ined.
Signed-off-by: Marcus B Spencer
---
libavcodec/bsf/noise.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavcodec/bsf/noise.c b/libavcodec/bsf/noise.c
index a622855717..d36be5fab4 100644
--- a/libavcodec/bsf/noise.c
+++ b/libavcodec/bsf/noise.c
@@ -173,7 +173,7
This patch does not meet the guidelines of commit messages.
On Thursday, April 25th, 2024 at 6:02 PM, Marcus B Spencer
wrote:
>
>
> Signed-off-by: Marcus B Spencer mar...@marcusspencer.xyz
>
> ---
> libavcodec/bsf/noise.c | 2 +-
> 1 file changed, 1 insertion(+), 1
Signed-off-by: Marcus B Spencer
---
libavcodec/bsf/noise.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavcodec/bsf/noise.c b/libavcodec/bsf/noise.c
index a622855717..d36be5fab4 100644
--- a/libavcodec/bsf/noise.c
+++ b/libavcodec/bsf/noise.c
@@ -173,7 +173,7 @@ static
use cases.
>
> On 1/16/20, Dylan Marcus wrote:
>> ---
>> Changelog | 1 +
>> libavcodec/aacenc.h| 16
>> libavutil/channel_layout.c | 8 ++
>> libavutil/channel_layout.h | 64
>> ++--
---
Changelog | 1 +
libavcodec/aacenc.h| 16
libavutil/channel_layout.c | 8 ++
libavutil/channel_layout.h | 64 ++
4 files changed, 61 insertions(+), 28 deletions(-)
diff --git a/Changelog b/Changelog
index
Hello,
I am interested in hiring someone to expand FFmpeg with a flag that changes the
behavior of channel ordering for as many audio codecs as possible. When this
flag is seen in the command line it has FFmpeg use slightly modified channel
order and channel assignment (TYPESCE instead of TYPE
I'm not sure where to post this message, but this list seemed the most
appropriate.
It seems like a better way to go about removing non-digital silence would be to
convert each sample to float or double, then compare that to the max dB
considered silence, then store the starting offset and endi
OOPS! forgot to include the command line output
THDFail.log
Description: Binary data
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The sample is exactly 1 minute long, 48000hz sample rate, 24 bits per
sample, 8 channels. and all samples are set to 0xFF
here's the thd sample, and here's one of the 8 mono wavs (they're all
literally the same, I manually created the file in a hex editor and copied
it for all the others.)
th
I like the John Nash idea.
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No, I'm writing my own codec from the ground up; not an implementation of
one that already exists based on a standardized codec like jpeg, but my own
from scratch.
I noticed that printf and scanf support format specifiers, so I was
wondering if there was a function like getopt, but specifically fo
I'm writing my own Video codec, and I'm trying to take multiple frames as
the input with a specifier like %03d, is there a stdlib function for this,
or will I have to write it myself?
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That's even worse, is there any way we can fix it? how much effort would it
require?
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I see, I thought it counted frames and not just multiplied the bitrate by
the number of frames.
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This reminds me of another bug with DTS files, it estimates the file
duration by counting each frame I assume, including the HD ones resulting
in it being massively incorrect for example here's the ffmpeg output of a
DTS-HD MA file that's actually 98 minutes long
Log:
ffmpeg -i /Us
I thought the patch on LibAV was completely removed? it was purged from the
codebase like 9 months ago or something, I stumbled on that while trying to
fix some of the issues with the white paper I was having.
I haven't bothered with the Core decoder, but everything I've extracted so
far is fixed
I've been working on adding XLL for the last couple months, it's still not
quite complete, basically I have to combine the Core and XLL samples before
it's output, and I also have to finish the latter stages of decoding the
XLL like channel decorreclation, and post processing.
There are a few issu
I just tried using skip_bits with a negative number and it seems to have
worked fine, thanks guys.
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Can this be done at all?
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What functions are used to estimate the duration and bit rate of a file?
I'd like to add support for these features for the DTS format, and would
rather not dig around the source looking for it.
thanks in advance.
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it's not "published" anywhere, I just used the github app and clicked the
new branch button, it's only on my computer right now
On Sat, Dec 20, 2014 at 5:33 AM, Carl Eugen Hoyos wrote:
> Marcus Johnson gmail.com> writes:
>
> > I'm working on the DTS c
I'm working on the DTS codec, and my code isn't complete yet, but a lot of
it is done, I've committed a few patches to my own branch, and I need to
update ffmpeg (there's a blocking change in the main tree so I can't
currently) I was wondering if I could push the uncomplete code in it's own
branch,
The SSR (Scalable Sample Rate) feature is not implemented, either add a
patch to add that feature, or decode it with something else.
On Tue, Nov 25, 2014 at 6:58 AM, Arpan Nag wrote:
> Hello,
>
> During the conversion process we are seeing this error in an output log
> file
>
> -
As of Windows 10, WinRT is deprecated, so it's kind of a waste of time dude
:/
On Mon, Nov 17, 2014 at 5:02 AM, Jesse Jiang
wrote:
> Hi All,
> I want to move ffmpeg to WinRT platform, like Windows Store and Windows
> Phone. As the GCC cannot compiler to ARM-COFF, so I convert the GNU-style
> ass
On Fri, Oct 31, 2014 at 10:09 AM, greeshma
wrote:
>
> https://docs.google.com/document/d/1oBy9AoGzJHR4UcvuogYa8sfFGUK96lcgF3qRF4HrIa4/edit?usp=sharing
That's not going to work either, you need to use git to create a patch
file, upload that file to dropbox or whatever, and paste a link to it, O
I don't have access to my command line atm, but i was using the channel map
feature to split a 24 bit 192khz 5.1 flac file into 6 mono wave files, and
ffmpeg would downcovert it to 16 bit unless i added -acodec pcm_s24le after
each -map flag, shouldn't ffmpeg read the bit depth of the input file
I'm building a custom executable that only supports lossless audio
demuxing/decoding, flac and wave work fine, but Apple Lossless (in the
itunes generated mpeg4 container) won't work.
my configure line is:
./configure --enable-lto --enable-shared --disable-static --disable-yasm
--disable-asm --di
No, it won't work because the libraries are for different purposes.
as per your example, 4xm in the libavcodec directory is for decoding the
file, it implements the actual algorithm. 4xm in the libavformat directory
is for demuxing the codec from the 4xm container, you need both for it to
work.
O
None of the other message groups will respond so I'm posting here.
I've included /usr/local/include/libavformat in my header search path, and
xcode finds avformat.h fine, but it can't find avcodec.h
also, your time.h header in libavutil is taking precedence over the system
time.h header, and it's
Hmm it worked for me, but I'll check it out.
On Tue, Sep 2, 2014 at 12:22 PM, Michael Niedermayer
wrote:
> On Tue, Sep 02, 2014 at 08:05:56AM -0400, Marcus Johnson wrote:
> > Alright, probably a bad place to post this, but I made a few cosmetic
> > changes to the dt
Alright, probably a bad place to post this, but I made a few cosmetic
changes to the dts decoder, the first replaces the request_channels calls
with the request_channel_layout function, and the second removes the code
that disabled deprecation warnings around those calls.
On Tue, Sep 2, 2014 at
If you don't mind me asking, what is DTS? because when I hear it I think of
the DTS audio codec, but I know it's obviously not that, and googling DTS
in the codec context won't get me anywhere.
On Sat, Aug 30, 2014 at 6:33 PM, wm4 wrote:
> On Sat, 30 Aug 2014 18:16:27 -040
Here's a PGS subtitle sample from Eac3to:
https://dl.dropboxusercontent.com/u/52358991/English.zip
On Sat, Aug 30, 2014 at 9:38 AM, wm4 wrote:
> ---
> Reverse engineered by looking at Petri Hintukainen's muxer patch, and
> also the mkvmerge sources. Tested with files extracted by mkvextract
> o
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