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
_______________________________________________
ffmpeg-user mailing list
[email protected]
https://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
[email protected] with subject "unsubscribe".
_______________________________________________
ffmpeg-user mailing list
[email protected]
https://ffmpeg.org/mailman/listinfo/ffmpeg-user
To unsubscribe, visit link above, or email
[email protected] with subject "unsubscribe".
