[FFmpeg-user] MLP encoder: encodes 24-bit wav files into 16-bit MLP

[fabrice nicol]

I'm experiencing a parallel issue with encoding, the reverse operation 
of my previous question about the MLP decoder.

ffmpeg is build from git source code. Choosing a simple 2-channel 24 
-bit 96khz wav file yields:

   `~ $  /usr/local/bin/ffmpeg -i a_2_24_96.wav -c:a mlp -strict 
experimental a_2_24_96.mlp`

ffmpeg version git-2019-07-05-7c64498 Copyright (c) 2000-2019 the FFmpeg 
developers
   built with gcc 9.1.0 (Gentoo 9.1.0-r1 p1.1)
   configuration:
   libavutil      56. 30.100 / 56. 30.100
   libavcodec     58. 53.101 / 58. 53.101
   libavformat    58. 28.101 / 58. 28.101
   libavdevice    58.  7.100 / 58.  7.100
   libavfilter     7. 56.100 /  7. 56.100
   libswscale      5.  4.101 /  5.  4.101
   libswresample   3.  4.100 /  3.  4.100
Input #0, wav, from 'a_2_24_96.wav':
   Duration: 00:14:14.05, bitrate: 4608 kb/s
     Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 96000 Hz, 
stereo, s32 (24 bit), 4608 kb/s
Stream mapping:
   Stream #0:0 -> #0:0 (pcm_s24le (native) -> mlp (native))
Press [q] to stop, [?] for help
Output #0, mlp, to 'a_2_24_96.mlp':
   Metadata:
     encoder         : Lavf58.28.101
     Stream #0:0: Audio: mlp, 96000 Hz, stereo, s16, 128 kb/s
     Metadata:
       encoder         : Lavc58.53.101 mlp
size=  138569kB time=00:14:14.05 bitrate=1329.1kbits/s speed=57.8x
video:0kB audio:138569kB subtitle:0kB other streams:0kB global 
headers:0kB muxing overhead: 0.000000%

The output is 16-bit depth not 24-bit:

Input #0, mlp, from 'a_2_24_96.mlp':
   Duration: N/A, start: 0.000000, bitrate: N/A
     Stream #0:0: Audio: mlp, 96000 Hz, stereo,s16

Is this a limitation of the currently still experimental encoder or am I 
doing something wrong?

Fabrice

Le 03/07/2019 à 11:17, Moritz Barsnick a écrit :
> On Wed, Jul 03, 2019 at 10:17:57 +0200, fabrice nicol wrote:
>> The MLP decoder does not seem to work for 24-bit MLP. It converts such
>> files into 16-bit WAV, whatever the sample rate.
> Actually, you could think so. But the decoder is fine. On the other
> hand, the wav output format defaults to 16 bits.
>
>>     `ffmpeg -i /mnt/cdrom/AUDIO_TS/ATS_01_3.AOB ~/a.wav`
> [...]
>> Stream mapping:
>>     Stream #0:0 -> #0:0 (mlp (native) -> pcm_s16le (native))
> [...]
>>       Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 96000 Hz, 5.1, s16, 9216 kb/s
> Just add "-c:a pcm_s24le" as an output option. Then you get the full
> sample depth.
>
>> sox --i  a.wav
> Tested and confirmed with sox here as well. ;-)
>
> Hope this helps,
> Moritz
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