntu 16.04 from PPA. I
think it includes all the libraries and command line tools.

I concatenated the initialization file init.mp4 with one of the
segments seg1.m4s to new_seg.mp4 using cat. With the help of mp4dump,
I could see that all the required information (e.g. [trak] and [trex])
presented in new_seg.mp4.

However, I still got the same errors when I was upconverting
new_seg.mp4. The following is the errors returned:

command: ffmpeg -i new_seg.mp4 -vf scale=1920:1080 seg_1080.mp4

errors:
ffmpeg version 3.3-1~16.04.york1 Copyright (c) 2000-2017 the FFmpeg 
developers

  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --prefix=/usr --extra-version='1~16.04.york1'
--toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl
--disable-stripping --enable-avresample --enable-avisynth
--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray
--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libmp3lame
--enable-libopenjpeg --enable-libopenmpt --enable-libopus
--enable-libpulse --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh
--enable-libtheora --enable-libtwolame --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265
--enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx
--enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394
--enable-libiec61883 --enable-chromaprint --enable-frei0r
--enable-libopencv --enable-libx264 --enable-shared

  WARNING: library configuration mismatch

  avcodec configuration: --prefix=/usr
--extra-version='1~16.04.york1' --toolchain=hardened
--libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl
--disable-stripping --enable-avresample --enable-avisynth
--enable-gnutls --enable-ladspa --enable-libass --enable-libbluray
--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite
--enable-libfontconfig --enable-libfreetype --enable-libfribidi
--enable-libgme --enable-libgsm --enable-libmp3lame
--enable-libopenjpeg --enable-libopenmpt --enable-libopus
--enable-libpulse --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh
--enable-libtheora --enable-libtwolame --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265
--enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx
--enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394
--enable-libiec61883 --enable-chromaprint --enable-frei0r
--enable-libopencv --enable-libx264 --enable-shared --enable-version3
--disable-doc --disable-programs --enable-libopencore_amrnb
--enable-libopencore_amrwb --enable-libtesseract
--enable-libvo_amrwbenc --enable-netcdf
  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libavresample   3.

[FFmpeg-user] IP camera stream video with audio

2017-05-23 Thread Gabor Alsecz 
Dear All,

I have 2 types of IP cameras (Bosch Dinion UHD, Samsung PNO9080) and both
capable of audio streaming alongside with video when attach MIC to it. On
it's own web server showing audio has input signal so should work
theoretically.
I can grab video but not audio? What should i do here?
I opened stream - of Bosch - with ffplay like this:
ffplay rtsp://192.168.1.105/rtsp_tunnel

Console output:
C:\Users\admin>ffplay rtsp://192.168.1.105/rtsp_tunnel
ffplay version N-83585-ga5c1c7a Copyright (c) 2003-2017 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-cuda
--enable-cuvid --e
nable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc
--enable-avisynth --
enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enable-iconv
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-li
bfreetype --enable-libgme --enable-libgsm --enable-libilbc
--enable-libmodplug -
-enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enabl
e-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-li
bsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
--enable-libtwolam
e --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx
 --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
--enable
-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib
  libavutil  55. 47.100 / 55. 47.100
  libavcodec 57. 80.101 / 57. 80.101
  libavformat57. 66.102 / 57. 66.102
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 73.100 /  6. 73.100
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
[udp @ 01ce9260] 'circular_buffer_size' option was set but it is
not sup
ported on this build (pthread support is required)
[udp @ 01cf94e0] 'circular_buffer_size' option was set but it is
not sup
ported on this build (pthread support is required)
Input #0, rtsp, from 'rtsp://192.168.1.105/rtsp_tunnel': 0B f=0/0
  Metadata:
title   : LIVE VIEW
  Duration: N/A, start: 0.04, bitrate: N/A
Stream #0:0: Video: h264 (Main), yuv420p(tv, bt470bg/bt709/bt709,
progressiv
e), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 180k tbc
  26.75 M-V:  0.015 fd=   8 aq=0KB vq=  792KB sq=0B f=0/0


So visible no audio stream at all and i have no clue how can i get it.
Thanks!

BR,
Gabor
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Re: [FFmpeg-user] upscaling a single DASH segment

2017-05-23 Thread Qian Li 

On 2017-05-23 04:09, Steven Liu wrote:

2017-05-22 17:26 GMT+08:00 Qian Li :


Hello,


I am a newbie of FFmpeg. I am trying to upscale a single video segment
(downloaded by a DASH client from a DASH server) to a higher 
resolution. I

tried the copy and scale filters, but both of them gave the following
errors:


[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] could not find 
corresponding

track id 1
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] could not find 
corresponding

trex
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x55f1d6eeb8e0] error reading header
BigBuckBunny_4s1.m4s: Invalid data found when processing input

The following is the meta data of BigBuckBunny_4s1.m4s generated by 
mp4dump


?[styp] size=8+16
[sidx] size=12+32
  reference_ID = 1
  timescale = 96
  earliest_presentation_time = 0
  first_offset = 0
[moof] size=8+464
  [mfhd] size=12+4
sequence number = 1
  [traf] size=8+440
[tfhd] size=12+4, flags=2
  track ID = 1
[tfdt] size=12+4
  base media decode time = 0
[trun] size=12+396, flags=205
  sample count = 96
  data offset = 480
  first sample flags = 0
[mdat] size=8+21112

The following is the meta data of the init.mp4 file
[ftyp] size=8+20
  major_brand = iso5
  minor_version = 1
  compatible_brand = avc1
  compatible_brand = dash
  compatible_brand = iso5
[free] size=8+50
[moov] size=8+783
...

Could you give me a hint on what is wrong and how to upscale a single 
DASH

segment?



which module do you using? ffmpeg dash muxer? or other?


Hi,

Thank you very much for the reply.

I installed ffmpeg 3.3-1~16.04.york1 for Ubuntu 16.04 from PPA. I think 
it includes all the libraries and command line tools.


I concatenated the initialization file init.mp4 with one of the segments 
seg1.m4s to new_seg.mp4 using cat. With the help of mp4dump, I could see 
that all the required information (e.g. [trak] and [trex]) presented in 
new_seg.mp4.


However, I still got the same errors when I was upconverting 
new_seg.mp4. The following is the errors returned:


command: ffmpeg -i new_seg.mp4 -vf scale=1920:1080 seg_1080.mp4

errors:
ffmpeg version 3.3-1~16.04.york1 Copyright (c) 2000-2017 the FFmpeg 
developers

  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --prefix=/usr --extra-version='1~16.04.york1' 
--toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu 
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping 
--enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa 
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca 
--enable-libcdio --enable-libflite --enable-libfontconfig 
--enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm 
--enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt 
--enable-libopus --enable-libpulse --enable-librubberband 
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex 
--enable-libssh --enable-libtheora --enable-libtwolame 
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp 
--enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi 
--enable-omx --enable-openal --enable-opengl --enable-sdl2 
--enable-libdc1394 --enable-libiec61883 --enable-chromaprint 
--enable-frei0r --enable-libopencv --enable-libx264 --enable-shared


  WARNING: library configuration mismatch

  avcodec configuration: --prefix=/usr 
--extra-version='1~16.04.york1' --toolchain=hardened 
--libdir=/usr/lib/x86_64-linux-gnu 
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping 
--enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa 
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca 
--enable-libcdio --enable-libflite --enable-libfontconfig 
--enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm 
--enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt 
--enable-libopus --enable-libpulse --enable-librubberband 
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex 
--enable-libssh --enable-libtheora --enable-libtwolame 
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp 
--enable-libx265 --enable-libxvid --enable-libzmq --enable-libzvbi 
--enable-omx --enable-openal --enable-opengl --enable-sdl2 
--enable-libdc1394 --enable-libiec61883 --enable-chromaprint 
--enable-frei0r --enable-libopencv --enable-libx264 --enable-shared 
--enable-version3 --disable-doc --disable-programs 
--enable-libopencore_amrnb --enable-libopencore_amrwb 
--enable-libtesseract --enable-libvo_amrwbenc --enable-netcdf

  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libavresample   3.  5.  0 /  3.  5.  0
  libswscale  4.  6.100 /  4.  6.100
  libswresample   2.  7.100 /  2.  7.100
  libpostproc54.  5.100 / 54.  5.100

[mov,mp4,m4a,3gp,3g2,mj2 @

Re: [FFmpeg-user] Images to video under Windows

2017-05-23 Thread Gyan 
On Tue, May 23, 2017 at 2:05 PM, Wolfgang Hugemann  wrote:

> > e.g.   cat *.jpg | ffmpeg -f image2pipe -framerate 25 -i - out.mp4
>
> Does piping really work under Windows? The Windows equivalent to 'cat'
> would be 'dir /b'. But changing the command line this way creates an error
> that basically says that the input stream is empty.
>
> I couldn't find any example for input piping with ffmpeg on Windows. Are
> you sure that it functions at all?
>
>
This works for me in the cmd shell on Win7:

type *.jpg | ffmpeg -f image2pipe -i - {encoding options) output.ext

In any case, dir /b produces a listing. cat/type emit the data of the
operands.
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Re: [FFmpeg-user] Images to video under Windows

2017-05-23 Thread Reindl Harald 



Am 23.05.2017 um 10:35 schrieb Wolfgang Hugemann:

e.g.   cat *.jpg | ffmpeg -f image2pipe -framerate 25 -i - out.mp4


Does piping really work under Windows? The Windows equivalent to 'cat' would be 
'dir /b'. But changing the command line this way creates an error that 
basically says that the input stream is empty.

I couldn't find any example for input piping with ffmpeg on Windows. Are you 
sure that it functions at all?

BTW: Is there any source of information on the specifics and limitations when 
using ffmpeg under Windows? The situation seems to be a little bit like that 
for ImageMagick, where I had to write that kind of webpage myself 
(http://www.imagemagick.org/Usage/windows)


well, the limitations of Windows are not really ffmpeg relevant itself
https://www.cygwin.com/

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Re: [FFmpeg-user] Images to video under Windows

2017-05-23 Thread Wolfgang Hugemann 
> e.g.   cat *.jpg | ffmpeg -f image2pipe -framerate 25 -i - out.mp4

Does piping really work under Windows? The Windows equivalent to 'cat' would be 
'dir /b'. But changing the command line this way creates an error that 
basically says that the input stream is empty.

I couldn't find any example for input piping with ffmpeg on Windows. Are you 
sure that it functions at all?

BTW: Is there any source of information on the specifics and limitations when 
using ffmpeg under Windows? The situation seems to be a little bit like that 
for ImageMagick, where I had to write that kind of webpage myself 
(http://www.imagemagick.org/Usage/windows).

Wolfgang Hugemann
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[FFmpeg-user] Capturing without break

2017-05-23 Thread LaHu 

Hi,

I'd like to know if there's a way to go on recording even when the 
source will disappear for a while?


Reason:
I plan to record several sources at a time as independent recordings. 
Source A will become Video A, Source B will become Video B and so on.
Source A will be the "master" and the duration of its recording is the 
reference for all other recordings.


If Source A lasts 30 minutes then all other sources should last 30 
minutes as well, no matter if their sources deliver valid signals or not.


It is not really neccessary that the duration of all recordings is 
exactly the same. If there's a loss of signal and the recording stops 
then it is fine enough to automatically start a second recording without 
a signal if the break between ending the first recording and start with 
the next recording takes not longer than one second. I would use then a 
script which puts both streams together.


Thanks in advance!

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