On Tue, Nov 29, 2016 at 09:35:35PM -0800, Len Day wrote:
> There's a call to avformat_write_header later which seg faults. BTW, I
> think based on what I see with wireshark it is not actually missing the
> packets it reports as missing.
>
> I'm using a build from source of ffmpeg-3.1.1.tar.bz2
Bluecherry LLC (USA) is looking for hiring software developer,
especially the one familiar with FFmpeg libraries, for part-time job.
For your curiosity, company sites are:
http://www.bluecherrydvr.com/
http://bluecherry.net/
You can check what the system is like by installing it following this
On Wed, May 04, 2016 at 10:39:58AM +, JULIAN GARDNER wrote:
> I have tried to use the scaler to take a 8bpp CLUT image and resize this but
> i get an error saying its not supported.
> How do you think i should go about resizing a CLUT based image?
> Will the libraries allow
> 8bit CLUT - >
On Thu, Apr 28, 2016 at 02:09:21PM +, JULIAN GARDNER wrote:
> Ok added the code from filtering_audio.c to create a filterchain, changed my
> code to take the decoded frames and push into the filter.
> filt_frame is returned and this is pushed into the avcodec_encode_audio2 and
> guess what
On Thu, Apr 28, 2016 at 01:00:37PM +, JULIAN GARDNER wrote:
> So my question is do I have to split the input to be the correct size or am i
> missing something in the audio encoding setup to allow it to do the buffering
> correctly?
Apparently yes.
I believe libavfilter API can do both
On Thu, Apr 28, 2016 at 04:54:27PM +0800, qw wrote:
> Hi,
>
> I define the following filter graph:
>
> setpts=PTS-STARTPTS, fps=fps=20.00, scale=500:280;
> drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:
> text='Test Text'"
> And avfilter_graph_parse_ptr(), and
On Thu, Apr 21, 2016 at 06:47:10PM +0300, Anton Yabchinskiy wrote:
> Any advice is appreciated.
When I had to handle IPTV streams which could have gaps, I used to pass
audio to asyncts filter [1] in a mode which fills gaps, and video to fps
filter [2] to make constant frame rate video from
Hi,
My project needs some handy interface to manage a lot of overlaid
objects (text, picture, etc.) which can move continuously. Generating
complex_filter specifications worked to some degree, but with the growth
of complexity (tens of objects having their motion trajectories) there's
a need of
Try linking also against libavformat and libavutil at last (maybe more
- try linking with all generated libs to be safe). Also try different
orders of -l options.
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is the kind of frame which gives such
situation (packet is marked as keyframe, but decoding doesn't give a
picture).
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: P (2)
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tool? What
ffprobe shows for output file?
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documented
and/or conventional for RTMP protocol.
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2013/12/11 Yamusani Vinay yamusanivi...@gmail.com:
Hi All,
How to get duration of song in milliseconds.Is there any method available to
get meta data of song.
AFAIR AVFormatContext.duration.
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It would be easier to check if you make up a minimal but complete
application that reproduces issue in stable way, and a sample source
file or URL.
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minimal code snippet to reproduce the problem, helping
would be easier.
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. you could do
ffmpeg (...options...) -f mpegts - | pv -L $BITRATE | socat (...options...)
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inside of libavcodec. Used nearly the only
thing - opening new AVCodecContext.
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to the server and therefore the connection times out and I
lose the stream.
How can I keep the connection alive?
Maybe check the source code of RTSP-related code and try to fix it.
Previously check that your assumption about the reason is right.
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problem, and compiles on linux (because the majority
of developers doesn't use Visual Studio).
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***.avi which consists of video and audio threads.
See avformat_new_stream(). Call it for each elementary stream you
have, on the output file context. The rest depends on whether you
transcode data or remux it (i.e. just copy AVPackets you got).
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Maybe you want to use quickstart option of mp4 muxer.
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2013/6/17 ... ... bersa...@hotmail.fr:
Are they known bugs ?
No.
How could I solve it ?
Learn ffmpeg API, look at different examples of usage, debug reduced code cases.
Good luck.
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recording. Check your original sample rate and
the one used by your player.
I'd suggest you producing a file in a media container, which holds
such metadata as sample rate, channels number etc., this avoids such
problems. You may produce formatted media files using libavformat.
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holds
such metadata as sample rate, channels number etc., this avoids such
problems. You may produce formatted media files using libavformat.
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between camera and ffmpeg, and i believe it
caused the data loss. If you have any middleware like this, check it
thoroughly.
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://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/muxing.c;h=7305cc6284a0e04aaddd3d79a0bc830b9dd53256;hb=HEAD
Also read installed header files, they explain API usage.
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2013/6/6 Alexander Wischnewski aw.wi...@googlemail.com:
As i said before, the camera itself is streaming rtsp via http tunnel but i
had no problems with this so far.
Obviously tunneling application looses stream data, and that is actual problem.
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);
av_dict_free(opts);
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Camera Man, this is HTTP, so no UDP-related problems.
Topicstarter, is there possibility of fetching the stream slower than
it is produced? If this is the case, camera may drop some content
which it cannot transmit in time.
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. Is there a way to change the http buffer size for incoming packets in
ffmpeg ?
What i don't understand is why you talk about RTSP and RTP while your
URL is HTTP?
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, you are not
required to restart everything after that.
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suggest you to re-read avcodec.h and avformat.h for
explanations on how to release context objects correclty, some places
in your code are suspicious for me.
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used right terms.
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, but
having overrun_nonfatal set to 1, you get decoding errors, but
consequence is just quality degradation, and possibly lost frames.
Correctly working application can live with it without need to restart
etc.
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people
can immediately compile it on their own and try debugging it.
As far as i know majority of ffmpeg developers uses Linux, so consider
making your minimal issue reproduction case to be working on Linux.
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developers agree with inclusion of such filters into
upstream?
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Hi Justin,
I have just tried your code. With trivial changes to make it compile
at work fine at my linux installation, it looks as posted below.
I have to say it doesn't fail for me, it runs cleanly and returns
normal retcode 0. Valgrind does not find any failures, too. So i won't
further
It would be easier to help you if you provide short but complete
compilable (i.e. with main() and so on) snippet reproducing your
problem.
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I'd also appreciate qualified answer to this question.
If there will not be found any smart ability to do so, you can go with
just recreating the whole filtergraph when settings change. I. e. you
flush remaining frames from filtering mechanism, free AVFilterGraph,
and create a new one for usage.
Please provide your minimized source code, which works the way you
describe, and relevant media file sample, also high-debug-level output
of it would be good.
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I can add to the above a couple of points.
You are not obliged to send your filter to us, it you are going to use
it just for yourself.
But if you are going to share some useful code, some development
community members may do some integration work for you, thus you get
your target achieved
.
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properties of certain
elementary stream.
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with highest resolution, or just first in the list.
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AVPacket data.
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2012/9/6 Tony Glover t.glo...@nottingham.ac.uk:
Can anybody shed some light on this or suggest an approach when the
situation occurs?
If (got_picture_ptr)
process_decoded_frame();
else
no_decoded_frame_to_process__its_ok__maybe_go_read_and_decode_next_packet();
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in http protocol driver, nor
in hls, mpegts demuxers.
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second. I don't remember
same about MP4.
You can convert timestamps from one timebase to another, using
function av_rescale_q().
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. Valgrind will show you exact places of leak.
If you cannot eliminate leaks in your application in such way, then
make up minimal app that uses ffmpeg API _the same way your app does_,
that reproduces the same valgrind output, and publish it in this
maillist.
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://127.0.0.1:1935/live/test.flv -loglevel debug
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libs API and wrire an app.
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, it works when the
characteristics of files is same. Also for remux method to work, mp4
files must be in Annex B format, it is quite fast to convert your
source files to this mp4 flavour.
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?
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`.
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/libavfilter.html#format - for pixfmt conversion
See doc/examples/filtering_video.c
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is not updated for
long time, look at git repos of libav.org, ffmpeg.org.
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the test output. Let test stream be bound to
localhost tcp socket, for example.
When you have the AVPacket, go muxing it into both outputs.
Then, read the test stream, and display the resulting picture after decode.
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on PC, and also with other players.
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2012/4/12 Gavin Kinsey gkin...@ad-holdings.co.uk:
On Wednesday 11 April 2012 16:35:24 Andrey Utkin wrote:
Hi. I can successfully build ffmpeg libs for iOS with --disable-asm,
but i'd like to enable it and compilation fails.
The ancient version of gcc shipped by Apple doesn't handle some
, don't
mind that it'll take many ffmpeg launches for now. I mean:
1. Cut off needed places of source media files to separate files.
2. Concatenate them as needed (possibly inserting silence where needed).
3. Create video (without audio) from images sequence.
4. Merge audio with video.
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in backend: Invalid operand for inline asm constraint 'i'!
make: *** [libavfilter/allfilters.o] Error 1
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post, describing the exact scenario you need, and we'll show
you ways to do that, and maybe other api usage examples closer to what
you need.
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? Is there any other
tutorial you could point me toward?
To _decode_ h264, you don't need x264 library support.
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AVIO_FLAG_READ))
s-local_port = port;
Indeed, i was wrong.
My apologies.
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];
[L1] [l2] amerge, aconvert=s16:stereo [out]
Did you try it with ffmpeg utility? Just to check that it's not ffmpeg
internal problem.
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.
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I'd try configure options used here
http://win32.libav.org/
if no luck, then contact libav.org (ffmpeg fork) guys as they maintain
these builds, to ask what environment they use.
Their recommendations should lead you to successful build of ffmpeg,
as forks are not so far diverged.
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'PackMan'.
You lack libraries' header files. Find and install packages that
contain files /usr/include/libavformat/avformat.h,
/usr/include/libavcodec/avcodec.h . I hope you got the point. I don't
know exact package names for suse and ubuntu.
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2012/3/16 Carl Eugen Hoyos ceho...@ag.or.at:
Andrey Utkin andrey.krieger.utkin@... writes:
I'd try configure options used here
http://win32.libav...
Since this packages have several severe problems, please do
not suggest them here.
I did not suggest packages itself. I advised to look
with
timestamps in encoder's AVCodecContext.time_base.
Printing pts of frames/packets you produce can help in debugging it.
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media containers support? If so,
what happens if you change container of media, produced by camera?
(Just a speculation, with hope to be useful.)
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to most of
maillist readers. Or we lack info to analyze.
there a minimal working example of decoding video frames?
http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/decoding_encoding.c;hb=HEAD#l405
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consider working with recent ffmpeg versions.
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() is returning -32 I can't find any
documentation
about what this return value means.
Errno=32 represents Broken pipe.
Most probably it can happen if you receive source video by network protocol.
You didn't tell what is the file and how you reach it from your app,
so above is just guess.
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much exactly resulting bitrate
differs from desired one.
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, and .rc_buffer_size (try also bitrate value, as trial
value). This possibly makes sense only if libx264 encoder is used.
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You didn't show how do you initialize contexts.
You probably miss avformat_find_stream_info() call, or/and somewhat
wrongly or incompletely copy file and codecs contexts settings (look
at avcodec_copy_context())
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as an example. But watch out - it is not
general-purposed, but just for exact use case.
https://github.com/krieger-od/imgs2video/blob/master/cat.c
Also there's very simple skeleton of remultiplexing:
https://gist.github.com/1950618
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contribute to upstream such script for
framework-izing.
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2012/2/28 Pavel Sokolov pa...@sokolov.me:
27.02.2012 22:11, Andrey Utkin пишет:
2012/2/27 Pavel Sokolovpa...@sokolov.me:
How I can convert this data to LPCM?
Isn't _that_ called LPCM?
I'm not an expert on media formats, so i can be wrong in some point.
So comments from other people
to, and why?
I'd use 0, if i am sure that only one instance of my app is launched.
Otherwise, i'd stay with 1 to avoid overloading CPU and host going
slowly responsive.
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2012/2/27 Pavel Sokolov pa...@sokolov.me:
How I can convert this data to LPCM?
Isn't _that_ called LPCM?
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meaning to be rude,
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receive
last data portion, use that data actually etc.
For immediate interruption (closing) of stream reading, use
avio_open2() int_cb parameter (see libavformat/avio.h).
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you, again, for advices!:)
I have no questions anymore
Then could you expose how your loop looks like?
Or even dream on which API calls do you want?
There's a WIP on non-blocking protocol functions, BTW.
grep NONBLOCK libavformat/*.h
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yes.
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:11:26) BBB: so then you have multiple pointers
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is used on server side, and is not distributed to consumers.
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. If crash takes place - give full
backtrace. For meaningful backtrace, build everything with debug
options.
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, it's impossible. For API that's re-initializing all
filtering stuff. You can do it, but it's a lot of code. You must read
your streams and push to filters manually, not by 'movie' filter.
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. I completely don't understand the question.
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]'
-s 1288x360 -vcodec rawvideo /home/fifo/mpegpipe99.yuv
This should skip first second of video from first source. Amount of
time can be adjusted (second operand in gte( ).
I can't test it, so i'm looking forward for your results.
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the decoding information lies in the frames.
Please correct me if i am not right.
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libavformat/http.c:101
For POSIX systems, use setenv(). For Windows, there're SetEnvironmentVariable().
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2012/2/10 Patrick Zielinski ptot...@live.ca:
I don't get the unit and opt_flags parameter, let's say I want to get width
from an ACCodecContext, how do I do it?
int width = codecCtx-width; // see avcodec.h
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with H.264 and
its flavours.
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AFAIK filter 'movie' is used for animated watermark.
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can check specific decoder
module used in your file, and find out in which cases got_picture is
set to 0.
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frame indeed happens regularly.
So you had mistake of perception.
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