Hello!
I'm looking to merge various audio and video sources into one. The sources
need to start at different time offsets. My guess is to use avfilter lib.
Would that do the trick? Is that the right direction?
If avfilter lib is the correct direction I would greatly appreciate if
someone could
Dear All,
I'm using libavcodec in a project that I'm working to decode a received
video stream. One of the requirements of this project is the ability to
perform fast switching to a different stream. The new stream may be encoded
at a different bitrate for example or a completely different video.
L'octidi 28 prairial, an CCXXI, Gonzalo Garramuño a écrit :
I am on windows using zeranoe builds. I am using swresample. I have
the following:
5.1(side), channels 6, format fltp, sample rate 48000
converting to:
stereo, channels 2, format s16, sample rate 48000
The result is that I hear
On Thu, Jun 13, 2013 at 5:52 PM, Carl Eugen Hoyos ceho...@ag.or.at wrote:
Dragos Iordache dragoshiordache@... writes:
On Thu, Jun 13, 2013 at 5:36 PM, Carl Eugen Hoyos wrote:
Dragos Iordache dragoshiordache at ... writes:
The stream is not showing and the recorded video does not
Dragos Iordache dragoshiordache@... writes:
If my output is a file (test.h264) then everything
works ok.
Does that mean if you feed test.h264 into ffmpeg to
produce the rtmp stream that you need (is that correct?),
it works fine, but if you call your application that
uses the libav*
Hi,
I am using ffmpeg version git-2013-06-14-7fff3df and when I use valgrind on a
program using ffmpeg, it displays the following messages :
==19153== 288 bytes in 1 blocks are possibly lost in loss record 112 of
223==19153==at 0x4C29DB4: calloc (in
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.
--
Andrey Utkin
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On 17/06/13 04:47, Nicolas George wrote:
Can you reproduce the problem using the ffmpeg command-line tools? I
just tried and the conversion works as expected.
How do I force ffmpeg to use s16 output?
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On 17/06/13 04:47, Nicolas George wrote:
Can you reproduce the problem using the ffmpeg command-line tools? I just
tried and the conversion works as expected.
I find ffmpeg does not rely on swresample for the conversion (at least
there's no swr_init in the code).
ffplay does use swresample,
On Mon, Jun 17, 2013 at 4:28 PM, Gonzalo Garramuno ggarr...@gmail.com wrote:
On 17/06/13 04:47, Nicolas George wrote:
Can you reproduce the problem using the ffmpeg command-line tools? I just
tried and the conversion works as expected.
I find ffmpeg does not rely on swresample for the
Hello everyone,
I 'm having a strange behaviour which is When I try to stream from an rtsp
server on UDP transport with ffplay (the latest one 1.2.1), there are only
a few packet drops, and it works well.
But when I try to do same test with iPhone 5 (same ffmpeg configuration),
many packets
El 17/06/2013 11:28 a.m., Gonzalo Garramuno escribió:
On 17/06/13 04:47, Nicolas George wrote:
Can you reproduce the problem using the ffmpeg command-line tools? I
just
tried and the conversion works as expected.
I found out the problem. It was a bad jack for headphones. Drove me nuts.
I want to decode an mp3-File with the ffmpeg library.
#include math.h
#include libavutil/opt.h
#include libavcodec/avcodec.h
#include libavutil/channel_layout.h
#include libavutil/common.h
#include libavutil/imgutils.h
#include libavutil/mathematics.h
#include
2013/6/17 Daniel Glaß danielgla...@aol.de:
original sound, but i sounds like the smurfs and its distorted. Have anyone
here an idea?
Does sounding like smurf means something like childish voice with
going faster than original? Then it may be played back at sample rate
higher than original
Does sounding like smurf means something like childish voice with
going faster than original? Then it may be played back at sample rate
higher than original 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
On 6/17/2013 9:38 AM, Gonzalo Garramuño wrote:
El 17/06/2013 11:28 a.m., Gonzalo Garramuno escribió:
On 17/06/13 04:47, Nicolas George wrote:
Can you reproduce the problem using the ffmpeg command-line tools? I
just
tried and the conversion works as expected.
I found out the problem. It was
Guys,
I have set:
decoderContext-err_recognition = AV_EF_EXPLODE;
but the call to avcodec_decoder_video2 never returns an error (though I can
clearly see artifacts on my screen). I'm feeding fully encoded frames using
my rtp library and this happens after packet loss. I have other ways of
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