On Tue, Mar 31, 2015 at 04:24:07PM +0200, Clément Champetier wrote:
> Hello,
>
> I work on an opensource project, AvTranscoder (
> https://github.com/mikrosimage/avTranscoder), which is basically a project
> to give a high level API of ffmpeg in C++, Java and python.
>
> To easily launch some enc
Hi all
On Fri, Aug 01, 2014 at 03:16:31AM +0200, Michael Niedermayer wrote:
> Hi all
>
> OPW (Outreach Program for Women) is twice per year (compared to google
> summer of code which is just once a year)
>
> FFmpeg can participate in the next round but we need to fund a
Hi all
OPW (Outreach Program for Women) is twice per year (compared to google
summer of code which is just once a year)
FFmpeg can participate in the next round but we need to fund at least
1 intern/student (6250 USD) for that. OPW is not run by a large
corporation with deep pockets.
Thus my mai
On Fri, May 23, 2014 at 11:23:17PM +0200, Benjamin Drung wrote:
> Hi Michael,
>
> Am Mittwoch, den 21.05.2014, 02:16 +0200 schrieb Michael Niedermayer:
> > > I like to commit your
> > > work to the Audacity SVN so that we have both - support for FFmpeg and
> >
Hi
On Thu, May 22, 2014 at 04:57:43AM +0100, Gale Andrews wrote:
>
> | From Michael Niedermayer
> | Wed, 21 May 2014 15:26:21 +0200
> | Subject: [Audacity-devel] [Libav-user] Requesting help to port Audacity to
> recent FFmpeg
> > On Wed, May 21, 2014 at 01:39:46AM +01
t can be simplified in it, and the
amount of "api-hassle" there would be in the future should be alot
less when the interface is using only what it actually needs to
>
> TTFN
> Martyn
>
> On 20/05/2014 09:14, Richard Ash wrote:
> > On Wed, 14 May 2014 19:27:38 +
Hi
On Tue, May 20, 2014 at 11:41:53PM +0200, Benjamin Drung wrote:
[...]
> > > As a result of this problem, one of the Audacity contributors, Leyland
> > > Lucius, has been perusing the use of Gstreamer as an abstraction layer
> > > for ffmpeg. This work has recently arrived in Audacity SVN, so yo
Hi
On Tue, May 20, 2014 at 09:14:33AM +0100, Richard Ash wrote:
> On Wed, 14 May 2014 19:27:38 +0200
> Michael Niedermayer wrote:
> > On Sun, May 11, 2014 at 09:16:29PM +0200, Benjamin Drung wrote:
> > > That's why I send this mail to this mailing list to request help.
On Wed, May 14, 2014 at 04:41:54PM -0300, Claudio Freire wrote:
> On Wed, May 14, 2014 at 2:27 PM, Michael Niedermayer wrote:
> > diff --git a/libavformat/libavformat.v b/libavformat/libavformat.v
> > index 0b47668..0d5d3b0 100644
> > --- a/libavformat/libavformat.v
Hi
On Sun, May 11, 2014 at 09:16:29PM +0200, Benjamin Drung wrote:
> Hi,
>
> Audacity, the digital audio editor used by millions, has import/export
> support for a wide range of audio formats using the FFmpeg libraries.
> Audacity compiles only against FFmpeg < 1.0 [1]. The FFmpeg libraries
> hav
On Tue, Apr 02, 2013 at 08:38:18PM +0200, Lars Hammarstrand wrote:
> 2013/4/2 Carl Eugen Hoyos
>
> > Lars Hammarstrand writes:
> >
> > > (gdb) disass $pc-32,$pc+32
> >
> > Please add the register dump, some developers can
> > see the problem if it is present.
> >
> >
>
> Complete trace:
> $ *un
On Wed, Feb 13, 2013 at 08:16:14PM +0100, Lars Hammarstrand wrote:
> Hello Michael - great news, thank you very much !! Which branch and repo
> is this patch applied to? /Thanks in advance, Lars.
patch is based on master, its not applied yet
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF
Hi
On Wed, Feb 13, 2013 at 04:36:38PM +0100, Lars Hammarstrand wrote:
> Hello, can someone please explain the reason (and how to cope with) that *
> ff_log2_tab* is defined multiple times with *#include "libavutil/log2_tab.c*"
> in the ffmpeg v1.1 libraries like libavcodec 54.91.102 and companions
On Mon, Sep 03, 2012 at 01:11:58PM +0200, Moises Ferrer Serra wrote:
> Hello,
>
> I have a working implementation of clip playback with ffmpeg, but I
> am finding some problems with looping. The documentation states that
> avcodec_flush_buffers should be called when seeking, but I was
> wondering
On Sun, Jul 22, 2012 at 02:01:02PM +0100, David Rodrigues wrote:
> Hi everyone.
>
> I've written some code to change the video container from MOV to MPEG-TS
> using ffmpeg API successfully. I'm using physical files.
>
> Now i have to change my input from file to custom IO. I changed the code fo
On Sun, Jul 08, 2012 at 01:04:17PM +0200, Robert Krüger wrote:
> Hi,
>
> I'm trying to do a colorspace conversion using libswscale. The
> documentation for sws_setColorspaceDetails doesn't have all params
> documented:
added docs for the rest, but note converting between different types
of YUV ma
On Tue, Jun 26, 2012 at 03:38:47PM -0600, Michael Bradshaw wrote:
> On Tue, Jun 26, 2012 at 2:16 PM, Michael Bradshaw
> wrote:
> > Hi,
> >
> > I'm encoding an AMR-NB stream, and I see that the libopen-core-amrnb
> > encoder has the CODEC_CAP_DELAY capability set. If I call
> > avcodec_encode_audio
On Tue, Jun 12, 2012 at 11:44:03AM -0600, Michael Bradshaw wrote:
> avformat_find_stream_info() attempts to find the file's and streams'
> durations, but it will guess them based on the bitrate if it can't
> determine the actual duration. This guess can be useful, but I need to
> be able to know if
On Mon, Jan 09, 2012 at 04:02:30PM -0300, Gonzalo Garramuno wrote:
> I have been trying to compile ffmpeg for windows either as a cross compile
> from linux.
> Under the cross compile all compiles fine, but the resulting dlls crash on
> some movies and ffmpeg reports that the compiler did not align
On Thu, Nov 17, 2011 at 08:48:15PM +, Andrea3000 wrote:
> I use latest git version of FFmpeg API as parser in my
> media player app (not as decoder).
> It works great with every format/codec/muxer/ecc..
>
> The only annoying thing is that with around 50%-60% of
> MPEG-TS file that I've teste
From: Alexander Strasser
We happily announce that FFmpeg will be represented at `Chemnitzer Linux-Tage'
in Chemnitz, Germany. The event will take place on 17th and 18th of March.
More information can be found here:
http://chemnitzer.linux-tage.de/2012/info/index?cookielang=en
We hereby
On Wed, Nov 02, 2011 at 05:48:16PM +0700, Anton Adamansky wrote:
> Hello!
>
> I'm converting sample rate for raw pcm buffer:
>
> input:2ch, in_srate: 22050 sample: 16bits
> output: 2ch, out_srate: 44100 sample: 16bits
>
> So I'm using swr_convert() function:
>
> int out_samples =
>
On Wed, Oct 12, 2011 at 08:05:44AM +0200, Marlon Reid wrote:
> Hi,
>
> I have a small application that takes a raw pcm stream, encodes it to
> mp3 and streams it to something like VLC. I am however havinging issues
> with encoding my raw pcm stream. The problem is that my outbuf from
> avcodec_enc
On Mon, Oct 17, 2011 at 07:12:54PM +0300, Andrey Utkin wrote:
> Hi.
> I do encoding h264 video in my app that uses libavcodec (git almost
> HEAD (f22bc68 Oct, 7), x264 git almost HEAD too(8a62835 Sep, 14)). I
> set to output AVCodecContext options bit_rate, rc_buffer_size,
> rc_max_rate, rc_min_rat
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