Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-07 Thread Carl Eugen Hoyos 
2018-05-06 23:04 GMT+02:00, Moritz Barsnick :

> The ffmpeg docs and source aren't absolutely clear about it: I
> assume ffmpeg's DCA encoder only support DCA Core?

Definitely.

Carl Eugen
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Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-06 Thread Moritz Barsnick 
On Sun, May 06, 2018 at 00:46:43 +0200, Nomis101 ? wrote:
> Am 05.05.18 um 23:45 schrieb Lou Logan:
> > Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000
> 
> OK, thanks. I did not know, that it is possible to look up the supported
> sample rates like this. Are there any known plans to support 96000 in
> the future?

foo86 is the maintainer of ffmpeg's DCA code, but hasn't been active
for the last few months.

The ffmpeg docs and source aren't absolutely clear about it: I assume
ffmpeg's DCA encoder only support DCA Core?

It seems to me that the DCA standard says that DCA Core only supports
up to 48 kHz. This could be the issue. You need an encoder which
supports extensions. (To be clear: ffmpeg's *de*coder does support DTS
X96k, which supports up to 96 kHz.)

Cheers,
Moritz
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Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-05 Thread Nomis101  
Am 05.05.18 um 23:45 schrieb Lou Logan:
> Both, kind of. I assume the following commit from today fixed the segfault 
> you experienced:
> https://git.videolan.org/?p=ffmpeg.git;a=commit;h=c1b282dc74d801f943db282e89f7729e90897669

Yes it does. 4.0 with patch:


$ ffmpeg -i Alchemy_edit.flac -strict -2 -acodec dts -ar 96000 -f spdif
Alchemy_96edit.spdif
ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
  built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
  configuration: --pkg-config-flags=--static --enable-fontconfig
--enable-gpl --enable-libass --enable-libbluray --enable-libfreetype
--enable-libsoxr --enable-libmp3lame --enable-libtheora
--enable-libvorbis --enable-libvpx --enable-libwavpack --disable-ffplay
--enable-libx264 --enable-libx265 --enable-libxvid --enable-version3
--enable-nonfree --enable-libfdk-aac --enable-librtmp --enable-opencl
--enable-libssh --enable-openssl --enable-bsf=aac_adtstoasc
--extra-cflags='-fPIE -pie -fstack-protector-strong -march=native
-mtune=native -fvectorize -ffast-math -fslp-vectorize
-D_FORTIFY_SOURCE=2 -mfpmath=sse'
--extra-cxxflags='-fstack-protector-strong -march=native -mtune=native
-fvectorize -ffast-math -fslp-vectorize -D_FORTIFY_SOURCE=2
-mfpmath=sse' --extra-ldflags=-L/opt/local/lib --extra-ldexeflags='-fPIE
-pie -fstack-protector-strong -D_FORTIFY_SOURCE=2'
  libavutil  56. 14.100 / 56. 14.100
  libavcodec 58. 18.100 / 58. 18.100
  libavformat    58. 12.100 / 58. 12.100
  libavdevice    58.  3.100 / 58.  3.100
  libavfilter 7. 16.100 /  7. 16.100
  libswscale  5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc    55.  1.100 / 55.  1.100
Input #0, flac, from 'Alchemy_edit.flac':
  Duration: 01:30:07.91, start: 0.00, bitrate: 8246 kb/s
    Stream #0:0: Audio: flac, 96000 Hz, 5.1(side), s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> dts (dca))
Press [q] to stop, [?] for help
[dca @ 0x7fb6d380ea00] Specified sample rate 96000 is not supported
Error initializing output stream 0:0 -- Error while opening encoder for
output stream #0:0 - maybe incorrect parameters such as bit_rate, rate,
width or height
Conversion failed!

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Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-05 Thread Nomis101  
Am 05.05.18 um 23:45 schrieb Lou Logan:
> Both, kind of. I assume the following commit from today fixed the segfault 
> you experienced:
> https://git.videolan.org/?p=ffmpeg.git;a=commit;h=c1b282dc74d801f943db282e89f7729e90897669
>
> But the encoder does not support 96000:
> $ ffmpeg -h encoder=dts
> ...
> Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000

OK, thanks. I did not know, that it is possible to look up the supported
sample rates like this. Are there any known plans to support 96000 in
the future?

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Re: [FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-05 Thread Lou Logan 
On Sat, May 5, 2018, at 7:57 AM, Nomis101  wrote:
> I do have a 96000 Hz FLAC file which I would like to convert to DTS and
> preserve the sample rate. But it seems only 41000 and 48000 is supported.
[...]
> Is this a bug or is this really not supported? 41000 and 48000 workes fine.

Both, kind of. I assume the following commit from today fixed the segfault you 
experienced:
https://git.videolan.org/?p=ffmpeg.git;a=commit;h=c1b282dc74d801f943db282e89f7729e90897669

But the encoder does not support 96000:
$ ffmpeg -h encoder=dts
...
Supported sample rates: 8000 16000 32000 11025 22050 44100 12000 24000 48000
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[FFmpeg-user] Convert to 96000 Hz DCA spdif not supported?

2018-05-05 Thread Nomis101  
I do have a 96000 Hz FLAC file which I would like to convert to DTS and
preserve the sample rate. But it seems only 41000 and 48000 is supported.


$ ffmpeg -i Alchemy_edit.flac -strict -2 -acodec dts -ar 96000 -f spdif
Alchemy_96edit.spdif
ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
  built with Apple LLVM version 9.1.0 (clang-902.0.39.1)
  configuration: --pkg-config-flags=--static --enable-fontconfig
--enable-gpl --enable-libass --enable-libbluray --enable-libfreetype
--enable-libsoxr --enable-libmp3lame --enable-libtheora
--enable-libvorbis --enable-libvpx --enable-libwavpack --disable-ffplay
--enable-libx264 --enable-libx265 --enable-libxvid --enable-version3
--enable-nonfree --enable-libfdk-aac --enable-librtmp --enable-opencl
--enable-libssh --enable-openssl --enable-bsf=aac_adtstoasc
--extra-cflags='-fPIE -pie -fstack-protector-strong -march=native
-mtune=native -fvectorize -fslp-vectorize -D_FORTIFY_SOURCE=2
-mfpmath=sse' --extra-cxxflags='-fstack-protector-strong -march=native
-mtune=native -fvectorize -fslp-vectorize -D_FORTIFY_SOURCE=2
-mfpmath=sse' --extra-ldflags=-L/opt/local/lib --extra-ldexeflags='-fPIE
-pie -fstack-protector-strong -D_FORTIFY_SOURCE=2'
  libavutil  56. 14.100 / 56. 14.100
  libavcodec 58. 18.100 / 58. 18.100
  libavformat    58. 12.100 / 58. 12.100
  libavdevice    58.  3.100 / 58.  3.100
  libavfilter 7. 16.100 /  7. 16.100
  libswscale  5.  1.100 /  5.  1.100
  libswresample   3.  1.100 /  3.  1.100
  libpostproc    55.  1.100 / 55.  1.100
Input #0, flac, from 'Alchemy_edit.flac':
  Duration: 01:30:07.91, start: 0.00, bitrate: 8246 kb/s
    Stream #0:0: Audio: flac, 96000 Hz, 5.1(side), s32 (24 bit)
Stream mapping:
  Stream #0:0 -> #0:0 (flac (native) -> dts (dca))
Press [q] to stop, [?] for help
[dca @ 0x7fcb5a80cc00] Specified sample rate 96000 is not supported
ffmpeg(3319,0x7fffa40ce380) malloc: *** error for object
0xfff0: pointer being freed was not allocated
*** set a breakpoint in malloc_error_break to debug
Abort trap: 6


Is this a bug or is this really not supported? 41000 and 48000 workes fine.

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