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Corona Radiata
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Lights in the Sky
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Head Down
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Echoplex
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Discipline
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Letting You
26 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
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Keeping control of your media in your hands
13 avril 2011, parThe vocabulary used on this site and around MediaSPIP in general, aims to avoid reference to Web 2.0 and the companies that profit from media-sharing.
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Sur d’autres sites (12775)
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how to automatically oversample and downsample when using filters ?
4 juin 2021, par Yue WangI am researching avfilters. Say I have an audio with sample rate
s
, and bit depthd
,

What I want to do is to write a graph that


- 

- upsample
s
by 4x, and set precision to 64bit float - apply some biquad filters with 64bit precision
- downsample by 4x back to
s
, and set bit depth back tod
.








The reason to oversample is to get better filtering result by antialiasing, and the reason to downsample is to stream using the original source format.


I don't know if there's way that I can do it automatically in the graph.


ashowinfo
could print out the sample rate, but seems there's no way to use the value later in the pipeline.

asoftclip
has aoversample
factor. but it's not available in other filters.

- upsample
-
bad audio mic recording quality with ffmpeg compared to sox
1er juillet 2021, par user2355330I am contacting you as after 3 days of searching I am stuck on a really simple point.


I want to record the sound of my mic on MacOS using ffmpeg.


I managed to do it using the following command :


ffmpeg -f avfoundation -audio_device_index 2 -i "none:-" -c:a pcm_s32l alexspeaking.wav -y -loglevel debug



The issue is that each time I am speaking, there are cracks and pop in the sound...


I tried to use sox and it gave me a perfect and crystal clear sound and I have no idea why... Below is the output of the sox command :


sox -t coreaudio "G935 Gaming Headset" toto.wav -V6
sox: SoX v
time: Nov 15 2020 01:06:02
uname: Darwin MacBook-Pro.local 20.5.0 Darwin Kernel Version 20.5.0: Sat May 8 05:10:33 PDT 2021; root:xnu-7195.121.3~9/RELEASE_X86_64 x86_64
compiler: gcc Apple LLVM 12.0.0 (clang-1200.0.32.27)
arch: 1288 48 88 L
sox INFO coreaudio: Found Audio Device "DELL U2721DE"
sox INFO coreaudio: Found Audio Device "G935 Gaming "
sox DBUG coreaudio: audio device did not accept 2 channels. Use 1 channels instead.
sox DBUG coreaudio: audio device did not accept 44100 sample rate. Use 48000 instead.
Input File : 'G935 Gaming Headset' (coreaudio)
Channels : 1
Sample Rate : 48000
Precision : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type : little
Reverse Nibbles: no
Reverse Bits : no
sox INFO sox: Overwriting `toto.wav'
sox DBUG wav: Writing Wave file: Microsoft PCM format, 1 channel, 48000 samp/sec
sox DBUG wav: 192000 byte/sec, 4 block align, 32 bits/samp
Output File : 'toto.wav'
Channels : 1
Sample Rate : 48000
Precision : 32-bit
Sample Encoding: 32-bit Signed Integer PCM
Endian Type : little
Reverse Nibbles: no
Reverse Bits : no
Comment : 'Processed by SoX'
sox DBUG effects: sox_add_effect: extending effects table, new size = 8
sox INFO sox: effects chain: input 48000Hz 1 channels (multi) 32 bits unknown length
sox INFO sox: effects chain: output 48000Hz 1 channels (multi) 32 bits unknown length
sox DBUG sox: start-up time = 0.051332
In:0.00% 00:00:07.13 [00:00:00.00] Out:340k [ | ] Clip:0 ^C
sox DBUG input: output buffer still held 2048 samples; dropped.
Aborted.
sox DBUG wav: Finished writing Wave file, 1359872 data bytes 339968 samples



I am pretty sure the issue is linked to the way the encoding is done and the params I used with ffmpeg but I don't seem to be able to grasp which one I must use.


Any ideas if there are ffmpeg experts here ?


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libavcodec/libx265 : add user data unregistered SEI encoding
12 juillet 2021, par Brad Hardslibavcodec/libx265 : add user data unregistered SEI encoding
MISB ST 0604 and ST 2101 require user data unregistered SEI messages
(precision timestamps and sensor identifiers) to be included. That
currently isn't supported for libx265. This patch adds support
for user data unregistered SEI messages in accordance with
ISO/IEC 23008-2:2020 Section D.2.7The design is based on nvenc, with support finished up at
57de80673cbSigned-off-by : Derek Buitenhuis <derek.buitenhuis@gmail.com>