Re: [FFmpeg-user] Retrieve duration of an m2v >MPEG>2 elementary stream file

2023-03-02 Thread Robert Krüger
On Wed, Mar 1, 2023 at 4:07 PM Ferdi Scholten  wrote:

> Hi,
> > if I invoke ffmpeg on an M2V MPEG-2 elementary stream, I get "N/A" for
> the
> > duration. I guess this is because this file doesn't contain a header that
> > contains the duration. Is it possible to force ffmpeg to parse the entire
> > file to obtain the duration?
> >
> > Full command line and output:
> >
> > ffmpeg -i
> >
> /Users/krueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v
> >
> > ffmpeg version N-109938-g67fd1b79e7 Copyright (c) 2000-2023 the FFmpeg
> > developers
> >built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> >configuration:
> >libavutil  58.  3.100 / 58.  3.100
> >libavcodec 60.  5.100 / 60.  5.100
> >libavformat60.  4.100 / 60.  4.100
> >libavdevice60.  2.100 / 60.  2.100
> >libavfilter 9.  4.100 /  9.  4.100
> >libswscale  7.  2.100 /  7.  2.100
> >libswresample   4. 11.100 /  4. 11.100
> > Input #0, mpegvideo, from
> >
> '/Users/krueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v':
> >Duration: N/A, bitrate: N/A
> >Stream #0:0: Video: mpeg2video (Main), yuv420p(tv, smpte170m,
> > progressive), 720x480 [SAR 32:27 DAR 16:9], 30 fps, 59.94 tbr, 1200k tbn
> >  Side data:
> >cpb: bitrate max/min/avg: 770/0/0 buffer size: 1835008
> vbv_delay:
> > N/A
> > At least one output file must be specified
> >
> > Thanks,
> > Robert
> >
> You can use  -analyzeduration 1000M -probesize 1000M for example
> to scan more of the input file. like this:
>
> ffmpeg -i -analyzeduration 1000M -probesize 1000M
> /Users/krueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v
>
>
Thanks. I tried that and used values that were way beyond the actual file
duration and size and that didn't change the result, i.e. I still get "N/A"
for the duration. It looks a bit as if getting exact duration by iterating
through the packets isn't supported at all. Can someone confirm this?

Thanks,
Robert
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[FFmpeg-user] Retrieve duration of an m2v >MPEG>2 elementary stream file

2023-03-01 Thread Robert Krüger
Hi,

if I invoke ffmpeg on an M2V MPEG-2 elementary stream, I get "N/A" for the
duration. I guess this is because this file doesn't contain a header that
contains the duration. Is it possible to force ffmpeg to parse the entire
file to obtain the duration?

Full command line and output:

ffmpeg -i
/Users/krueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v

ffmpeg version N-109938-g67fd1b79e7 Copyright (c) 2000-2023 the FFmpeg
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration:
  libavutil  58.  3.100 / 58.  3.100
  libavcodec 60.  5.100 / 60.  5.100
  libavformat60.  4.100 / 60.  4.100
  libavdevice60.  2.100 / 60.  2.100
  libavfilter 9.  4.100 /  9.  4.100
  libswscale  7.  2.100 /  7.  2.100
  libswresample   4. 11.100 /  4. 11.100
Input #0, mpegvideo, from
'/Users/krueger/lesspain/samples/software/compressor/compressor_dvd_mpeg2_480_24p.m2v':
  Duration: N/A, bitrate: N/A
  Stream #0:0: Video: mpeg2video (Main), yuv420p(tv, smpte170m,
progressive), 720x480 [SAR 32:27 DAR 16:9], 30 fps, 59.94 tbr, 1200k tbn
Side data:
  cpb: bitrate max/min/avg: 770/0/0 buffer size: 1835008 vbv_delay:
N/A
At least one output file must be specified

Thanks,
Robert
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Re: [FFmpeg-user] Errors and artefacts when decoding a .h264 file from security camera

2020-09-14 Thread Robert Krüger
On Mon, Sep 14, 2020 at 9:53 AM Robert Krüger 
wrote:

>
>
> On Sat, Sep 12, 2020 at 3:43 PM Carl Eugen Hoyos 
> wrote:
>
>> Am Mi., 9. Sept. 2020 um 17:18 Uhr schrieb Robert Krüger
>> :
>> >
>> > Hi,
>> >
>> > I have a security camera file that gives me decoding errors which are
>> very
>> > likely the reason for several macroblocks being broken in the resulting
>> > video (by far not all, you can still recognize the motive).
>>
>> You forgot to mention that the file contains images from several
>> cameras...
>>
>
> I am assuming this is how the system records the feed from multiple
> cameras, at least that's what I was told by the people who provided the
> file. So, it's a recording device that records a signal that apparently
> comes from a king of mixer, to a file. No processing was done on my side.
>
>
>>
>> Paul sent a patch.
>>
>
> Awesome, I will check it out!
>
>

OK, so it's really two video streams in the file. Cool. The patch works
perfectly, thanks!
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Re: [FFmpeg-user] Errors and artefacts when decoding a .h264 file from security camera

2020-09-14 Thread Robert Krüger
On Sat, Sep 12, 2020 at 3:43 PM Carl Eugen Hoyos  wrote:

> Am Mi., 9. Sept. 2020 um 17:18 Uhr schrieb Robert Krüger
> :
> >
> > Hi,
> >
> > I have a security camera file that gives me decoding errors which are
> very
> > likely the reason for several macroblocks being broken in the resulting
> > video (by far not all, you can still recognize the motive).
>
> You forgot to mention that the file contains images from several cameras...
>

I am assuming this is how the system records the feed from multiple
cameras, at least that's what I was told by the people who provided the
file. So, it's a recording device that records a signal that apparently
comes from a king of mixer, to a file. No processing was done on my side.


>
> Paul sent a patch.
>

Awesome, I will check it out!
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Re: [FFmpeg-user] rav1e encoding only using one core

2020-09-14 Thread Robert Krüger
On Sat, Sep 12, 2020 at 3:48 PM Carl Eugen Hoyos  wrote:

> Am Fr., 11. Sept. 2020 um 15:48 Uhr schrieb Robert Krüger
> :
>
> > I'm playing around with rav1e and noticed in my first test that only one
> > core is used of the 8 (16 virtual) I have. I tried with and without
> > -threads setting. Since most other codecs behave this way I was expecting
> > it to use as many cores as possible if not constrained by the command
> line.
>
> Your question sounds as if you tested the rav1e command line utility
> and it was able to use more than one thread.
> Did you?
>
>
No, and I should do that. I tested libaom-av1, found its speed far too slow
to be usable, then read that rav1 had been developed exactly to provide a
faster encoder for real-world use and assumed that multithreading must be a
part of it, because without it, it seems to be just as unusable as
libaom-av1 for many practical use cases.

Robert
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Re: [FFmpeg-user] Errors and artefacts when decoding a .h264 file from security camera

2020-09-11 Thread Robert Krüger
On Fri, Sep 11, 2020 at 9:47 AM Carl Eugen Hoyos  wrote:

>
>
> > Am 09.09.2020 um 17:18 schrieb Robert Krüger  >:
> >
> > I have a security camera file that gives me decoding errors which are
> very
> > likely the reason for several macroblocks being broken in the resulting
> > video (by far not all, you can still recognize the motive).
> >
> > Are there any command line flags I can try to fix this or shall I submit
> a
> > trac ticket with the file?
>
> Please provide a sample file.
>
>
>
Created ticket https://trac.ffmpeg.org/ticket/ and added a link to a
sample file.

Thanks,
Robert
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[FFmpeg-user] rav1e encoding only using one core

2020-09-11 Thread Robert Krüger
Hi,

I'm playing around with rav1e and noticed in my first test that only one
core is used of the 8 (16 virtual) I have. I tried with and without
-threads setting. Since most other codecs behave this way I was expecting
it to use as many cores as possible if not constrained by the command line.

I tried:

ffmpeg -i prores_hq_1080_24p_varying_motives_80s.mov -c:v librav1e -speed 4
-an prores_hq_1080_24p_varying_motives_80s_rav1e_speed4.mp4

and the same with "-threads 8" after the input file with identical results.
Without multithreading this is extremely slow.

Other codec like x264, x265, kvazaar multithread fine with the same built.
So it's not a general problem with multithreading.

Working on OS X Catalina.

Thanks for any hints,

Robert
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[FFmpeg-user] Errors and artefacts when decoding a .h264 file from security camera

2020-09-09 Thread Robert Krüger
Hi,

I have a security camera file that gives me decoding errors which are very
likely the reason for several macroblocks being broken in the resulting
video (by far not all, you can still recognize the motive).

Are there any command line flags I can try to fix this or shall I submit a
trac ticket with the file?

Thanks,
Robert

Command line and output:

~/src/FFmpeg/ffmpeg -i file.h264 -c:v mpeg4 -q 1 file-transcoded.mov
ffmpeg version N-98974-g2a19232c19 Copyright (c) 2000-2020 the FFmpeg
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --enable-libkvazaar --enable-gpl --enable-libx264
--enable-libx265
  libavutil  56. 58.100 / 56. 58.100
  libavcodec 58.101.101 / 58.101.101
  libavformat58. 51.101 / 58. 51.101
  libavdevice58. 11.101 / 58. 11.101
  libavfilter 7. 87.100 /  7. 87.100
  libswscale  5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc55.  8.100 / 55.  8.100
[h264 @ 0x7fb2b0009400] Increasing reorder buffer to 1
[h264 @ 0x7fb2b0009400] Increasing reorder buffer to 2
[h264 @ 0x7fb2b0009400] Increasing reorder buffer to 4
Input #0, h264, from 'file.h264':
  Duration: N/A, bitrate: N/A
Stream #0:0: Video: h264 (High), yuvj420p(pc, bt709, progressive),
1280x720, 25 fps, 25 tbr, 1200k tbn, 50 tbc
File 'file-transcoded.mov' already exists. Overwrite? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> mpeg4 (native))
Press [q] to stop, [?] for help
[swscaler @ 0x7fb260f0] deprecated pixel format used, make sure you did
set range correctly
Output #0, mov, to 'file-transcoded.mov':
  Metadata:
encoder : Lavf58.51.101
Stream #0:0: Video: mpeg4 (mp4v / 0x7634706D), yuv420p, 1280x720,
q=2-31, 200 kb/s, 25 fps, 12800 tbn, 25 tbc
Metadata:
  encoder : Lavc58.101.101 mpeg4
Side data:
  cpb: bitrate max/min/avg: 0/0/20 buffer size: 0 vbv_delay: N/A
[h264 @ 0x7fb2ae80ee00] Invalid NAL unit 0, skipping.
[h264 @ 0x7fb2ae823000] Invalid NAL unit 0, skipping.
[h264 @ 0x7fb2ae810600] A non-intra slice in an IDR NAL
unit.trate=3503.0kbits/s speed=17.9x
[h264 @ 0x7fb2ae810600] decode_slice_header error
[h264 @ 0x7fb2ae80ca00] A non-intra slice in an IDR NAL unit.
[h264 @ 0x7fb2ae80ca00] decode_slice_header error
[h264 @ 0x7fb2ae80f400] Invalid NAL unit 0, skipping.
[h264 @ 0x7fb2ae80c400] Invalid NAL unit 0, skipping.
[h264 @ 0x7fb2ae80ca00] A non-intra slice in an IDR NAL
unit.trate=3498.9kbits/s speed=19.1x
[h264 @ 0x7fb2ae80ca00] decode_slice_header error
[NULL @ 0x7fb2b0009400] sps_id 32 out of range
[h264 @ 0x7fb2ae80f400] Invalid NAL unit 0, skipping.
[NULL @ 0x7fb2b0009400] sps_id 32 out of range
[h264 @ 0x7fb2ae80c400] Invalid NAL unit 0, skipping.
[NULL @ 0x7fb2b0009400] pps_id 3199971767 out of range8.44
bitrate=3480.9kbits/s speed=19.4x
[h264 @ 0x7fb2ae80f400] Invalid NAL unit 0, skipping.
[NULL @ 0x7fb2b0009400] pps_id 3199971767 out of range
[h264 @ 0x7fb2ae80c400] Invalid NAL unit 0, skipping.
[NULL @ 0x7fb2b0009400] too many reference frames 32:08.40
bitrate=3495.3kbits/s speed=19.5x
[h264 @ 0x7fb2ae823000] too many reference frames 32
Last message repeated 2 times
frame= 2014 fps=490 q=1.0 Lsize=   34484kB time=00:01:20.52
bitrate=3508.3kbits/s speed=19.6x
video:34474kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.028308%
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[FFmpeg-user] Encode metadata into jpg and png

2020-03-24 Thread Robert Krüger
Hi,

is there a way to encode any arbitrary metadata into jpg or png files using
ffmpeg? I am looking for a way to "mark" files as coming from a certain
source. I don't care what kind of metadata it would be in, e.g. EXIF, IPTC
or whatever. Any arbitrary string or number would work.

Thanks,
Robert
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-20 Thread Robert Krüger
On Tue, Dec 18, 2018 at 11:40 PM Carl Eugen Hoyos 
wrote:

> 2018-12-18 21:49 GMT+01:00, Robert Krüger :
> > On Fri, Dec 14, 2018 at 5:28 PM Carl Eugen Hoyos 
> wrote:
> >
> >> 2018-12-14 17:19 GMT+01:00, Robert Krüger :
> >> > On Fri, Dec 14, 2018 at 3:03 AM Carl Eugen Hoyos 
> >> wrote:
> >> >
> >> >> 2018-12-13 17:10 GMT+01:00, Robert Krüger  >:
> >> >>
> >> >> > is there a way to get the exact duration of a raw mpeg1 file using
> >> >> > the
> >> >> > ffmpeg command line? For testing I created a 30 minute test file
> >> >> > using
> >> >> > ffmpeg (using -c:v mpeg1video -f mpeg1video), then verified the
> >> duration
> >> >> > using VLC and mediainfo and it is indeed exactly 30 minutes long
> but
> >> >> > ffmpeg -i shows the file's duration as 00:00:04.98 after displaying
> >> this
> >> >> > warning:
> >> >> >
> >> >> > [mpegvideo @ 0x7fe212806000] Estimating duration from bitrate, this
> >> >> > may be inaccurate
> >> >> >
> >> >> > Are there command line parameters to make it compute the exact
> >> >> > size or is this a design or implementation limitation?
> >> >>
> >> >> Just the mpeg specification;-)
> >> >>
> >> > Yes, sure, having to count packets/frames is unavoidable for some
> >> formats.
> >> > My question is rather whether ffmpeg's (or rather libavformat's)
> >> > duration
> >> > computation code can be made to do this internally without these
> >> > workarounds.
> >> >
> >> >> $ ffmpeg -i input -c copy -f null -
> >> >
> >> > Thanks for the command line.
> >> >
> >> > Just a comparison: This approach gives me the duration information
> for a
> >> > 180 minute file in 1.8 seconds while mediainfo does it in 0.25 seconds
> >>
> >> (Does your input have timestamp discontinuities? You could try to
> >> concatenate a few parts - or cut away something - from the input file.)
> >>
> >
> > Very unlikely since I generated the sample using an ffmpeg command line
> > using testsrc.
>
> But you should test with a file with timestamp discontinuities to
> test your claim that it can be done faster than with FFmpeg.
>

I begin to understand. You are saying that it is possible that Mediainfo is
more or less "cheating" to obtain its results. Interesting. For my use
cases I could probably live with the risk that the method is inexact in the
case of timestamp discontinuities.


>
> >> > (being exact while ffmpeg is a few frames off, but that's a different
> >> > topic), so there must be a faster way to implement it.
> >>
> >> Patch probably welcome.
> >>
> >
> > My asking is actually with the motivation to get enough information about
> > the problem as possible to decide whether to offer sponsoring for a patch
> > implementing the behaviour I describe, so it's good to know a patch would
> > be welcome.
> >
> >
> >>
> >> > Tested with a number
> >> > of different files with identical results in principle.
> >>
> >> > What keeps confusing me is that analyzeduration and probesize really
> >> > seem
> >> > to be the parameters that should address the behaviour
> >>
> >> No, you misunderstand "analyzeduration", it does not analyze the
> >> duration of the input but specifies the duration that should be used
> >> for analysis.
> >>
> >
> > That's how I understood it, so in my tests I set it to a value beyond the
> > duration of the file so that, if needed, the entire file is parsed for
> > calculating its duration (as well as setting probesize to a value larger
> > than the file size) but that did not happen.
>
> Sorry for being unclear: The "analysis" does not contain file duration.
>

Thanks, I did not know that. That is a good explanation for what I observe.

Robert
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-18 Thread Robert Krüger
On Fri, Dec 14, 2018 at 5:31 PM Carl Zwanzig  wrote:

> On 12/14/2018 8:19 AM, Robert Krüger wrote:
> > Just a comparison: This approach gives me the duration information for a
> > 180 minute file in 1.8 seconds while mediainfo does it in 0.25 seconds
> > (being exact while ffmpeg is a few frames off, but that's a different
> > topic), so there must be a faster way to implement it. Tested with a
> number
> > of different files with identical results in principle.
>
> I'd have to look at the code, and I'm not going to now :), but maybe
> mediainfo is looking at time stamps and calculating. As another tool, have
> you tried mplayer to see what it comes up with?
>
> know because I'm actually more interested in getting this to work using
libavformat and I am just testing the possibilities using the command line.
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-18 Thread Robert Krüger
On Fri, Dec 14, 2018 at 5:28 PM Carl Eugen Hoyos  wrote:

> 2018-12-14 17:19 GMT+01:00, Robert Krüger :
> > On Fri, Dec 14, 2018 at 3:03 AM Carl Eugen Hoyos 
> wrote:
> >
> >> 2018-12-13 17:10 GMT+01:00, Robert Krüger :
> >>
> >> > is there a way to get the exact duration of a raw mpeg1 file using the
> >> > ffmpeg command line? For testing I created a 30 minute test file using
> >> > ffmpeg (using -c:v mpeg1video -f mpeg1video), then verified the
> duration
> >> > using VLC and mediainfo and it is indeed exactly 30 minutes long but
> >> > ffmpeg -i shows the file's duration as 00:00:04.98 after displaying
> this
> >> > warning:
> >> >
> >> > [mpegvideo @ 0x7fe212806000] Estimating duration from bitrate, this
> >> > may be inaccurate
> >> >
> >> > Are there command line parameters to make it compute the exact
> >> > size or is this a design or implementation limitation?
> >>
> >> Just the mpeg specification;-)
> >>
> > Yes, sure, having to count packets/frames is unavoidable for some
> formats.
> > My question is rather whether ffmpeg's (or rather libavformat's) duration
> > computation code can be made to do this internally without these
> > workarounds.
> >
> >> $ ffmpeg -i input -c copy -f null -
> >
> > Thanks for the command line.
> >
> > Just a comparison: This approach gives me the duration information for a
> > 180 minute file in 1.8 seconds while mediainfo does it in 0.25 seconds
>
> (Does your input have timestamp discontinuities? You could try to
> concatenate a few parts - or cut away something - from the input file.)
>

Very unlikely since I generated the sample using an ffmpeg command line
using testsrc.


>
> > (being exact while ffmpeg is a few frames off, but that's a different
> > topic), so there must be a faster way to implement it.
>
> Patch probably welcome.
>

My asking is actually with the motivation to get enough information about
the problem as possible to decide whether to offer sponsoring for a patch
implementing the behaviour I describe, so it's good to know a patch would
be welcome.


>
> > Tested with a number
> > of different files with identical results in principle.
>
> > What keeps confusing me is that analyzeduration and probesize really seem
> > to be the parameters that should address the behaviour
>
> No, you misunderstand "analyzeduration", it does not analyze the
> duration of the input but specifies the duration that should be used
> for analysis.
>

That's how I understood it, so in my tests I set it to a value beyond the
duration of the file so that, if needed, the entire file is parsed for
calculating its duration (as well as setting probesize to a value larger
than the file size) but that did not happen.

Robert


>
> Carl Eugen
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-14 Thread Robert Krüger
On Fri, Dec 14, 2018 at 3:03 AM Carl Eugen Hoyos  wrote:

> 2018-12-13 17:10 GMT+01:00, Robert Krüger :
>
> > is there a way to get the exact duration of a raw mpeg1 file using the
> > ffmpeg command line? For testing I created a 30 minute test file using
> > ffmpeg (using -c:v mpeg1video -f mpeg1video), then verified the duration
> > using VLC and mediainfo and it is indeed exactly 30 minutes long but
> ffmpeg
> > -i shows the file's duration as 00:00:04.98 after displaying this
> warning:
> >
> > [mpegvideo @ 0x7fe212806000] Estimating duration from bitrate, this may
> be
> > inaccurate
> >
> > Are there command line parameters to make it compute the exact size or is
> > this a design or implementation limitation?
>
> Just the mpeg specification;-)
>
>
Yes, sure, having to count packets/frames is unavoidable for some formats.
My question is rather whether ffmpeg's (or rather libavformat's) duration
computation code can be made to do this internally without these
workarounds.


> $ ffmpeg -i input -c copy -f null -
>

Thanks for the command line.

Just a comparison: This approach gives me the duration information for a
180 minute file in 1.8 seconds while mediainfo does it in 0.25 seconds
(being exact while ffmpeg is a few frames off, but that's a different
topic), so there must be a faster way to implement it. Tested with a number
of different files with identical results in principle.

What keeps confusing me is that analyzeduration and probesize really seem
to be the parameters that should address the behaviour and give the user a
choice to make the trade-off between accuracy and performance but
specifying these with limits beyond file duration and size does not change
anything in this case. Looking at avformat_find_stream_info this looks like
what it was meant to do but I am by far no ffmpeg code expert, so I might
be misreading it.

Disclaimer: I am neither criticising ffmpeg nor promoting mediainfo :-).
Just interested in getting closer to a solution for the ffmpeg/libavformat
community and trying to understand what the status quo is.

Thanks,
Robert
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-13 Thread Robert Krüger
Hi Carl,

On Thu, Dec 13, 2018 at 5:31 PM Carl Zwanzig  wrote:

> On 12/13/2018 8:10 AM, Robert Krüger wrote:
> > Are there command line parameters to make it compute the exact size or is
> > this a design or implementation limitation?
>
> IIRC (and I could be wrong), the generic "info" uses metadata whereas
> mediainfo actually counts frames and applies the frame rate to that.
> Without
> a container, there's no appropriate metadata.
>

yes, absolutely, that's what I assumed, I just didn't know the option to
force it to do something else.


>
> It's easy enough to tell, if ffmpeg -i returns almost immediately, it's
> probably not counting frames. You also might get a more accurate answer by
> using -probesize with a parameter larger than the actual file.
>
>
that did the trick, thanks! There still a minor inaccuracy of a few frames
but that is probably still https://trac.ffmpeg.org/ticket/3683. My main
concern was estimating vs. actually parsing.

Thanks again,
Robert
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Re: [FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-13 Thread Robert Krüger
Hi again,

On Thu, Dec 13, 2018 at 6:01 PM Robert Krüger 
wrote:

> Hi Carl,
>
> On Thu, Dec 13, 2018 at 5:31 PM Carl Zwanzig  wrote:
>
>> On 12/13/2018 8:10 AM, Robert Krüger wrote:
>> > Are there command line parameters to make it compute the exact size or
>> is
>> > this a design or implementation limitation?
>>
>> IIRC (and I could be wrong), the generic "info" uses metadata whereas
>> mediainfo actually counts frames and applies the frame rate to that.
>> Without
>> a container, there's no appropriate metadata.
>>
>
> yes, absolutely, that's what I assumed, I just didn't know the option to
> force it to do something else.
>
>
>>
>> It's easy enough to tell, if ffmpeg -i returns almost immediately, it's
>> probably not counting frames. You also might get a more accurate answer
>> by
>> using -probesize with a parameter larger than the actual file.
>>
>>
> that did the trick, thanks! There still a minor inaccuracy of a few frames
> but that is probably still https://trac.ffmpeg.org/ticket/3683. My main
> concern was estimating vs. actually parsing.
>
>
>
Sorry, I have to take that back as I used the wrong file and got something
mixed up when looking at it. -probesize doesn't change the behaviour at
all. I set it to a value larger than the file and still got the warning and
the incorrect number. So for mpeg1 this does not seem to work and seems to
be consistent with my last test a long time ago.

Sorry about the confusion.

Robert
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[FFmpeg-user] Get exact duration of an mpeg1 file

2018-12-13 Thread Robert Krüger
Hi,

is there a way to get the exact duration of a raw mpeg1 file using the
ffmpeg command line? For testing I created a 30 minute test file using
ffmpeg (using -c:v mpeg1video -f mpeg1video), then verified the duration
using VLC and mediainfo and it is indeed exactly 30 minutes long but ffmpeg
-i shows the file's duration as 00:00:04.98 after displaying this warning:

[mpegvideo @ 0x7fe212806000] Estimating duration from bitrate, this may be
inaccurate

Are there command line parameters to make it compute the exact size or is
this a design or implementation limitation?

Thanks,
Robert
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[FFmpeg-user] Extraction of timebase/frame rate for timecodes of audio files

2017-12-12 Thread Robert Krüger
Hi,

is there a way to extract the information about the timebase of a timecode
in an audio file, like you can find in the wild e.g. in MXF or in Mov
files? I can see that ffmpeg apis and thus ffprobe expose the timecode as a
string-valued tag (e.g. TAG:timecode=00:00:00:00) but the information what
timebase the timecode uses, which IIRC is encoded in MXF or Mov is not
exposed or am I missing something? When the file contains a video stream a
workaround is to use the frame rate of the video stream to guess what the
timebase of the timecode probably is but for audio files I fail to see a
possibility to extract the correct timecode.

Is this a limitation or am I in fact missing something?

Thanks,

Robert
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Re: [FFmpeg-user] Muxing audio-only MXF

2017-12-01 Thread Robert Krüger
Hi Erik,

On Fri, Dec 1, 2017 at 12:25 PM, Erik Dobberkau 
wrote:

> Am Freitag, 1. Dezember 2017 schrieb Robert Krüger :
>
> > Hi Kieran,
> >
> > On Thu, Nov 30, 2017 at 5:39 PM, Kieran O Leary <
> kieran.o.le...@gmail.com
> > >
> > wrote:
> >
> > > On Thu, Nov 30, 2017 at 4:38 PM, Kieran O Leary
> > > > wrote:
> > > > Hi,
> > > >
> > > > On Thu, Nov 30, 2017 at 3:46 PM, Robert Krüger
> > > >  wrote:
> > > >> Hi,
> > > >>
> > > >> is there a way to mux an mxf file only containing a pcm audio
> stream?
> > > When
> > > >
> > > > You could try using the op-atom muxer as it looks like your command
> is
> > > > defaulting to op1a. Add -f  mxf_opatom to your command line.
> > > > ffmpeg -i
> > > > /Users/krueger/lesspain/samples/software/compressor/
> > > compressor4_1080_25p_apple_devices_best_compat_2s.m4v
> > > > -vn -c:a pcm_s16le -f mxf_opatom ~/tmp/ffmpeg_pcm16_rewrap.mxf
> > > >
> > > > [...]
> > > >
> > > >>
> > > >> Is this a conscious limitation of the muxer or am I missing a
> command
> > > line
> > > >> option?
> > > >>
> > > >
> > > > It looks like it. The reasons why I suggested trying OP-Atom is that
> > > > this seems to be performing a specific check for OP1a.
> >
> >
> Which is wrong, btw. There is a spec for MXF_OP1a_AES3 to wrap the
> respective audio without a picture essence. I would guess MCA labels will
> have to be provided though, and also the audio samples would have to be
> formatted to AES, which ffmpeg afaik currently doesn't do.
>
> Best,
> Erik
>
>
tanks a lot for the additional info! In my case, it's fine for my limited
scope of application where I need to generate audio-only MXF files that
just need to be decodable by ffmpeg for internal testing purposes, so I'm
still OK with what it does now but I will keep that in mind for later when
we might want to do more with it.

Best,
Robert
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Re: [FFmpeg-user] Muxing audio-only MXF

2017-12-01 Thread Robert Krüger
Hi Kieran,

On Thu, Nov 30, 2017 at 5:39 PM, Kieran O Leary 
wrote:

> On Thu, Nov 30, 2017 at 4:38 PM, Kieran O Leary
>  wrote:
> > Hi,
> >
> > On Thu, Nov 30, 2017 at 3:46 PM, Robert Krüger
> >  wrote:
> >> Hi,
> >>
> >> is there a way to mux an mxf file only containing a pcm audio stream?
> When
> >
> > You could try using the op-atom muxer as it looks like your command is
> > defaulting to op1a. Add -f  mxf_opatom to your command line.
> > ffmpeg -i
> > /Users/krueger/lesspain/samples/software/compressor/
> compressor4_1080_25p_apple_devices_best_compat_2s.m4v
> > -vn -c:a pcm_s16le -f mxf_opatom ~/tmp/ffmpeg_pcm16_rewrap.mxf
> >
> > [...]
> >
> >>
> >> Is this a conscious limitation of the muxer or am I missing a command
> line
> >> option?
> >>
> >
> > It looks like it. The reasons why I suggested trying OP-Atom is that
> > this seems to be performing a specific check for OP1a.
> >
> > Best,
> >
> > Kieran.
>
> Eek, I meant to link to the line of code where this check seems to
> occur: https://github.com/FFmpeg/FFmpeg/blob/master/
> libavformat/mxfenc.c#L2151
>
>
perfect, that worked, thanks a lot! The only thing I had to do is to
convert the audio to single-channel "[mxf_opatom @ 0x7feb4e009a00] MXF
OPAtom only supports single channel audio".

Cheers,

Robert
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[FFmpeg-user] Muxing audio-only MXF

2017-11-30 Thread Robert Krüger
Hi,

is there a way to mux an mxf file only containing a pcm audio stream? When
I try I get this:

ffmpeg -i
/Users/krueger/lesspain/samples/software/compressor/compressor4_1080_25p_apple_devices_best_compat_2s.m4v
-vn -c:a pcm_s16le ~/tmp/ffmpeg_pcm16_rewrap.mxf
ffmpeg version git-2017-06-12-358ddb4 Copyright (c) 2000-2017 the FFmpeg
developers
  built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
  configuration: --enable-shared --disable-static --disable-doc
--disable-devices --enable-indev=lavfi
--prefix=/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/ffmpeg
--install-name-dir='@rpath'
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libmp3lame/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libmp3lame/lib/
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libogg/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libogg/lib/
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libvorbis/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libvorbis/lib/
--extra-ldflags='-Wl,-rpath,@loader_path/../lib' --enable-libvorbis
--enable-libvpx --enable-libmp3lame --enable-libopenh264
--enable-libfreetype
  libavutil  55. 63.100 / 55. 63.100
  libavcodec 57. 98.100 / 57. 98.100
  libavformat57. 73.100 / 57. 73.100
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 91.100 /  6. 91.100
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Users/krueger/lesspain/samples/software/compressor/compressor4_1080_25p_apple_devices_best_compat_2s.m4v':
  Metadata:
major_brand : M4V
minor_version   : 1
compatible_brands: M4V M4A mp42isom
creation_time   : 2015-09-15T08:19:57.00Z
  Duration: 00:00:02.00, start: 0.00, bitrate: 3881 kb/s
Stream #0:0(deu): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, fltp, 63 kb/s (default)
Metadata:
  creation_time   : 2015-09-15T08:19:57.00Z
  handler_name: Core Media Audio
Stream #0:1(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv,
smpte170m/smpte170m/bt709, progressive), 960x540 [SAR 1:1 DAR 16:9], 3807
kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc (default)
Metadata:
  creation_time   : 2015-09-15T08:19:57.00Z
  handler_name: Core Media Video
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
[mxf @ 0x7f8ccd803200] there must be exactly one video stream and it must
be the first one
Could not write header for output file #0 (incorrect codec parameters ?):
Operation not permitted
Error initializing output stream 0:0 --
Conversion failed!

Is this a conscious limitation of the muxer or am I missing a command line
option?

Thanks,

Robert
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Re: [FFmpeg-user] Current master not compiling on Mac

2017-06-24 Thread Robert Krüger
Hi,

On Sat, Jun 24, 2017 at 3:22 AM, Moritz Barsnick  wrote:

> On Fri, Jun 23, 2017 at 20:59:50 +0200, Robert Krüger wrote:
> > Hi,
> >
> > I just did a clean check-out on master did a plain configure with no
> > options and the build failed. Heres the end of the output when the errors
> > start:
> >
> > CC libavfilter/vsrc_life.o
> > CC libavfilter/vsrc_mandelbrot.o
> > CC libavfilter/vsrc_testsrc.o
> > CC libavfilter/window_func.o
> > X86ASM libavfilter/x86/af_afir.o
> > yasm: warning: can open only one input file, only the last file will be
> processed
>
> It sound as if this issue's description matches your observation:
> http://ffmpeg.org/pipermail/ffmpeg-devel/2017-June/212757.html
>
> That provided fix was just pushed to git (see third e-mail in the
> thread:
> http://ffmpeg.org/pipermail/ffmpeg-devel/2017-June/212766.html
> )
>
> git pull should fix your issue.
>
> Moritz
>
>
great, thanks! Pull fixed it now.

Robert
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[FFmpeg-user] Current master not compiling on Mac

2017-06-23 Thread Robert Krüger
Hi,

I just did a clean check-out on master did a plain configure with no
options and the build failed. Heres the end of the output when the errors
start:

CC libavfilter/vsrc_life.o
CC libavfilter/vsrc_mandelbrot.o
CC libavfilter/vsrc_testsrc.o
CC libavfilter/window_func.o
X86ASM libavfilter/x86/af_afir.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/af_afir.o: libavfilter/x86/af_afir.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/af_afir.o
error:
/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/strip:
can't open file: libavfilter/x86/af_afir.o (No such file or directory)
make: [libavfilter/x86/af_afir.o] Error 1 (ignored)
CC libavfilter/x86/af_afir_init.o
X86ASM libavfilter/x86/af_volume.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/af_volume.o: libavfilter/x86/af_volume.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/af_volume.o
CC libavfilter/x86/af_volume_init.o
X86ASM libavfilter/x86/avf_showcqt.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/avf_showcqt.o: libavfilter/x86/avf_showcqt.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/avf_showcqt.o
CC libavfilter/x86/avf_showcqt_init.o
X86ASM libavfilter/x86/colorspacedsp.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/colorspacedsp.o: libavfilter/x86/colorspacedsp.asm \
   config.asm libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/colorspacedsp.o
CC libavfilter/x86/colorspacedsp_init.o
X86ASM libavfilter/x86/vf_blend.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_blend.o: libavfilter/x86/vf_blend.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_blend.o
CC libavfilter/x86/vf_blend_init.o
X86ASM libavfilter/x86/vf_bwdif.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_bwdif.o: libavfilter/x86/vf_bwdif.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_bwdif.o
CC libavfilter/x86/vf_bwdif_init.o
X86ASM libavfilter/x86/vf_gradfun.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_gradfun.o: libavfilter/x86/vf_gradfun.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_gradfun.o
CC libavfilter/x86/vf_gradfun_init.o
X86ASM libavfilter/x86/vf_idet.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_idet.o: libavfilter/x86/vf_idet.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_idet.o
CC libavfilter/x86/vf_idet_init.o
X86ASM libavfilter/x86/vf_maskedmerge.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_maskedmerge.o: libavfilter/x86/vf_maskedmerge.asm \
   config.asm libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_maskedmerge.o
CC libavfilter/x86/vf_maskedmerge_init.o
CC libavfilter/x86/vf_noise.o
X86ASM libavfilter/x86/vf_psnr.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_psnr.o: libavfilter/x86/vf_psnr.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_psnr.o
CC libavfilter/x86/vf_psnr_init.o
CC libavfilter/x86/vf_removegrain_init.o
X86ASM libavfilter/x86/vf_ssim.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_ssim.o: libavfilter/x86/vf_ssim.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_ssim.o
CC libavfilter/x86/vf_ssim_init.o
X86ASM libavfilter/x86/vf_w3fdif.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_w3fdif.o: libavfilter/x86/vf_w3fdif.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_w3fdif.o
CC libavfilter/x86/vf_w3fdif_init.o
X86ASM libavfilter/x86/vf_yadif.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/vf_yadif.o: libavfilter/x86/vf_yadif.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/vf_yadif.o
CC libavfilter/x86/vf_yadif_init.o
X86ASM libavfilter/x86/yadif-10.o
yasm: warning: can open only one input file, only the last file will be
processed
libavfilter/x86/yadif-10.o: libavfilter/x86/yadif-10.asm config.asm \
   libavutil/x86/x86util.asm libavutil/x86/x86inc.asm
STRIP libavfilter/x86/yadif-10.o
X86ASM libavfilter/x86/yadif-16.o
yasm: warning: can open only one input file, onl

Re: [FFmpeg-user] Writing EXIF metadata

2016-10-26 Thread Robert Krüger
Hi Erik,

thanks for the hint. I know Exiftool but in this case I need the capability
within ffmpeg. At least that would make it a lot easier.

Robert

On Tue, Oct 25, 2016 at 8:25 PM, Erik Dobberkau 
wrote:

> Am Dienstag, 25. Oktober 2016 schrieb Robert Krüger :
>
> > Hi,
> >
> > is it possible to add EXIF metadata like title or description to a JPG or
> > PNG image using ffmpeg on the command line (or via api but that's
> probably
> > for the other list)?
> >
>
> Hi Robert,
>
> if not, maybe exiftool is a viable alternative...
>
> Erik
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-- 
Robert Krüger
Managing Partner
Lesspain GmbH & Co. KG

http://lesspain.software
http://kyno.software
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[FFmpeg-user] Writing EXIF metadata

2016-10-25 Thread Robert Krüger
Hi,

is it possible to add EXIF metadata like title or description to a JPG or
PNG image using ffmpeg on the command line (or via api but that's probably
for the other list)?

I did not find anything regarding that in the docs.

Thanks in advance,

Robert
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Re: [FFmpeg-user] Extracting embedded xml metadata im mp4

2016-10-21 Thread Robert Krüger
On Thu, Oct 20, 2016 at 10:49 AM, Carl Eugen Hoyos 
wrote:

> 2016-10-20 10:41 GMT+02:00 Robert Krüger :
>
> > So would a patch be accepted that implements reading of this
> > additional metadata?
>
> (Sorry if this was not directed to me...)
>
> I am not maintainer of mov.c and therefore cannot answer but
> the only way to find out is to send a (clean) patch made with
> git format-patch to the development mailing list.
>
>
OK, I created ticket 5901 so it is documented and can be referred to.
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Re: [FFmpeg-user] Extracting embedded xml metadata im mp4

2016-10-20 Thread Robert Krüger
On Wed, Oct 19, 2016 at 11:35 PM, Carl Eugen Hoyos 
wrote:

> 2016-10-19 19:31 GMT+02:00 Robert Krüger :
>
> > Are those meta atoms parsed in the mov demuxer?
>
> Yes, but most content (except "hdlr") is ignored.
>
>
Thanks for the clarification!

So would a patch be accepted that implements reading of this additional
metadata?
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Re: [FFmpeg-user] Extracting embedded xml metadata im mp4

2016-10-19 Thread Robert Krüger
On Wed, Oct 19, 2016 at 5:39 PM, Moritz Barsnick  wrote:

> On Wed, Oct 19, 2016 at 12:32:00 +0200, Robert Krüger wrote:
> > Hi,
> >
> > I am currently looking at files from Sony A7S cameras (h264 in mp4) and
> > looking at those files in a hex/text editor I can see that they have xml
> > metadata written in a "meta" tag as far as I can see:
>
> I couldn't find a sample to test. I found one with a "Timed Metadata
> Media Handler" data stream in this ticket:
> https://trac.ffmpeg.org/ticket/5645
> but it seems to contain binary data.
>
> Using
> $ ffprobe -show_packets -show_data
> or even
> $ ffprobe -show_packets -show_data -print_format compact
> you should see each packet's data in hexadecimal form. That could be
> parsed back to plain text (or binary, or whatever was in your data
> stream).
>
> To actually *extract* the data, I used this patch:
> http://ffmpeg.org/pipermail/ffmpeg-devel/2016-May/194445.html
>
> A newer version is here, though I think that's not the one I tested:
> http://ffmpeg.org/pipermail/ffmpeg-devel/2016-June/195929.html
>
> Note that both were rejected on ffmpeg-devel.
>
> I used that patch (patched ffmpeg) to extract the data stream to
> (Base64 text) file, and converted that to original data:
>
> $ grep -Ev ':|^\;' sony_a7s_XAVC_S_1080_25p_10s.fftextdata | perl
> -MMIME::Base64 -e 'while(<>) { print decode_base64($_) . "\n"; }'
>
>
I was wondering why the sample name looked familiar. It's mine :-).


> But as I said, there's only binary data in that sample apparently, and
> I *seem* to see the same stuff in the extracted data an in the MP4
> file.
>
>
In my case, I would not expect the data to be in a packet, so I guess the
-show_packets approach would not work. I want to extract the data using API
anyway but if it works on the command line, I know it works via API as well.

I just checked. The sample from ticket 5645 also contains that xml metadata
in the "meta" atom which is the last atom of the file. Contained data is:

http://www.w3.org/2001/XMLSchema-instance";
lastUpdate="2016-03-05T14:17:25Z">











Are those meta atoms parsed in the mov demuxer?
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[FFmpeg-user] Extracting embedded xml metadata im mp4

2016-10-19 Thread Robert Krüger
Hi,

I am currently looking at files from Sony A7S cameras (h264 in mp4) and
looking at those files in a hex/text editor I can see that they have xml
metadata written in a "meta" tag as far as I can see:


http://www.w3.org/2001/XMLSchema-instance";
lastUpdate="2016-03-14T14:00:37Z">











Is it possible to extract that XML using ffmpeg or ffprobe? I tried this:

ffprobe -export_all 1 -show_format /Volumes/LPSSD2/Shoots/Animals/C0032.MP4
ffprobe version git-2016-10-17-2f0ea8d Copyright (c) 2007-2016 the FFmpeg
developers
  built with Apple LLVM version 6.0 (clang-600.0.57) (based on LLVM 3.5svn)
  configuration: --enable-shared --disable-static --disable-doc
--disable-devices --enable-indev=lavfi
--prefix=/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/ffmpeg
--install-name-dir='@rpath'
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libmp3lame/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libmp3lame/lib/
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libogg/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libogg/lib/
--extra-cflags=-I/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libvorbis/include/
--extra-ldflags=-L/Users/teamcity/buildAgent/work/3b1b1b1b725f407e/build/libvorbis/lib/
--extra-ldflags='-Wl,-rpath,@loader_path/../lib' --enable-libvorbis
--enable-libvpx --enable-libmp3lame --enable-libopenh264
--enable-libfreetype
  libavutil  55. 32.100 / 55. 32.100
  libavcodec 57. 61.103 / 57. 61.103
  libavformat57. 52.100 / 57. 52.100
  libavdevice57.  0.102 / 57.  0.102
  libavfilter 6. 64.100 /  6. 64.100
  libswscale  4.  1.100 /  4.  1.100
  libswresample   2.  2.100 /  2.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Volumes/LPSSD2/Shoots/Animals/C0032.MP4':
  Metadata:
major_brand : XAVC
minor_version   : 16785407
compatible_brands: XAVCmp42iso2
creation_time   : 2016-03-14T14:00:37.00Z
  Duration: 00:00:05.51, start: 0.00, bitrate: 48999 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv),
1920x1080 [SAR 1:1 DAR 16:9], 47263 kb/s, 23.98 fps, 23.98 tbr, 24k tbn,
47.95 tbc (default)
Metadata:
  creation_time   : 2016-03-14T14:00:37.00Z
  handler_name: Video Media Handler
  encoder : AVC Coding
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2
channels, s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2016-03-14T14:00:37.00Z
  handler_name: Sound Media Handler
Stream #0:2(und): Data: none (rtmd / 0x646D7472), 196 kb/s (default)
Metadata:
  creation_time   : 2016-03-14T14:00:37.00Z
  handler_name: Timed Metadata Media Handler
  timecode: 00:15:40:08
Unsupported codec with id 0 for input stream 2
[FORMAT]
filename=/Volumes/LPSSD2/Shoots/Animals/C0032.MP4
nb_streams=3
nb_programs=0
format_name=mov,mp4,m4a,3gp,3g2,mj2
format_long_name=QuickTime / MOV
start_time=0.00
duration=5.505500
size=33721095
bit_rate=48999865
probe_score=100
TAG:major_brand=XAVC
TAG:minor_version=16785407
TAG:compatible_brands=XAVCmp42iso2
TAG:creation_time=2016-03-14T14:00:37.00Z
[/FORMAT]

but as you can see, the metadata is not exported.

Any hints would be appreciated.

Thanks,

Robert
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Re: [FFmpeg-user] How to create a multichannel wav file without a specific channel layout

2016-09-26 Thread Robert Krüger
Hi Tobias,

On Mon, Sep 26, 2016 at 3:58 PM, Tobias Rapp  wrote:

> On 26.09.2016 13:10, Robert Krüger wrote:
>
>> Hi,
>>
>> On Mon, Sep 26, 2016 at 12:42 PM, Carl Eugen Hoyos 
>> wrote:
>>
>> 2016-09-26 12:17 GMT+02:00 Robert Krüger :
>>>
>>> I would like to know if it is possible to create a wave file that
>>>>
>>> contains
>>>
>>>> n channels of audio which do not have a channel layout like 5.1 or
>>>>
>>> stereo,
>>>
>>> (I assumed n==1 here because it should work fine for n>2)
>>>
>>> much like a file used in broadcast for just storing n signals in one wav
>>>> file.
>>>>
>>>
>>> It works fine for >16bit or >48kHz.
>>> I just sent a patch that fixes it for all (pcm) audio, that will not fix
>>> the
>>> front center (mono) case though: If this is an issue (if your software
>>> does not default to front center for 1-channel pcm wav), please
>>> tell me;-)
>>>
>>>
>>>
>>> Thanks for the quick reply! I meant something like writing a 4,6 or
>> 8-channel wav that just really contains 4,6,8 mono channels. E.g. some
>> cameras or other multi-track recording devices generate signals like these
>> and I would like to be able to put them in one wav file for archiving or
>> later processing.
>>
>
> If the input file doesn't have a channel mask you should be able to use
> the "-guess_layout_max 0" input option.
>
> Reading a bit through a few specs I am not sure whether
>> this requires using RF64 features (specifically the new constant
>> SPEAKER_ALL 0x8000 for the channel assignment) and, to be honest, if
>> that makes sense at all.
>>
>
> In my opinion an undefined channel mask makes sense in your case. I'm in a
> very similar situation when recording AVI files with PCM audio and there I
> use the "-write_channel_mask off" output option.
>
> You don't need RF64 features, see section "Details about dwChannelMask" in
> https://msdn.microsoft.com/en-us/library/windows/hardware/dn653308.aspx :
>
> "If, for example in a multi-channel audio authoring application, no
> speaker location is desired on any of the mono streams, the
> dwChannelMask should explicitly be set to 0."
>
> Best regards,
> Tobias
>

Thanks a lot! That was exactly the information I was looking for.

Best regards,

Robert
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Re: [FFmpeg-user] How to create a multichannel wav file without a specific channel layout

2016-09-26 Thread Robert Krüger
On Mon, Sep 26, 2016 at 2:04 PM, Carl Eugen Hoyos 
wrote:

> 2016-09-26 13:10 GMT+02:00 Robert Krüger :
>
> > I meant something like writing a 4,6 or 8-channel wav that just
> > really contains 4,6,8 mono channels.
>
Not sure if I understand this sentence.
>
> I meant something like writing an n-channel wav (n > 1) that just really
contains n mono channels.



> That fact that the option -channel_layout 0 does not work as expected
> is a long-time regression.
>
>
Is there a trac ticket for that?
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Re: [FFmpeg-user] How to create a multichannel wav file without a specific channel layout

2016-09-26 Thread Robert Krüger
Hi,

On Mon, Sep 26, 2016 at 12:42 PM, Carl Eugen Hoyos 
wrote:

> 2016-09-26 12:17 GMT+02:00 Robert Krüger :
>
> > I would like to know if it is possible to create a wave file that
> contains
> > n channels of audio which do not have a channel layout like 5.1 or
> stereo,
>
> (I assumed n==1 here because it should work fine for n>2)
>
> > much like a file used in broadcast for just storing n signals in one wav
> > file.
>
> It works fine for >16bit or >48kHz.
> I just sent a patch that fixes it for all (pcm) audio, that will not fix
> the
> front center (mono) case though: If this is an issue (if your software
> does not default to front center for 1-channel pcm wav), please
> tell me;-)
>
>
>
Thanks for the quick reply! I meant something like writing a 4,6 or
8-channel wav that just really contains 4,6,8 mono channels. E.g. some
cameras or other multi-track recording devices generate signals like these
and I would like to be able to put them in one wav file for archiving or
later processing. Reading a bit through a few specs I am not sure whether
this requires using RF64 features (specifically the new constant
SPEAKER_ALL 0x8000 for the channel assignment) and, to be honest, if
that makes sense at all.

Thanks,

Robert
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[FFmpeg-user] How to create a multichannel wav file without a specific channel layout

2016-09-26 Thread Robert Krüger
Hi,

I would like to know if it is possible to create a wave file that contains
n channels of audio which do not have a channel layout like 5.1 or stereo,
much like a file used in broadcast for just storing n signals in one wav
file.

Thanks for any hints on this,

Robert
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Re: [FFmpeg-user] Find out supported pixel formats for a codec

2016-08-21 Thread Robert Krüger
Hi Kieran,

On Sun, Aug 21, 2016 at 11:09 AM, Kieran O Leary 
wrote:

> Hi,
>
> On Sun, Aug 21, 2016 at 8:52 AM, Robert Krüger
>  wrote:
> > Hi,
> >
> > is there a systematic way to find out supported pixel formats for a codec
> > (or filter) on the command line? I only found ffmpeg -pix_fmts but that
> is
> > the global list of all known pixel formats.
> >
> Yes.
> ffmpeg -h encoder=codecname
>
> change 'encoder' to 'decoder' or 'muxer' for more info, like 'ffmpeg
> -h muxer=matroska'
>
> ffmpeg -h encoder=ffv1 -v quiet
> Encoder ffv1 [FFmpeg video codec #1]:
> General capabilities: delay threads
> Threading capabilities: slice
> Supported pixel formats: yuv420p yuva420p yuva422p yuv444p
> yuva444p yuv440p yuv422p yuv411p yuv410p bgr0 bgra yuv420p16le
> yuv422p16le yuv444p16le yuv444p9le yuv422p9le yuv420p9le yuv420p10le
> yuv422p10le yuv444p10le yuva444p16le yuva422p16le yuva420p16le
> yuva444p10le yuva422p10le yuva420p10le yuva444p9le yuva422p9le
> yuva420p9le gray16le gray gbrp9le gbrp10le gbrp12le gbrp14le ya8
> gbrp16le rgb48le
> ffv1 encoder AVOptions:
>   -slicecrc  E..V Protect slices with CRCs
> (default auto)
>   -coder E..V Coder type (from -2 to 2)
> (default rice)
>  rice E..V Golomb rice
>  range_defE..V Range with default table
>  range_tabE..V Range with custom table
>  ac   E..V Range with custom table
> (the ac option exists for compatibility and is deprecated)
>   -context   E..V Context model (from 0 to 1)
> (default 0)
>
> Best,
>
> Kieran.
>
>
supercool! You made my day!

Thanks,

Robert
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[FFmpeg-user] Find out supported pixel formats for a codec

2016-08-21 Thread Robert Krüger
Hi,

is there a systematic way to find out supported pixel formats for a codec
(or filter) on the command line? I only found ffmpeg -pix_fmts but that is
the global list of all known pixel formats.

Thanks,

Robert
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Re: [FFmpeg-user] Enable all audio tracks for MOV container

2016-08-08 Thread Robert Krüger
On Tue, Jul 19, 2016 at 11:39 PM, Christian Johannesen <
chrisjohanne...@gmail.com> wrote:

> On Tue, Jul 19, 2016 at 4:07 PM, Rens Dijkshoorn 
> wrote:
>
> > - Original Message -
> > > From: "Christian Johannesen" 
> > > To: "FFmpeg user questions" 
> > > Sent: Tuesday, 19 July, 2016 21:41:48
> > > Subject: [FFmpeg-user] Enable all audio tracks for MOV container
> >
> > > Hi all -
> > >
> > > I had been using a patch from this mailing list on the FFMPEG source
> code
> > > to enable all audio tracks when encoding QuickTime MOV.
> > >
> > >
> > > However, it seems as though this no longer works as movenc has been
> > > modified since it last worked for me with source code from March 25th
> > 2016.
> > > Is there a way to enable all audio tracks when encoding a MOV with the
> > > current source or does anyone have an updated patch?
> > >
> > > Thanks,
> > > Chris
> >
> > Hi Chris,
> >
> > this is my update patch, should work with current GIT version.
> >
> > Regards
> >
> > Rens
> >
>
> Hi Rens,
>
> Appreciate the help! Your updated patch worked for me and all tracks are
> enabled after patching and compiling the most recent source.
>
> Thanks,
> Chris
>
> Hi,

before I start using the patch myself (thanks for the patch, btw.), just
asking the obvious question. Was this discussed as a patch and rejected? At
least having it as a muxer or command line option seems to be a better
solution than not having it at all in master as there are obviously use
cases for it out there.

Thanks,
Robert
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Re: [FFmpeg-user] Unable to create movs with multi-channel tracks with discrete channel layout

2016-08-06 Thread Robert Krüger
On Fri, Aug 5, 2016 at 6:28 PM, Paul B Mahol  wrote:

> On 8/5/16, Robert Krüger  wrote:
> > Hi,
> >
> > I would like to create a file that has a "discrete" channel layout, e.g.
> > have an audio track with 8 channels that are not automatically set to 7.1
> > surround. Is that possible with the command line? My attempts so far have
> > been that the files are then created with 7.1 channel layout (checked
> using
> > Quicktime 7 which displays the assignment of each channel and compared
> that
> > to files created by cameras or other software (e.g. Adobe Media Encoder)
> > that have this property.
>
> Do you want to not write chan mov tag? This is trivial to implement.
>
>
I thought so at first but I checked the sample file I have from Adobe Media
Encoder, which has a discrete layout and that has the chan atom set.
However, looking at the quicktime spec, I would guess that not writing the
chan atome should have the same result. For now, I just need it to create
some sample files, so it would be worth a try.
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Re: [FFmpeg-user] Unable to create movs with multi-channel tracks with discrete channel layout

2016-08-05 Thread Robert Krüger
On Fri, Aug 5, 2016 at 3:52 PM, Carl Eugen Hoyos  wrote:

> Hi!
>
> > Am 05.08.2016 um 10:36 schrieb Robert Krüger  >:
> >
> > I would like to create a file that has a "discrete" channel layout, e.g.
> > have an audio track with 8 channels that are not automatically set to 7.1
> > surround. Is that possible with the command line?
>
> This did work in (very) old versions of FFmpeg with -channel_layout 0,
> unfortunately this was broken for 0.11.
>
>
thanks for clarifying. Is this considered a regression or rather by design
and the possibility was dropped consciously or in other words, does it make
sense to file a trac ticket for this?
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[FFmpeg-user] Unable to create movs with multi-channel tracks with discrete channel layout

2016-08-05 Thread Robert Krüger
Hi,

I would like to create a file that has a "discrete" channel layout, e.g.
have an audio track with 8 channels that are not automatically set to 7.1
surround. Is that possible with the command line? My attempts so far have
been that the files are then created with 7.1 channel layout (checked using
Quicktime 7 which displays the assignment of each channel and compared that
to files created by cameras or other software (e.g. Adobe Media Encoder)
that have this property.

Thanks,

Robert
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Re: [FFmpeg-user] pcm in mp4

2016-03-21 Thread Robert Krüger
On Mon, Mar 21, 2016 at 6:32 PM, Robert Krüger  wrote:

>
>
> On Mon, Mar 21, 2016 at 5:31 PM, Carl Eugen Hoyos 
> wrote:
>
>> Robert Krüger  lesspain.de> writes:
>>
>> > major_brand : mp42
>>
>> > Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774)
>>
>> If this is not invalid (I don't know) than the issue remains
>> that FFmpeg does not write mp42 files.
>>
>> Thanks for the info. I wasn't aware of that.
>

Just realized that I had come across a similar file in the past:
https://trac.ffmpeg.org/ticket/3818

Sorry for the duplication.
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Re: [FFmpeg-user] pcm in mp4

2016-03-21 Thread Robert Krüger
On Mon, Mar 21, 2016 at 5:31 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > major_brand : mp42
>
> > Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774)
>
> If this is not invalid (I don't know) than the issue remains
> that FFmpeg does not write mp42 files.
>
> Thanks for the info. I wasn't aware of that.
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Re: [FFmpeg-user] pcm in mp4

2016-03-21 Thread Robert Krüger
On Mon, Mar 21, 2016 at 4:59 PM, Moritz Barsnick  wrote:

> On Mon, Mar 21, 2016 at 15:59:04 +0100, Robert Krüger wrote:
>
> > the mp4 muxer currently does not accept pcm in mp4. However, there are
> > several cameras out there, that produce such files. Would it be possible
> to
> > support this option when using -strict -2?
>
> If the cameras are doing it wrong, why would *you* want to do so as
> well?
>
> > This way a user could create non-spec-conformant files but it would
> > not happen, when not explicitly overriding the default. Is this not
> > the way this is handled in other places?
>
> I can't speak for the MP4 standard, but if it's not defined, ffmpeg
> would have to define a (non-spec-conformant) way to do this, or imitate
> some other format, like that of your camera.
>
> But again, why would you want to do this? Which client needs this?
>
> Perhaps the QuickTime (mov) container could be an option for you, it is
> very similar to the mp4 container. (mp4 was indeed derived from
> QuickTime.) It all depends on what you are trying to achieve.
>
>
you're of course right. I can work with mov instead and that's what I do. I
was just asking, if there was an easy fix to not deal with the case that
someone rewraps footage, extracting a subclip and not being able to use the
same format as the input format. not terribly important, though.
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[FFmpeg-user] pcm in mp4

2016-03-21 Thread Robert Krüger
Hi,

the mp4 muxer currently does not accept pcm in mp4. However, there are
several cameras out there, that produce such files. Would it be possible to
support this option when using -strict -2? This way a user could create
non-spec-conformant files but it would not happen, when not explicitly
overriding the default. Is this not the way this is handled in other
places?

If you are interested in a sample file, I can provide one.

Command line and output:

ffmpeg -i
/Users/krueger/lesspain/samples/cameras/panasonic/gh4/panasonic_gh4_1080p_24_100mbit_lpcm_14s.MP4
-strict -2 -c copy -y ~/tmp/gh4_lpcm_remuxed.mp4
ffmpeg version N-79103-gd4773c9 Copyright (c) 2000-2016 the FFmpeg
developers
  built with Apple LLVM version 7.0.2 (clang-700.1.81)
  configuration:
  libavutil  55. 19.100 / 55. 19.100
  libavcodec 57. 28.103 / 57. 28.103
  libavformat57. 28.102 / 57. 28.102
  libavdevice57.  0.101 / 57.  0.101
  libavfilter 6. 39.102 /  6. 39.102
  libswscale  4.  0.100 /  4.  0.100
  libswresample   2.  0.101 /  2.  0.101
[h264 @ 0x7fa896003c00] Increasing reorder buffer to 1
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Users/krueger/lesspain/samples/cameras/panasonic/gh4/panasonic_gh4_1080p_24_100mbit_lpcm_14s.MP4':
  Metadata:
major_brand : mp42
minor_version   : 1
compatible_brands: mp42avc1
creation_time   : 2015-08-02 17:43:47
  Duration: 00:00:14.50, start: 0.00, bitrate: 86357 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc,
bt709), 1920x1080 [SAR 1:1 DAR 16:9], 84676 kb/s, 24 fps, 24 tbr, 360k tbn,
48 tbc (default)
Metadata:
  creation_time   : 2015-08-02 17:43:47
  timecode: 14:21:03:11
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, 2
channels, s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2015-08-02 17:43:47
  timecode: 14:21:03:11
Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
  creation_time   : 2015-08-02 17:43:47
  timecode: 14:21:03:11
[mp4 @ 0x7fa896009000] Could not find tag for codec pcm_s16be in stream #1,
codec not currently supported in container
Output #0, mp4, to '/Users/krueger/tmp/gh4_lpcm_remuxed.mp4':
  Metadata:
major_brand : mp42
minor_version   : 1
compatible_brands: mp42avc1
encoder : Lavf57.28.102
Stream #0:0(und): Video: h264 ([33][0][0][0] / 0x0021), yuvj420p,
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 84676 kb/s, 24 fps, 24 tbr, 360k tbn,
360k tbc (default)
Metadata:
  creation_time   : 2015-08-02 17:43:47
  timecode: 14:21:03:11
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz,
stereo, 1536 kb/s (default)
Metadata:
  creation_time   : 2015-08-02 17:43:47
  timecode: 14:21:03:11
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument

Best regards,

Robert
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Re: [FFmpeg-user] Why is it possible to set a timecode with 24:00:00:00 and beyond?

2016-01-13 Thread Robert Krüger
On Wed, Jan 13, 2016 at 3:06 PM, Phil Rhodes 
wrote:

> I'd be a very careful here.
> Linear timecode (which is fundamentally an audio signal) cannot represent
> a value of more than two in the tens-of-hours position, because only two
> bits are allocated to it. Although other systems may be more forgiving, LTC
> is still very commonly used.
> Furthermore, the standards which specify various types of MXF refer to
> "timecode time addresses" which mirror LTC timecode in that they "consist
> of 64 data bits... 26 bits specify hours, minutes, seconds and frames...
> The combination of these values is the time address of the SMPTE 12M-1
> timecode value. "
> At least some types of MXF will therefore suffer the same limitation.
> The standard I have always referred to is SMPTE-12M (now two documents). I
> don't have access to my copy of 12M right now, and I can't recall if it
> specifically disallows time beyond 24 hours, but I would suggest that the
> safest approach to reject hours > 24 as it is far from universally
> supported. Whenever I have written implementations of 12M, which I have
> done at least three times, I have rejected hours > 24 on this basis.
> P
> __
>

Thanks a lot for the explanation.

Robert
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Re: [FFmpeg-user] Why is it possible to set a timecode with 24:00:00:00 and beyond?

2016-01-13 Thread Robert Krüger
On Wed, Jan 13, 2016 at 12:53 PM, Christoph Gerstbauer <
christophgerstba...@gmail.com> wrote:

> FFmpeg allows me to set a timecode of 24:00:00:00 and beyond.
>
> e.g "-timecode 24:00:00:00"
>
> When I make an MXF D10 for example, the value of 24:00:00:00 is still in
> the file.
> Most but not all programs ignore this and display 00:00:00:00 correctly.
>
> Here are some sample extractions of the timecode offset value in different
> mxf d10 files generated by ffmpeg:
>
> PAL 25fps:
>
> -timecode 12:34:56:11  -> in file: 11477b hex -> 1132411 dec
>
> -timecode 10:00:00:00 -> in file: 0dbba0 hex -> 90 dec
>
> -timecode 20:00:00:00 -> in file: 1b7740 hex -> 180 dec
>
> -timecode 23:59:59:24 -> in file: 20f57f hex -> 215 dec
>
> -timecode 24:00:00:00 -> in file: 20f580 hex -> 216 dec -> 24h
>
> -timecode 24:00:00:01 -> in file: 20f581 hex -> 2160001 dec
>
> -timecode 25:00:00:00 -> in file: 225510 hex -> 225 dec -> 25h
>
>
> NTSC 30fps
>
> -timecode 23:59:59;29 -> in file: 2782df hex -> 2589407 dec
>
> -timecode 24:00:00;00 -> in file: 2782e0 hex -> 2589408 dec
>
>
> Why is this possible to set a timecode after 23:59:59:XX?
> Is there a usecase?
>

I wasn't able to find any official spec that says what the limit for the
hour part is but since it in the general case does not define a time of day
but an offset, why limit it to 25 hours and not use the full 100? On a
side-note, I have not come across a real-world use case for a timecode of
that magnitude either.
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Re: [FFmpeg-user] Converting a 23.98p source to 29.97i ProRes w/interlaced 3:2 pulldown?

2015-10-27 Thread Robert Krüger
On Mon, Oct 26, 2015 at 8:42 PM, Mel Matsuoka  wrote:

>
>
> > On Oct 26, 2015, at 9:15 AM, Carl Eugen Hoyos  wrote:
> >
> > I believe (strongly) that what you like as output
> > framerate is not 29.97 but 3/1001
>
>
> Hi Carl,
>
> I guess I'm looking for the "why",  rather than the "what", as far as this
> syntax is concerned.
>
> In NTSC broadcast land, 29.97 frame rate is *always* defined and referred
> to as just that: 29.97. No professional video tool that I'm aware of
> requires the user to define that frame rate with the level of precision
> that dividing 3 by 1001 gives you.
>
> And I suppose this begs the question, where do the numbers 3 and 1001
> come from to begin with?
>
>
Just some heuristics. As explained by Moritz, the correct rate is
3/1001 due to its analogue heritage and most tools (software, cameras)
I know produce timestamps (and thus a frame rate) based on that. However,
some don't. E.g. Final Cut Pro 7 and earlier and most Apple Tools from that
time even used the rounded version 29.97 in the files they produced. In
theory the difference is so marginal, that it should not cause problems (it
takes more than nine hours for the rounding difference to produce 1 frame
A/V desync if a tool interpretes a 29.97 file as 3/1001 which
definitely happens in some cases). My experience has been, that the safer
side is to use 3/1001, simply because more tools use it and even
Apple's new tools (like FCPX use 3/1001 internally). You will probably
not run into these problems if you use 29.97 but if you do, errors might be
very subtle and hard to diagnose.
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Re: [FFmpeg-user] Deterministic results with atadenoise

2015-09-30 Thread Robert Krüger
On Tue, Sep 29, 2015 at 2:24 PM, Paul B Mahol  wrote:

> On 9/29/15, Robert Krueger  wrote:
> > Hi,
> >
> > is anything in atadenoise random? I am trying to get reproducible results
> > for an automated test setup but just running the same command line twice
> > produces a files which have differences when comparing them via
> framemd5. I
> > thought it might be multithreading but even with threads set to 1 this
> > differs.
> >
> > I tried a command line like this:
> >
> > ffmpeg -i noise_luma20.mov -threads 1 -vf atadenoise -c:v mpeg4 -q:v 1
> > noise_luma20_denoised_t1_2.mov
> >
> > Running it twice and then comparing framemd5s of the resulting files, I
> > find all frames to have different md5 hashes.
> > Thanks,
>
> I may know why this happens and will fix it ASAP.
>
>
fixed by a019149249695f7b812ead18bc51fc2d8df57499, i.e. produces identical
md5s regardless of threads setting.

Thanks!
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Re: [FFmpeg-user] Deterministic results with atadenoise

2015-09-29 Thread Robert Krüger
On Tue, Sep 29, 2015 at 2:23 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > ffmpeg -i noise_luma20.mov -threads 1 -vf atadenoise
> > -c:v mpeg4 -q:v 1 noise_luma20_denoised_t1_2.mov
>
> You have to use -threads 1 on the input file as well.
>
>
so the -threads on the input affects the threads used in the filters as
well?
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[FFmpeg-user] Deterministic results with atadenoise

2015-09-29 Thread Robert Krüger
Hi,

is anything in atadenoise random? I am trying to get reproducible results
for an automated test setup but just running the same command line twice
produces a files which have differences when comparing them via framemd5. I
thought it might be multithreading but even with threads set to 1 this
differs.

I tried a command line like this:

ffmpeg -i noise_luma20.mov -threads 1 -vf atadenoise -c:v mpeg4 -q:v 1
noise_luma20_denoised_t1_2.mov

Running it twice and then comparing framemd5s of the resulting files, I
find all frames to have different md5 hashes.
Thanks,

Robert
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Re: [FFmpeg-user] Display rgb statistics

2015-09-28 Thread Robert Krüger
On Fri, Sep 25, 2015 at 11:52 AM, Paul B Mahol  wrote:

> On 9/25/15, Robert Krüger  wrote:
> > Hi,
> >
> > what is the recommended way to display stats about the pixel values of
> rgb
> > files? The signalstats filter seems to only support yuv, at least the
> > documentation does not mention anything about R(GB)MAX/AVG and the filter
> > does not output anything either when specifying those.
>
> There is none, but I guess it should not be hard to add RGB mode or
> rgbstats filter.
>

Thanks for the info.
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[FFmpeg-user] Display rgb statistics

2015-09-25 Thread Robert Krüger
Hi,

what is the recommended way to display stats about the pixel values of rgb
files? The signalstats filter seems to only support yuv, at least the
documentation does not mention anything about R(GB)MAX/AVG and the filter
does not output anything either when specifying those.

I am asking this to make a bug report that does not require anything but
ffmpeg command line tools.

Thanks in advance,

Robert
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Re: [FFmpeg-user] MJPEG Quantization Matrix

2015-09-17 Thread Robert Krüger
On Wed, Sep 16, 2015 at 4:40 PM, Dsoccer16  wrote:

> Thank you very much.
> It works but there are two things I don't understand.
> Why can't I set all the coefficients equal 1? Is it possible to use a
> custom
> quantization matrix and at the same time control the coefficients using the
> q parameter?
>
>
If you succeed doing this, it would be nice if you could share your
findings here. I failed a few years ago to use custom matrices to drive
bitrate up higher than with qmin 1 qmax 1.
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Re: [FFmpeg-user] ct a video precisly

2015-09-16 Thread Robert Krüger
On Wed, Sep 16, 2015 at 12:26 PM, Henk D. Schoneveld 
wrote:

>
> On 16 Sep 2015, at 11:33, gag...@free.fr wrote:
>
> > Hello,
> >
> > I'm trying to cut a video to get 12s sequences. It needs to be accurate.
> Cutting is done on begin and end of a Group Of Pictures, a GOP.
> Splitting always occurs the way you mention, it is ‘about 12 seconds’
> What is, could be the reason, you need it absoute accurate ?
> One way around could be to 1st encode your source to all I-frame, key
> frames. This way you get a GOP-size of 1


That's not necessary. AFAIR ffmpeg will subclip accurately also for
non-i-frame-only codecs when transcoding the video. So if your aoutput is
not supposed to be i-frame-only (which is probably not the case), you would
have an unneeded encoding step (and thus quality loss), unless your
intermediate format is lossless of course. So, should it not suffice to
replace copy by a re-encoding, you could do that as a workaround (i.e.
transcode to ffv1-iframe-only and then segment that) but as I said, it
should not be required.
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Re: [FFmpeg-user] ct a video precisly

2015-09-16 Thread Robert Krüger
On Wed, Sep 16, 2015 at 11:33 AM,  wrote:

> Hello,
>
> I'm trying to cut a video to get 12s sequences. It needs to be accurate.
> But when I ask a 12 seconds sequence (not from the begining for this
> example) , I get around 12:01 seconds.
> The command line is :
> ffmpeg -ss 00:00:00.00 -i GOPR0151.mp4 -b:v 30150 -t 00:00:12.00 -acodec
> copy -vcodec copy GOPR0151_002.mp4
>
> and even, when I start from 00:02:12.00, the output lasts 12:33 seconds.
>
> Can anybody tell me who to get more accuracy, please ?
> I noticed that the bitrate is not the same in the orginal video than in
> the output. Is it important ?
> Here are the informations of the original video :
> ffprobe.exe GOPR0151.mp4
> ffprobe version N-74751-gb54e03c Copyright (c) 2007-2015 the FFmpeg
> developers
>   built with gcc 4.9.3 (GCC)
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
> --enable-gnu
> tls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b
> --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme
> --enab
> le-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
> --enable
> -libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
> --enable-libspeex --enable-libtheora --enable-libtwolame
> --enable-libvidstab --ena
> ble-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
> --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
> --enable-libx265 --e
> nable-libxavs --enable-libxvid --enable-lzma --enable-decklink
> --enable-zlib
>   libavutil  54. 31.100 / 54. 31.100
>   libavcodec 56. 59.100 / 56. 59.100
>   libavformat56. 40.101 / 56. 40.101
>   libavdevice56.  4.100 / 56.  4.100
>   libavfilter 5. 40.100 /  5. 40.100
>   libswscale  3.  1.101 /  3.  1.101
>   libswresample   1.  2.101 /  1.  2.101
>   libpostproc53.  3.100 / 53.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'GOPR0151.mp4':
>   Metadata:
> major_brand : avc1
> minor_version   : 0
> compatible_brands: avc1isom
> creation_time   : 2014-10-25 09:19:36
>   Duration: 00:17:24.30, start: 0.00, bitrate: 30150 kb/s
> Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc,
> bt709), 1280x960 [SAR 1:1 DAR 4:3], 30002 kb/s, 100 fps, 100 tbr, 180k tbn,
>  200 tbc (default)
> Metadata:
>   creation_time   : 2014-10-25 09:19:36
>   handler_name:  GoPro AVC
>   encoder : GoPro AVC encoder
> Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
> stereo, fltp, 128 kb/s (default)
> Metadata:
>   creation_time   : 2014-10-25 09:19:36
>   handler_name:  GoPro AAC
>
>
> And the output :
> ffprobe.exe GOPR0151_002.mp4
> ffprobe version N-74751-gb54e03c Copyright (c) 2007-2015 the FFmpeg
> developers
>   built with gcc 4.9.3 (GCC)
>   configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
> --enable-gnu
> tls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b
> --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme
> --enab
> le-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame
> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg
> --enable
> -libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr
> --enable-libspeex --enable-libtheora --enable-libtwolame
> --enable-libvidstab --ena
> ble-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
> --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
> --enable-libx265 --e
> nable-libxavs --enable-libxvid --enable-lzma --enable-decklink
> --enable-zlib
>   libavutil  54. 31.100 / 54. 31.100
>   libavcodec 56. 59.100 / 56. 59.100
>   libavformat56. 40.101 / 56. 40.101
>   libavdevice56.  4.100 / 56.  4.100
>   libavfilter 5. 40.100 /  5. 40.100
>   libswscale  3.  1.101 /  3.  1.101
>   libswresample   1.  2.101 /  1.  2.101
>   libpostproc53.  3.100 / 53.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'GOPR0151_002.mp4':
>   Metadata:
> major_brand : isom
> minor_version   : 512
> compatible_brands: isomiso2avc1mp41
> encoder : Lavf56.40.101
>   Duration: 00:00:12.33, start: 0.309333, bitrate: 30273 kb/s
> Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc,
> bt709), 1280x960 [SAR 1:1 DAR 4:3], 30187 kb/s, 100 fps, 100 tbr, 180k tbn,
>  200 tbc (default)
> Metadata:
>   handler_name: VideoHandler
> Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
> stereo, fltp, 128 kb/s (default)
> Metadata:
>   handler_name: SoundHandler
>
>
> Thanks in advance
>
> Fridu
> ___
>

What you try will not work, i.e. accurately cut su

Re: [FFmpeg-user] Application provided invalid, non monotonically increasing dts to muxer in stream

2015-09-16 Thread Robert Krüger
On Tue, Sep 15, 2015 at 9:56 PM, MrNice  wrote:

>
> On 14/09/15 13:54, MrNice wrote:
> > Hello,
> >
> > When I run the following command, I get the error messages in red:
> > Application provided invalid, non monotonically increasing dts to muxer
> > in stream
> >
> > Theses messages are in group of 4-5 and happen around every minute.
> > Sometime there is frames drop, see second part output bellow 4 drops are
> > dropped, or no drop.
> > Drops seem to lead to audio/video de-sync.
> >
> > How to fix that?
> > Thanks
> >
>
> DTS is for Display Time Stamp (nothing to do with DTS audio codec)
>

No, DTS stands for "decoding timestamp" (time when this buffer will be
decoded on the same time axis used for presentation time) in this case as
opposed to "presentation timestamp" (time when the data will be displayed
in the case of video or played in the case of audio).
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Re: [FFmpeg-user] Failing to build ffmpeg with ffplay support

2015-09-11 Thread Robert Krüger
On Fri, Sep 11, 2015 at 4:17 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > It may be that ffplay for OSX is currently not
> > available.
>
> I think this is not correct:
> https://evermeet.cx/ffmpeg/
> (It also failed for me but I didn't try
> very hard.)
>
>
Yes, I found that but did they specify which OSX version they built on? I
didn't see that on the page.
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Re: [FFmpeg-user] 422 to 420 H264 : the chroma is not good

2015-09-11 Thread Robert Krüger
On Fri, Sep 11, 2015 at 3:37 PM, Alexandre Fressenon <
alexan...@fressenon.com> wrote:

> Hi all,
> I have a chroma problem with a conversion.
>
> As input I have 2 files : 422 Uncompressed (yuv) or AppleProRes 422 (yuv),
> and I want to create a new file with H264 420 (yuv).
>
> But when I try to convert the sources from 422 to 420 I still have a
> problem with the chroma, and this elements is very important for me.
> The problem is not the H264 because if I already have a 420 uncompressed,
> the conversion is perfect, but my input is a 422, and QuickTime can't read
> a H264 422, so the final file has to be a H264 420…
>
> I know some software can create a H264 / 420 with the same chroma, and I
> would like do this part with ffmpeg.
>
>
could you describe your problem with the chroma output more precisely?
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Re: [FFmpeg-user] Failing to build ffmpeg with ffplay support

2015-09-11 Thread Robert Krüger
On Fri, Sep 11, 2015 at 2:26 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > Ist Version 2.0 of SDL not supported
>
> It is not supported iirc.
>
>
thanks for the info. it seems libsdl1.2.x no longer compiles on OSX.

In file included from ./src/video/quartz/SDL_QuartzEvents.m:24:
./src/video/quartz/SDL_QuartzVideo.h:94:5: error: unknown type name
  'CGDirectPaletteRef'

This seems to be the case from OS 10.9 on (
http://hg.libsdl.org/SDL/rev/e9466ead70e5).

Even fixing that by the described hack, there seem to be further errors. It
may be that ffplay for OSX is currently not available. Maybe I am missing
something and a fellow OSX ffmpeg user can enlighten me.
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[FFmpeg-user] Failing to build ffmpeg with ffplay support

2015-09-11 Thread Robert Krüger
Hi,

I am trying to build ffmpeg with ffplay support for the first time for a
while and fail to do it.

I installed libsdl 2.0.3 from source using make install (Mac OS 10.10
btw.), I have pkg-config installed (0.28) and config.h contains

#define HAVE_SDL 0

and no ffplay is built.

Ist Version 2.0 of SDL not supported or where else can I look for clues?

Thanks in advance,

Robert
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Re: [FFmpeg-user] Frame interpolation in FFmpeg

2015-08-29 Thread Robert Krüger
On Sat, Aug 29, 2015 at 5:17 PM, Hu,Dapeng  wrote:

> Hi guys:
>
> Can any FFmpeg experts tell me if FFmpeg can do any sophisticated frame
> interpolation? If so, what would be some suggested settings to try?
> I am developing a media player based on FFmpeg and want to improve smooth
> through frame interpolation.
>
>
the only frame interpolation method available currently AKAIK is the
framerate filter that was just committed. It uses frame blending (i.e.
weighted averaging) just as many editing programs do. More sophisticated
methods are not available and are probably not fast enough for use in a
player anyway.

Robert
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Re: [FFmpeg-user] scale filter does not do anything when only color matrix is changed

2015-08-26 Thread Robert Krüger
On Wed, Aug 26, 2015 at 11:28 AM, Paul B Mahol  wrote:

> On 8/26/15, Robert Krueger  wrote:
> > Hi,
> >
> > I created two files using these command lines:
> >
> > ffmpeg -i fcp7_xdcam_ex_1080_25p_1s.mov -vf
> > scale=in_color_matrix=bt709:out_color_matrix=bt601 -c:v ffv1 -an
> > scale_test_bt601.nut
> >
> > ffmpeg -i fcp7_xdcam_ex_1080_25p_1s.mov -c:v ffv1 -an
> > scale_test_unscaled.nut
> >
> > Using framemd5 I verified that the resulting output is identical. Is this
> > some sort of optmization of swsscale (i.e. not doing anything when size
> and
> > pixel format remain unchanged)? Can it be forced to do the conversion
> > anyway? I know that there is also vf_colormatrix but that is much more
> > limited than swscale and it's not LGPL.
>
> Looking at code and if I'm not missing something it's just not implemented
> to
> be used like that, perhaps better to fill bug report.
>

Thanks. I created ticket 4805. This also happens when using swsscale
directly using the api, which is where I ran into it. I just used the
command line to verify this. So the ticket is more general.
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[FFmpeg-user] scale filter does not do anything when only color matrix is changed

2015-08-26 Thread Robert Krüger
Hi,

I created two files using these command lines:

ffmpeg -i fcp7_xdcam_ex_1080_25p_1s.mov -vf
scale=in_color_matrix=bt709:out_color_matrix=bt601 -c:v ffv1 -an
scale_test_bt601.nut

ffmpeg -i fcp7_xdcam_ex_1080_25p_1s.mov -c:v ffv1 -an
scale_test_unscaled.nut

Using framemd5 I verified that the resulting output is identical. Is this
some sort of optmization of swsscale (i.e. not doing anything when size and
pixel format remain unchanged)? Can it be forced to do the conversion
anyway? I know that there is also vf_colormatrix but that is much more
limited than swscale and it's not LGPL.

Thanks in advance,

Robert
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[FFmpeg-user] Possible to set color_range on the command line for mjpeg?

2015-08-25 Thread Robert Krüger
Hi,

for the purpose of debugging something in the libraries and submitting a
corresponding bug report, I would like to specify the color_range of the
codec context on the command line, i.e. in this case I want to create an
mjpeg file with full range without using the deprecated yuvj pixel formats.

I tried this here:

ffmpeg -i
~/lesspain/samples/software/fcp_7/xdcam_ex/fcp7_xdcam_ex_1080_25p_1s.mov
-vf scale=out_range=full -c:v mjpeg -an -pix_fmt yuv444p
mjpeg-fullrange-ffmpeg-noyuvj.mov
ffmpeg version N-74034-gce46627 Copyright (c) 2000-2015 the FFmpeg
developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM
3.6.0svn)
  configuration: --enable-gpl --enable-libx264
  libavutil  54. 29.100 / 54. 29.100
  libavcodec 56. 55.100 / 56. 55.100
  libavformat56. 40.101 / 56. 40.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 29.100 /  5. 29.100
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc53.  3.100 / 53.  3.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fddd181bc00] nclc: pri 1 trc 1 matrix
1Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Users/krueger/lesspain/samples/software/fcp_7/xdcam_ex/fcp7_xdcam_ex_1080_25p_1s.mov':
  Metadata:
major_brand : qt
minor_version   : 537199360
compatible_brands: qt
creation_time   : 2013-12-11 15:38:12
  Duration: 00:00:01.00, start: 0.00, bitrate: 25365 kb/s
Stream #0:0(eng): Video: mpeg2video (Main) (xdve / 0x65766478),
yuv420p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 23773 kb/s, 25 fps, 25
tbr, 25 tbn, 50 tbc (default)
Metadata:
  creation_time   : 2013-12-11 15:38:12
  handler_name: Apple Alias-Datensteuerung
  encoder : XDCAM EX 1080p25 (35 Mb/s VBR)
  timecode: 01:00:00:00
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, 2
channels, s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2013-12-11 15:38:12
  handler_name: Apple Alias-Datensteuerung
Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
  creation_time   : 2013-12-11 15:38:16
  handler_name: Apple Alias-Datensteuerung
  timecode: 01:00:00:00
Incompatible pixel format 'yuv444p' for codec 'mjpeg', auto-selecting
format 'yuvj444p'
[swscaler @ 0x7fddd104a000] deprecated pixel format used, make sure you did
set range correctly
Output #0, mov, to 'mjpeg-fullrange-ffmpeg-noyuvj.mov':
  Metadata:
major_brand : qt
minor_version   : 537199360
compatible_brands: qt
encoder : Lavf56.40.101
Stream #0:0(eng): Video: mjpeg (jpeg / 0x6765706A), yuvj444p(pc),
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 12800 tbn, 25 tbc
(default)
Metadata:
  creation_time   : 2013-12-11 15:38:12
  handler_name: Apple Alias-Datensteuerung
  timecode: 01:00:00:00
  encoder : Lavc56.55.100 mjpeg
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg2video (native) -> mjpeg (native))
Press [q] to stop, [?] for help
frame=   24 fps=0.0 q=24.8 size=2066kB time=00:00:00.96
bitrate=17626.5kbitsframe=   25 fps=0.0 q=24.8 Lsize=2147kB
time=00:00:01.00 bitrate=17589.0kbits/s
video:2146kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.067178%

but obviously the mjpeg encoder refuses the pixel format although it should
not do this if the deprecated pixel formats are really not to be used. Am I
missing something or is the mjpeg encoder simply not capable yet of working
with color_range instead of the deprecated yuvj pixel formats?

Thanks in advance,

Robert
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Re: [FFmpeg-user] Misdetected color info for h264 mov file

2015-08-20 Thread Robert Krüger
On Thu, Aug 20, 2015 at 8:30 AM, Robert Krüger  wrote:

>
>
> On Wed, Aug 19, 2015 at 8:08 PM, Carl Eugen Hoyos 
> wrote:
>
>> Robert Krüger  lesspain.de> writes:
>>
>> > encoder : Lavf56.23.100
>>
>> How was the input file created?
>>
>> using the apple h.264 encoder for encoding and libavformat for muxing. I
> presume something is at least unusual in the file and I am trying to deduce
> what from analyzing the file but have not found anything, that's why I ask
> about ffmpeg internals of h.264/mov metadata detection as that might give
> me a clue. I thought I remembered that ffmpeg's policy was to give
> container metadata higher priority than bitstream metadata but that would
> contradict what I am observing.
>

I just enabled logging and could verify that the colr atom vaues are
correctly parsed:

[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fa82b802a00] nclc: pri 1 trc 1 matrix 1Input
#0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Users/krueger/tmp/colortest/h264-pipeline-out.mov':

so it seems to indicate that something else overrides this but I am unable
to find in the code where that might happen.
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Re: [FFmpeg-user] Misdetected color info for h264 mov file

2015-08-19 Thread Robert Krüger
On Wed, Aug 19, 2015 at 8:08 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > encoder : Lavf56.23.100
>
> How was the input file created?
>
> using the apple h.264 encoder for encoding and libavformat for muxing. I
presume something is at least unusual in the file and I am trying to deduce
what from analyzing the file but have not found anything, that's why I ask
about ffmpeg internals of h.264/mov metadata detection as that might give
me a clue. I thought I remembered that ffmpeg's policy was to give
container metadata higher priority than bitstream metadata but that would
contradict what I am observing.
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[FFmpeg-user] Misdetected color info for h264 mov file

2015-08-19 Thread Robert Krüger
Hi,

I have a case where ffprobe does not detect color information (matrix,
primaries) that is in the mov file. I have checked that in the file the
respective values of the indices in the colr atom are correctly set to
ITU-R BT.709 but ffprobe does not detect that.

./ffprobe -show_format -show_streams ~/tmp/colortest/h264-pipeline-out.mov
| grep color
ffprobe version N-74034-gce46627 Copyright (c) 2007-2015 the FFmpeg
developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM
3.6.0svn)
  configuration: --enable-gpl --enable-libx264
  libavutil  54. 29.100 / 54. 29.100
  libavcodec 56. 55.100 / 56. 55.100
  libavformat56. 40.101 / 56. 40.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 29.100 /  5. 29.100
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc53.  3.100 / 53.  3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from
'/Users/krueger/tmp/colortest/h264-pipeline-out.mov':
  Metadata:
major_brand : qt
minor_version   : 512
compatible_brands: qt
creation_time   : 2015-08-06 16:31:40
encoder : Lavf56.23.100
  Duration: 00:00:01.00, start: 0.00, bitrate: 10549 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv),
1920x1080 [SAR 1:1 DAR 16:9], 10485 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
(default)
Metadata:
  creation_time   : 2015-08-06 16:31:40
  handler_name: DataHandler
  timecode: 01:00:00:00
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, fltp, 74 kb/s (default)
Metadata:
  creation_time   : 2015-08-06 16:31:40
  handler_name: DataHandler
Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
Metadata:
  creation_time   : 2015-08-06 16:31:40
  handler_name: DataHandler
  timecode: 01:00:00:00
Unsupported codec with id 0 for input stream 2
color_range=tv
color_space=unknown
color_transfer=unknown
color_primaries=unknown

(BTW, I get the same result when retrieving the info via libav api)

Does avformat_find_stream_info maybe use bitstream info over container info
for h.264?

What could be wrong in the file to trigger that behaviour? I have other
h264 mov files with the same colr atom values that are detected correctly.

Thanks for any insights and best regards,

Robert
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Re: [FFmpeg-user] Extracting video levels on the command line

2015-08-04 Thread Robert Krüger
On Tue, Aug 4, 2015 at 4:11 PM, Dave Rice  wrote:

>
> > On Aug 4, 2015, at 6:22 AM, Robert Krüger  wrote:
> >
> > On Mon, Aug 3, 2015 at 10:49 PM, Paul B Mahol  wrote:
> >
> >> Dana 3. 8. 2015. 22:28 osoba "Moritz Barsnick" 
> napisala
> >> je:
> >>>
> >>> On Mon, Aug 03, 2015 at 21:55:59 +0200, Robert Krüger wrote:
> >>>> Is there any other way than looking at each frame's histogram (using
> >> the
> >>>> histogram filter) and counting pixels?
> >>>
> >>> It should probably be easy to write such a filter.
> >>>
> >>> Have you looked at signalstats? It does seem to have the capability to
> >>> identify various statistics on the Y, U, V planes, but reports them
> >>> only in metadata, not in console output (IIUC).
> >>
> >> ffprobe can output any metadata
> >
> >
> > Thanks. I overlooked signalstats. It does not give me quite what I want,
> > because it's per frame but it is a lot better than abusing the histogram
> > filter for that. No idea if this makes sense but maybe this total min/max
> > thing could even be implemented generically in ffprobe for any
> > number-valued metadata but that's just an idea.
>
> A summary section for signalstats would make sense. As a workaround you
> could output from the filter to a csv and then evaluate the csv:
>
> ffprobe -f lavfi movie=file.mov,signalstats -show_entries
> frame_tags=lavfi.signalstats.YMIN,lavfi.signalstats.YMAX -of csv > stats.csv
>
>
yes, scripting that is easy in csv or xml. I will probably do that (and
then hope that someone adds this functionality eventually, I know, patch
welcome :-S)
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Re: [FFmpeg-user] Extracting video levels on the command line

2015-08-04 Thread Robert Krüger
On Mon, Aug 3, 2015 at 10:49 PM, Paul B Mahol  wrote:

> Dana 3. 8. 2015. 22:28 osoba "Moritz Barsnick"  napisala
> je:
> >
> > On Mon, Aug 03, 2015 at 21:55:59 +0200, Robert Krüger wrote:
> > > Is there any other way than looking at each frame's histogram (using
> the
> > > histogram filter) and counting pixels?
> >
> > It should probably be easy to write such a filter.
> >
> > Have you looked at signalstats? It does seem to have the capability to
> > identify various statistics on the Y, U, V planes, but reports them
> > only in metadata, not in console output (IIUC).
> >
>
> ffprobe can output any metadata


Thanks. I overlooked signalstats. It does not give me quite what I want,
because it's per frame but it is a lot better than abusing the histogram
filter for that. No idea if this makes sense but maybe this total min/max
thing could even be implemented generically in ffprobe for any
number-valued metadata but that's just an idea.
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[FFmpeg-user] Extracting video levels on the command line

2015-08-03 Thread Robert Krüger
Hi,

is there an easy way to extract things like min/max of each video component
for a video using the command line? I first thought, there may be a vstats
filter that more or less does what astats does for audio but that does not
seem to be the case.

Is there any other way than looking at each frame's histogram (using the
histogram filter) and counting pixels?

Robert
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Re: [FFmpeg-user] Converting .mov files playable on html5 players..but very slow

2015-06-27 Thread Robert Krüger
On Sat, Jun 27, 2015 at 8:30 AM, Dani A 
wrote:

> I tried to convert a .mov file to mp4 using the above command and I
> recieved the below errors. what is wrong?
>
> [root@test video]# ffmpeg -i MVI_0797.MOV  -c:v copy  -c:a copy
> -movflags faststart MVI_0797-fast.MP4
> ffmpeg version 2.6.2-   http://johnvansickle.com/ffmpeg/Copyright (c)
> 2000-2015 the FFmpeg developers
>   built with gcc 4.9.2 (Debian 4.9.2-10)
>   configuration: --enable-gpl --enable-version3 --disable-shared
> --disable-debug --enable-runtime-cpudetect --enable-libmp3lame
> --enable-libx264 --enable-libx265 --enable-libwebp --enable-libspeex
> --enable-libvorbis --enable-libvpx --enable-libfreetype --enable-fontconfig
> --enable-libxvid --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc
> --enable-gray --enable-libopenjpeg --enable-libopus --enable-libass
> --enable-gnutls --enable-libvidstab --enable-libsoxr --cc=gcc-4.9
>   libavutil  54. 20.100 / 54. 20.100
>   libavcodec 56. 26.100 / 56. 26.100
>   libavformat56. 25.101 / 56. 25.101
>   libavdevice56.  4.100 / 56.  4.100
>   libavfilter 5. 11.102 /  5. 11.102
>   libswscale  3.  1.101 /  3.  1.101
>   libswresample   1.  1.100 /  1.  1.100
>   libpostproc53.  3.100 / 53.  3.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'MVI_0797.MOV':
>   Metadata:
> major_brand : qt
> minor_version   : 537331968
> compatible_brands: qt  CAEP
> creation_time   : 2014-06-15 02:43:16
>   Duration: 00:00:01.00, start: 0.00, bitrate: 36419 kb/s
> Stream #0:0(eng): Video: h264 (Constrained Baseline) (avc1 /
> 0x31637661), yuvj420p(pc, bt709), 1920x1080, 34098 kb/s, 23.98 fps, 23.98
> tbr, 24k tbn, 48k tbc (default)
> Metadata:
>   creation_time   : 2014-06-15 02:43:16
> Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz,
> stereo, s16, 1536 kb/s (default)
> Metadata:
>   creation_time   : 2014-06-15 02:43:16
> File 'MVI_0797-fast.MP4' already exists. Overwrite ? [y/N] y
>



> [mp4 @ 0xbe8a6c0] Could not find tag for codec pcm_s16le in stream #1,
> codec not currently supported in container
>


pcm audio is not supported by mp4 and not a good idea for web use anyway.
try mp3 or aac.
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Re: [FFmpeg-user] Audio Tapes - How to restore audio quality

2015-05-30 Thread Robert Krüger
On Sat, May 30, 2015 at 4:57 AM, jd1008  wrote:

> Audacity???
>
> I have never been able to make audacity improve
> the signal to noise ratio, nor eliminate clicks and pops.
>
> How do you do that?
>
>
Depending on the kind of noise you have and if you have enough isolated
noise to generate a good noise profile Audacity does help in some cases but
it's not exactly a cutting edge noise reducer comparable to commercial
tools like Adobe Audition but this is a bit OT here. But I seriously doubt
it will do anything about clicks and pops.
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Re: [FFmpeg-user] Audio Tapes - How to restore audio quality

2015-05-30 Thread Robert Krüger
On Sat, May 30, 2015 at 4:34 AM, jd1008  wrote:

> I need to transfer a bunch of audio tapes to digital media.
> But I would like to
> 1. improve the signal to noise ratio.
> 2. get rid of clicks and pops.
>
> Is there a way to do this using ffmpeg?


I would say no. I am not aware of any audio filters that would help in this
case (see http://ffmpeg.org/ffmpeg-filters.html#Audio-Filters).
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Re: [FFmpeg-user] Smooth frame rate reduction?

2015-03-31 Thread Robert Krüger
On Tue, Mar 31, 2015 at 5:58 PM, Carl Eugen Hoyos  wrote:

> Robert Krüger  lesspain.de> writes:
>
> > I think Mark mentioned once the intent to port it to
> > ffmpeg but I am not sure. If you're in a hurry, you
> > can apply it to ffmbc and use that (I hope I am
> > allowed to say that here but I would of course
> > favour this filter to be ported to ffmpeg).
>
> Where can I find your patch?
>
>
I know I practically begged for this response ;-)
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Re: [FFmpeg-user] Smooth frame rate reduction?

2015-03-31 Thread Robert Krüger
On Tue, Mar 31, 2015 at 5:22 PM, Petr Tresnak  wrote:

> Hi,
> I need to reduce frame-rate from my video shoot by mobile ( variable
> frame-rate about 29 fps) to 25 fps.
> As you know using -r 25 just drops every 6th frame resulting in choppy
> (jerky) footage.
>
> Is there any better option how to do it more smoothly (blending etc.)?
> Maintaining the same length of course:)
>
> Thanks
> Peter
>
> From: peter.che...@quick.cz
> To: ffmpeg-user@ffmpeg.org
> Subject: 90k tbn vs 30k tbn
> Date: Tue, 24 Feb 2015 16:46:15 +
>
>
Mark Himsley has written a filter for that which unfortunately is currently
only available as an ffmbc patch (http://mdsh.com/wiki/jsp/Wiki?FFmbc+0.7
 , look for "framerate converter") but it does exactly what you want, i.e.
frame blending and has decent results.

I think Mark mentioned once the intent to port it to ffmpeg but I am not
sure. If you're in a hurry, you can apply it to ffmbc and use that (I hope
I am allowed to say that here but I would of course favour this filter to
be ported to ffmpeg).

HTH

Robert
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Re: [FFmpeg-user] commercial application compatible with FFmpeg

2015-03-28 Thread Robert Krüger
>
> Oops, sorry for top-posting.
>
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Re: [FFmpeg-user] commercial application compatible with FFmpeg

2015-03-28 Thread Robert Krüger
the only way to be sure is to consult a lawyer but in my non-lawyer opinion
if you do not distribute any ffmpeg code but only allow your software to
use a pre-installed ffmpeg binary via command line, I don't see what
problems you should run into. It is then the user's choice to install/use
the ffmpeg version with a certain list of codecs/features enabled which may
or may not require additional patent licensing (e.g. like almost all mpeg
codecs) for certain kinds of uses, so it should not be your problem legally.


On Sat, Mar 28, 2015 at 5:13 AM, Juan Urrego  wrote:

> Hi, I'm currently developing a commercial software product, and I would
> like to know if there are any legal problems with licenses or patents if I
> make my software compatible with FFmpeg. I do not plan to distribute the
> ffmpeg.exe file, so that the end user makes on his own. I only provided
> support for my software using command lines provided by FFmpeg. I wonder if
> you might have a problem with that. In advance thank you very much for
> attention.
> Sorry if there are any errors in the text, I'm Hispanic and I use google
> translator to make this message.
>
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>



-- 
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Managing Partner
Lesspain GmbH & Co. KG

www.lesspain-software.com
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Re: [FFmpeg-user] pitch corecting audio?

2015-03-26 Thread Robert Krüger
On Thu, Mar 26, 2015 at 3:03 PM, Moritz Barsnick  wrote:

> On Wed, Mar 25, 2015 at 20:20:04 -0700, James Heliker wrote:
> > If I could access the original 23.976 source audio and apply "-af
> > atempo=25*1001/24000", would that accomplish the job without affecting
> > pitch in the first place?
>
> Indeed it should!
>
> You may or may not need to use the setpts filter if you use my first
> example, see here:
>
> http://superuser.com/a/611927/361295
>
> (You can experiment and tell us your results.)
>
> > Then I could merge the 25fps video with my correctly sped-up audio
>
> Possibly with my mentioned caveat regarding the A/V sync.
>

I have done that for the exact same use case in the past for a music video
(i.e. very tough A/V sync requirements) with perfect results, so I would
expect this to work.
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Re: [FFmpeg-user] Test if ProRes 4444 are real 12bit

2015-01-25 Thread Robert Krüger
On Sat, Jan 24, 2015 at 7:11 PM, Christoph Gerstbauer <
christophgerstba...@gmail.com> wrote:

> Hello,
>
> I want to test ifthe ProRes of the ARRI Alexa are real 12 Bit? Is
> there method to proove it with ffmpeg?
>
> AS far as I know ffmpeg is only ready to open prores with yuv422p10le and
> yuv444p10le (10bit), but no 12bit.
>


Last time I checked/tried, it wasn't possible. The best I got was a
statement by one developer that, looking at the code, it would be
theoretically possible to support more than 10bit, so my understanding is
that ffmpeg currently destroys any headroom beyond 10bit but since I am not
qualified enough to read the code, I cannot be 100% sure.
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Re: [FFmpeg-user] XAVC to Quicktime MOV

2014-11-28 Thread Robert Krüger
On Fri, Nov 28, 2014 at 4:06 PM, Rens Dijkshoorn  wrote:
>
> The ffmpeg-2.4.3 version is broken for this command
>
> ffmpeg -i AVC100CBG_1920_1080_H422IP.MXF -map 0:0 -map 0:1 -map 0:2 -c copy 
> -vtag ai12 FFMPEG.MOV
>

Just curious but why ai12 when it's XAVC? Isn't aivx the correct
fourCC in this case?

http://ffmpeg.org/pipermail/ffmpeg-user/2013-October/018055.html
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[FFmpeg-user] Start time/duration question when transcoding to WMV/WMA

2014-10-14 Thread Robert Krüger
I observed that start times and durations changed when transcoding
whatever input (tried a few) to wmv/wma in a wmv container.

Here's a sample command line with output:

ffmpeg -f lavfi -i testsrc=s=640x480:d=1 -f lavfi -i
sine=frequency=1000:sample_rate=48000:duration=1 -c:v wmv2 -c:a wmav2
~/tmp/out.wmv
ffmpeg version N-66701-g33db995 Copyright (c) 2000-2014 the FFmpeg developers
  built on Oct  8 2014 21:12:12 with Apple LLVM version 6.0
(clang-600.0.51) (based on LLVM 3.5svn)
  configuration: --enable-gpl --enable-libx264
--extra-cflags=-I/Users/krueger/src/x264
--extra-ldflags=-L/Users/krueger/src/x264
  libavutil  54.  9.100 / 54.  9.100
  libavcodec 56.  3.101 / 56.  3.101
  libavformat56.  7.104 / 56.  7.104
  libavdevice56.  1.100 / 56.  1.100
  libavfilter 5.  1.103 /  5.  1.103
  libswscale  3.  1.100 /  3.  1.100
  libswresample   1.  1.100 /  1.  1.100
  libpostproc53.  1.100 / 53.  1.100
Input #0, lavfi, from 'testsrc=s=640x480:d=1':
  Duration: N/A, start: 0.00, bitrate: N/A
Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24,
640x480 [SAR 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
Input #1, lavfi, from 'sine=frequency=1000:sample_rate=48000:duration=1':
  Duration: N/A, start: 0.00, bitrate: 768 kb/s
Stream #1:0: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
Output #0, asf, to '/Users/krueger/tmp/out.wmv':
  Metadata:
WM/EncodingSettings: Lavf56.7.104
Stream #0:0: Video: wmv2 (WMV2 / 0x32564D57), yuv420p, 640x480
[SAR 1:1 DAR 4:3], q=2-31, 200 kb/s, 25 fps, 1k tbn, 25 tbc
Metadata:
  encoder : Lavc56.3.101 wmv2
Stream #0:1: Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, mono,
fltp, 128 kb/s
Metadata:
  encoder : Lavc56.3.101 wmav2
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> wmv2 (native))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> wmav2 (native))
Press [q] to stop, [?] for help
frame=   25 fps=0.0 q=1.7 Lsize= 107kB time=00:00:01.00 bitrate=
877.6kbits/s
video:85kB audio:16kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 6.368599%

This should result in a file of one second length but this is what
ffprobe tells me:

see ffprobe output:

ffprobe -show_streams -i ~/tmp/out.wmv
ffprobe version N-66701-g33db995 Copyright (c) 2007-2014 the FFmpeg developers
  built on Oct  8 2014 21:12:12 with Apple LLVM version 6.0
(clang-600.0.51) (based on LLVM 3.5svn)
  configuration: --enable-gpl --enable-libx264
--extra-cflags=-I/Users/krueger/src/x264
--extra-ldflags=-L/Users/krueger/src/x264
  libavutil  54.  9.100 / 54.  9.100
  libavcodec 56.  3.101 / 56.  3.101
  libavformat56.  7.104 / 56.  7.104
  libavdevice56.  1.100 / 56.  1.100
  libavfilter 5.  1.103 /  5.  1.103
  libswscale  3.  1.100 /  3.  1.100
  libswresample   1.  1.100 /  1.  1.100
  libpostproc53.  1.100 / 53.  1.100
Input #0, asf, from '/Users/krueger/tmp/out.wmv':
  Metadata:
encoder : Lavf56.7.104
  Duration: 00:00:01.09, start: 0.00, bitrate: 808 kb/s
Stream #0:0: Video: wmv2 (WMV2 / 0x32564D57), yuv420p, 640x480,
SAR 1:1 DAR 4:3, 25 fps, 25 tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: wmav2 (a[1][0][0] / 0x0161), 48000 Hz, 1
channels, fltp, 128 kb/s
[STREAM]
index=0
codec_name=wmv2
codec_long_name=Windows Media Video 8
profile=unknown
codec_type=video
codec_time_base=1/1000
codec_tag_string=WMV2
codec_tag=0x32564d57
width=640
height=480
has_b_frames=0
sample_aspect_ratio=1:1
display_aspect_ratio=4:3
pix_fmt=yuv420p
level=-99
color_range=N/A
color_space=unknown
timecode=N/A
id=N/A
r_frame_rate=25/1
avg_frame_rate=25/1
time_base=1/1000
start_pts=43

start_time=0.043000
^
duration_ts=1043
duration=1.043000
^^^

bit_rate=N/A
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
[/STREAM]
[STREAM]
index=1
codec_name=wmav2
codec_long_name=Windows Media Audio 2
profile=unknown
codec_type=audio
codec_time_base=1/48000
codec_tag_string=a[1][0][0]
codec_tag=0x0161
sample_fmt=fltp
sample_rate=48000
channels=1
channel_layout=unknown
bits_per_sample=0
id=N/A
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/1000
start_pts=0
start_time=0.00
duration_ts=1043
duration=1.043000

bit_rate=128000
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
[/STREAM]

Is this due to encoder delay of WMA (2064 samples?) and i

[FFmpeg-user] Help with command line syntax change for flags

2014-08-01 Thread Robert Krüger
Hi,

I have an old command line using "-flags2 +ivlc+sgop+non_linear_q".
The syntax was changed a while ago, I think with mpeg-specific flags
moving to -mpv_flags.

I have not found (maybe I just missed it) in the docs how that would
translate into current syntax. Could someone help me out?

Thanks,

Robert
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[FFmpeg-user] new website

2014-07-27 Thread Robert Krüger
congratulations on the new website! Well done.

Regards,

Robert
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Re: [FFmpeg-user] Possibility to dump AVCodecContext, AVFormatContext fields for a command line?

2014-07-18 Thread Robert Krüger
On Jul 18, 2014 6:32 PM, "Carl Eugen Hoyos"  wrote:
>
> Robert Krüger  lesspain.de> writes:
>
> > >> is there any way to see the AVCodecContext, AVFormatContext fields
> > >> used for a given command line other than using a debugger? Maybe a
> > >> compile flag of some sort?
> > >
> > > It is currently not easily possible but could you explain your
> > > usecase?
> > > This would make a patch more likely imo.
> > >
> >
> > The use case is simply when working with the api doing
> > encoding/muxing and running into a problem in many
> > cases the key is some field value in one of those
> > contexts not being set correctly, where correctly is
> > not always obvious (see the posting on libav-user
> > yesterday where you helped me).
>
> I don't think this new feature would have helped you
> in that case.
>
> I am a little surprised that you always mention
> gdb, I personally find printf much easier to use.
> (But that may be just me.)
I have done that as well a few times when I knew what I was looking for but
those two structs have a _lot_ of fields.

>
> I suggest you write a patch yourself, it should be
> trivial and you probably learn something about
> FFmpeg internals.

Fair enough. I'll think about it.
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Re: [FFmpeg-user] Possibility to dump AVCodecContext, AVFormatContext fields for a command line?

2014-07-18 Thread Robert Krüger
On Fri, Jul 18, 2014 at 1:52 PM, Carl Eugen Hoyos  wrote:
> Robert Krüger  lesspain.de> writes:
>
>> is there any way to see the AVCodecContext, AVFormatContext fields
>> used for a given command line other than using a debugger? Maybe a
>> compile flag of some sort?
>
> It is currently not easily possible but could you explain your
> usecase?
> This would make a patch more likely imo.
>

The use case is simply when working with the api doing encoding/muxing
and running into a problem in many cases the key is some field value
in one of those contexts not being set correctly, where correctly is
not always obvious (see the posting on libav-user yesterday where you
helped me). Simulating a similar encoding/muxing using an ffmpeg
command line is the first thing I do to see if it should generally
work. If that is the case, looking at the context fields helps a lot.
I used to fire up a gcc/gdb based tool for that but that stopped to
work on my platform now. Probably not a typical ffmpeg user use case.
If there is no way currently then I will just have to get familiar
with the standard debugging tools. I just thought I'd ask before I do
that.

Robert
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[FFmpeg-user] Possibility to dump AVCodecContext, AVFormatContext fields for a command line?

2014-07-18 Thread Robert Krüger
Hi,

is there any way to see the AVCodecContext, AVFormatContext fields
used for a given command line other than using a debugger? Maybe a
compile flag of some sort?

Thanks,

Robert
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Re: [FFmpeg-user] ffprobe duration analysis and -analyzeduration and -probesize

2014-07-15 Thread Robert Krüger
Hi,


thanks for the quick response.

On Tue, Jul 15, 2014 at 6:54 PM, Moritz Barsnick  wrote:
> Hi Robert,
>
> On Tue, Jul 15, 2014 at 17:09:32 +0200, Robert Krüger wrote:
>
>> Now the command line options -analyzeduration and
>> -probesize exist and so I naively thought, if I just specified them
>> large enough so the entire file is parsed, durations must be exact
>> but that does not seem to be the case (I compared the values to the
>> ones obtained by adding up all packet durations of a given stream and
>> those make more sense).
>>
>> Where is my mistake?
>
> I'm guessing here, as I don't know how the duration is determined for
> this specific type of container and codec. But I assume that those two
> options don't change the way it's determined, they only change two
> parameters. So if ffmpeg happens to do something like checking file
> headers or estimating from the initial frame and bit rates, this may
> not change.

If someone could confirm this, this would answer my question.

>
> If you're going to have ffmpeg analyze the whole file anyway, there are
> several other ways to do this:
> * Have ffprobe output data for each and every frame, and then count
> those frames.

I know that alternative and I have a solution doing that via API up
and running. I was just looking for a way to simply make ffmpeg (or
rather the api) do that automatically for me to avoid that but if
there is no other way, I will live with that.

>
> * Have ffmpeg operate on the complete file, and grab its results:
> $ ffmpeg -i VTS_03_1.VOB -an -sn -f null -
> [...]
> Input #0, mpeg, from 'VTS_03_1.VOB':
>   Duration: 00:00:31.55, start: 0.28, bitrate: 8357 kb/s
> [...]
> frame=  791 fps=352 q=0.0 Lsize=N/A time=00:00:31.64 bitrate=N/A
>
> (I think I have seen a solution on this list to parse the latter.)
>
> Note the final "frame=" output. I don't know whether adding "-c
> copy", which is noticably faster, always definitely produces the same
> results.

Since the output of this is not really "api", the ffprobe variant
would be safer but everyone has to make their own call regarding the
risk of having this text output change from one version to another.

Cheers,

Robert


-- 
Robert Krüger
Managing Partner
Lesspain GmbH & Co. KG

www.lesspain-software.com
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[FFmpeg-user] ffprobe duration analysis and -analyzeduration and -probesize

2014-07-15 Thread Robert Krüger
Hi,

I am trying to get ffprobe to correctly extract durations from some
MTS files. My understanding is that the transport stream format does
not provide a cheap way to obtain the duration but one has to parse
each packet from start to end to reliably get the duration. Since
ffmpeg and ffprobe only do that by default to a certain extend they
have to "guess" (no idea how that is implemented for ts atm) the
correct duration. Now the command line options -analyzeduration and
-probesize exist and so I naively thought, if I just specified them
large enough so the entire file is parsed, durations must be exact but
that does not seem to be the case (I compared the values to the ones
obtained by adding up all packet durations of a given stream and those
make more sense).

Where is my mistake?

Command line and uncut output below:

ffprobe -analyzeduration 10 -probesize 10 -i
~/samples/cameras/canon/hf_100/canon_hf_100_fxp_1080_25i_10s.MTS
-show_streams -select_streams v:0
ffprobe version git-2014-07-11-96a860a Copyright (c) 2007-2014 the
FFmpeg developers
  built on Jul 11 2014 16:33:17 with Apple LLVM version 5.1
(clang-503.0.40) (based on LLVM 3.4svn)
  configuration: --enable-shared --disable-static --disable-doc
--disable-avdevice
--prefix=/Users/admin/repos/de.lesspain/de.lesspain.ffmpeg/ffmpeg/build/ffmpeg
--extra-cflags=-I/Users/admin/repos/de.lesspain/de.lesspain.ffmpeg/ffmpeg/build/libmp3lame/include/
--extra-ldflags=-L/Users/admin/repos/de.lesspain/de.lesspain.ffmpeg/ffmpeg/build/libmp3lame/lib/
--extra-ldflags='-Wl,-rpath,@loader_path/../lib' --enable-libmp3lame
  libavutil  52. 92.100 / 52. 92.100
  libavcodec 55. 69.100 / 55. 69.100
  libavformat55. 45.100 / 55. 45.100
  libavfilter 4. 10.100 /  4. 10.100
  libswscale  2.  6.100 /  2.  6.100
  libswresample   0. 19.100 /  0. 19.100
Input #0, mpegts, from
'/Users/krueger/samples/cameras/canon/hf_100/canon_hf_100_fxp_1080_25i_10s.MTS':
  Duration: 00:00:10.56, start: 0.86, bitrate: 16346 kb/s
  Program 1
Stream #0:0[0x1011]: Video: h264 (High) (HDMV / 0x564D4448),
yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x1100]: Audio: ac3 (AC-3 / 0x332D4341), 48000 Hz,
stereo, fltp, 256 kb/s
[STREAM]
index=0
codec_name=h264
codec_long_name=H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10
profile=High
codec_type=video
codec_time_base=1/50
codec_tag_string=HDMV
codec_tag=0x564d4448
width=1920
height=1080
has_b_frames=1
sample_aspect_ratio=1:1
display_aspect_ratio=16:9
pix_fmt=yuv420p
level=40
color_range=N/A
color_space=unknown
timecode=N/A
id=0x1011
r_frame_rate=25/1
avg_frame_rate=25/1
time_base=1/9
start_pts=84600
start_time=0.94
duration_ts=939600
duration=10.44
bit_rate=N/A
max_bit_rate=N/A
nb_frames=N/A
nb_read_frames=N/A
nb_read_packets=N/A
DISPOSITION:default=0
DISPOSITION:dub=0
DISPOSITION:original=0
DISPOSITION:comment=0
DISPOSITION:lyrics=0
DISPOSITION:karaoke=0
DISPOSITION:forced=0
DISPOSITION:hearing_impaired=0
DISPOSITION:visual_impaired=0
DISPOSITION:clean_effects=0
DISPOSITION:attached_pic=0
[/STREAM]

Best regards,

Robert
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[FFmpeg-user] How does ffmpeg calculate bitrate?

2014-07-14 Thread Robert Krüger
Hi,

when specifying bitrate using minrate/maxrate arguments, how does
ffmpeg measure the bitrate in its rate control algorithm? Is this the
bitrate as a GOP average, i.e. sum(packet sizes of gop)/gop duration
or simply packet size/packet duration that ffmpeg attempts to keep
within the limits? The latter would seem odd for non-I-frame-only
material as an I-frame of a certain quality is typically (depending on
the motive, I know) a lot larger than the following P- or B-frames of
the same quality. If the answer is "it depends on the codec" then I
would like to know how it is for the mpeg-style codecs, e.g.
mpeg2video.

Thanks for any insights into this,

Robert
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