> […]
>
> With qmin4 there are still tiny blocks (looks better than qmin=1), and I
> now from other transcoders that the frames were these tiny blocks happen
> CAN BE better encoded so that no blocks are visible for the whole GOP.
> My questions are:
> 1.) Why do I get a better quality with
>> Hi there,
>>
>> I sometimes need to adjust true-peak of audio files, only for small
>> corrections (from -0.7 to -1.0 for example), and leave everything else
>> unchanged. As there's no dedicated true-peak limiter in ffmpeg audio filters
>> (if I'm not wrong), I tried with loudnorm filter,
>> Hi there,
>>
>> I sometimes need to adjust true-peak of audio files, only for small
>> corrections (from -0.7 to -1.0 for example), and leave everything else
>> unchanged. As there's no dedicated true-peak limiter in ffmpeg audio filters
>> (if I'm not wrong), I tried with loudnorm filter,
Hi there,
I sometimes need to adjust true-peak of audio files, only for small corrections
(from -0.7 to -1.0 for example), and leave everything else unchanged.
As there's no dedicated true-peak limiter in ffmpeg audio filters (if I'm not
wrong), I tried with loudnorm filter, but even when
>>> Hello,
>>> I'm very new to ffmpeg. I work at a production company and I need to make
>>> Broadcast ProResHQ files with 5.1 audio plus Stereo on Ch.7&8.
>>>
>>> I haven't been able to figure out the command line inputs to get the
>>> correct audio configuration. I'm attaching a screen grab of
> Hey everyone,
>
> I've found that my video files captured with FFmpeg don't play nice with
> other programs. Many video players display
> incorrect duration, others allow no scrubbing, and some won't even open the
> files at all. But perhaps the most annoying
> incompatibility is with Adobe
>> Hi all,
>>> Strange thing, more than 8 channels give silent output:
>>>
>>> ffmpeg -i 9Ch_orMore.WAV -filter_complex "[0:0][0:0]
>>> amix=inputs=2,pan=mono|c0=c0+c1" -ac 1 mono.wav
>>>
>>> gives on errors whatsoever, but output is silent.
>>>
>>> Is this a bug or user error?
>>
>> User
> The needed options for this test are:
> $ ./configure && make ffmpeg
Hi again,
Just had a look at it, I think that I could as well add this line after line
1165 :
case 2160: f1 = 42; f2 = 0; break; // progressive
Right ?
Fred
___
> The needed options for this test are:
> $ ./configure && make ffmpeg
Yes, but I sometimes need some additional features and codecs. ;-)
Best regards,
Fred
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
>>> Sorry, I am not sure I understand correctly:
>>> If you change the value "0" for Video Line Map in a 3840x2160 file
>>> created with FFmpeg to "42", the file works with Avid Media Composer?
>>
>> As far as I know, yes.
>
> How do you know?
Technical advise of the TV Network that handled the
>> Thanks for support again !
>> As I'm not the end user, it's difficult to tell you exactly, but the problem
>> occurs with Avid Media Composer.
>> Video Line Map should be at 42, and it's at 0.
>
> Sorry, I am not sure I understand correctly:
> If you change the value "0" for Video Line Map in
>> Most of time it's not an issue, but in a particular case, the player
>> can't read properly the file as the "Video Line Map" tag value is
>> not what expected.
>
> How do you know that this is the (one) value that stops your player
> from decoding FFmpeg's output file?
> What is the value that
>> I use ffmpeg to replace audio streams in MXF containers
>
> The tool ffmpeg does not support "replacing" streams in
> containers.
>
> If you provide the command line you tested and the complete,
> uncut console output, we may be able to understand what
> missing feature in the mxf (de-)muxer
Hi,
I use ffmpeg to replace audio streams in MXF containers, with -c:v option to
leave video stream unchanged.
I thought that all MXF metadata and tags would be unchanged, but that's not the
case and everything's is changed in the header.
Most of time it's not an issue, but in a particular
Hi Olivier,
You should have a look here :
https://trac.ffmpeg.org/wiki/FancyFilteringExamples#waveform
And also at the Multimedia Filters section (#42) of the documentation here :
https://ffmpeg.org/ffmpeg-all.html
You'll probably find a way to do what you want there.
Fred
> Hello everyone,
> [jpeg2000 @ 0x7ff4f2018000] Support for 4 components is not implemented.
> Update your FFmpeg version to the newest one from Git. If the problem still
> occurs, it means that your file has a feature which has not been
> implemented.
Can't try at the moment, but your version is very outdated, 4
>> Looks like you need to resample audio to sample rate ac3 supports.
>
> I've just tried to convert two different Dolby E source from wav to u8, then
> ffprobe recognize them as a Dolby E stream, but it says that they're 44800Hz,
> I guess it should be 48000Hz. Perhaps there's a bug here ?
>
> Looks like you need to resample audio to sample rate ac3 supports.
I've just tried to convert two different Dolby E source from wav to u8, then
ffprobe recognize them as a Dolby E stream, but it says that they're 44800Hz, I
guess it should be 48000Hz. Perhaps there's a bug here ?
But I
> Afaict, SMPTE 337M is a file format.
OK, thanks. I have made many search but can't find any info about it. With what
I've found, it seems to be more like a protocol (non-PCM stream transport in an
AES-3 link) than a file format ?
> Please do not top-post here, Carl Eugen
Sorry, I don't
Hi Carl Eugen,
Thanks for this quick reply !
Not sure to have well understood: can a SMPTE 337M file be in a wav container,
or it has a specific wrap and file extension ?
Best regards,
Fred
> 2017-09-07 15:20 GMT+02:00 Frédéric Busnel-Joncour <fred...@free.fr>:
>
>> I've
Hi there,
First question here !
I've seen that a Dolby E decoder has been added to the last release of
libavcodec.
I'm sometimes delivered of stereo 24bit/48Khz wav file that are Dolby E encoded
streams.
Is there a way to tell ffplay to play it as Dolby E and decode it ? By default
it plays
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