I am unable to compile your code, but at first glance you are missing the
main loop. The one that would call writeaudiochunk.
El feb. 20, 2017 12:41 PM, "Prakash Rokade"
escribió:
Hiii Gonzalo Garramuño,
I have modified the code but there is issue that i am not
On 24/07/15 15:39, Robin Stevens wrote:
If you want to get a rough idea of the amount of caching going on,
how about this: when you first start feeding packets into the
codec, count how many you've fed in before it starts producing
output. That could be a rough indicator. You
On 16/06/15 09:06, Kevin J. Brooks wrote:
Yes I am. For more information, I am developing on Windoze 7.
avcodec_open2 can fail if you lack the library for the codec you want to
decode or encode. For example, libx264. The libraries in
ffmpeg.zeranoe.com contain all codecs, but they are GPL
You can build ffmpeg statically into your executable, but it does not
change the licence policy. So, no, you can't work around it that way.
2015-04-07 7:31 GMT-03:00 Phil Freeman skylinedri...@hotmail.com:
Hi Gonzalo, I probably should have mentioned that I did try the amended
linker
You should take a look at the fifo functions that would cache the audio for
you and you would feed the encoder the number of samples you see fit.
2015-01-21 8:31 GMT-03:00 Max Vlasov max.vla...@gmail.com:
Hi,
When sample rate conversion is needed, one can face another problem. Some
codecs
I want to know how can I retrieve an animated GIF duration (length in
frames or seconds). I tried context-duration and stream-duration and
both return AV_NOPTS_VALUE. Can somebody help?
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On 02/10/14 11:49, Gonzalo Garramuno wrote:
I did a git pull to get the latest head. Run configure:
$ make distclean
$ ./configure --samples=samples --enable-shared --enable-gpl
--enable-zlib --enable-libmp3lame --enable-libx264
Never mind. I found out I need to upgrade libx264 to latest
On 10/07/14 06:33, Psychesnet Hsieh wrote:
2. Who has the sample code or any suggestion, please help me out??
Seems you are missing calling avcodec_encode_video2() to fill the packet
with compressed data.
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On 11/06/14 18:56, Gonzalo Garramuno wrote:
I have written a viewer that allows writing h264 video with libx264.
However the audio (mp3, ac3 or aac) comes out of sync with the video.
The problem seems to be the first 20 frames or so that are not written
out to the file.
I can reproduce
I have written a viewer that allows writing h264 video with libx264.
However the audio (mp3, ac3 or aac) comes out of sync with the video.
The problem seems to be the first 20 frames or so that are not written
out to the file.
I can reproduce the problem with the muxing.c example from the
On 11/06/14 18:56, Gonzalo Garramuno wrote:
I have written a viewer that allows writing h264 video with libx264.
However the audio (mp3, ac3 or aac) comes out of sync with the video.
The problem seems to be the first 20 frames or so that are not written
out to the file.
I can reproduce
On 29/05/14 13:08, Gonzalo Garramuno wrote:
I have added muxing to my gpl viewer and found some issues.
I have now solved the issues. The problem was that c-frame_size was
being incorrectly changed for codecs that have
CODEC_CAP_VARIABLE_FRAME_SIZE not set.
I solved it by adding an audio fifo
I am trying to mux audio into a movie file using mp3 and one of the
conditions is that for that codec CODEC_CAP_VARIABLE_FRAME_SIZE is not
set. Currently I receive a frame_size of 1152 samples.
However, the audio frames I want to transcode have 1839 samples. I am
wondering whether I can use
On 28/05/14 22:36, Jason Blum wrote:
Wall of text incoming. Sorry if this is too long. Let me know and I
will use a pastebin next time. Below the dashed lines is a series of
compiling errors based on a sequence of adding and removing the ffmpeg
linking flags in various ways to create a
I have added muxing to my gpl viewer and found some issues.
With saving vcodec mpeg4, acodec libmp3lame, my video works fine and
audio is in sync.
However the audio (without any sound) extends past the video for several
hundred frames. The flushing of the audio seems to create this. If I
On 28/05/14 23:46, Jason Blum wrote:
I shouldn't need to link avutil explicitly to use av_malloc( ), right?
Or has that been changed in more recent ffmpeg versions?
No, you must link it. It was always like that.
Unfortunately I have no idea what is controlling the linking order in
a ROS
On 26/05/14 10:24, caderbe.mat...@libero.it wrote:
Hi,
I want to create an ancoder that take a coded jpeg images end create video
using MJPEG codec in order to save time and do not encode and recode the
images.
Can you help me?
Matteo
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On 16/05/14 17:02, Info || Non-Lethal Applications wrote:
On 16 May 2014, at 19:02, Gonzalo Garramuno ggarr...@gmail.com wrote:
On 16/05/14 10:27, Info || Non-Lethal Applications wrote:
Hey there,
I’m trying to encode a sequence of images into Pro Res 422 in the MOV container.
I already had
I have trouble understanding what the following line in muxing.c does:
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx,
c-sample_rate) + src_nb_samples, c-sample_rate, c-sample_rate,
AV_ROUND_UP);
I was always expecting that:
On 16/05/14 10:27, Info || Non-Lethal Applications wrote:
Hey there,
I’m trying to encode a sequence of images into Pro Res 422 in the MOV
container.
I already had a look at the example: muxing.c and it looks promising.
However, I can’t figure out how to tell the encoder to use ProRes
I have an application that saves an .avi using the default mpeg4 encoder.
Since my application is GPL, I wanted to add H264 encoding. I compiled
libx264 and linked without any problem.
When I try to save with the h264 codec changing the code that works for
AV_CODEC_ID_MPEG4 (to
I tried modifying muxing.c and that works without problems.
Any help or ideas are welcome.
Ok. Some more info. The avcodec_video_encode2 returns got_packet false
for 18 frames at the beginning. This seems to happen also in muxing.c.
My video is very short (40 frames). And 19 frames are
On 14/05/14 15:06, Anshul wrote:
It looks like end frames are cached inside, you might need to call
avcodec_video_encode2 till u recieve all buffered frames.
-Anshul
Thanks. I figured it out by myself after I sent the mail. I call
avcodec_video_encode2 with a NULL pointer.
I am now having
On 03/05/14 17:21, Jorge Lúcio wrote:
The output could be BMP or PNG. I've tried to write my own function,
but so far the best I could do was to write a black and white version
of the frame. It's hard to figure out how the image data is stored on
the data field inside the AVFrame.
There
On 27/04/14 09:06, wm4 wrote:
Wouldn't it be better to make the exr decoder output float frames? That
would be the harder but cleaner solution, because we don't have a float
pixel format yet, apparently.
That would be ideal, but it is something I lack the knowledge (and time)
to do as it
On 27/04/14 19:29, b.mcdon...@sky.com wrote:
I have spent the last few weeks learning about libav to try to decode
mp3 files so I can create videos with the audio from the mp3 files.
Let me know if any more information is required.
Thanks
Bradley
Bradley, more info is required. Can you post
On 27/04/14 20:03, b.mcdon...@sky.com wrote:
Thank you for your response Gonzalo.
Maybe my code sample in http://sourceforge.net/projects/mrviewer will
help you out. You should look at the file called CMedia_audio.cpp and
aviImage.cpp.
The viewer plays mp3 and has never needed to call the
On 27/04/14 20:54, b.mcdon...@sky.com wrote:
I've had a look at your code, thanks. It seems you are using
av_read_frame which for me crashes with mp3 files as previously described.
I have just quickly written some code which has the crashes in when I
run it - I hope this can help somebody to
Find attached a first proposal for a patch to the exr reader to support
a gamma flag. Currently the half float routine is unoptimized and I
will change it in the near future.
diff --git a/libavcodec/exr.c b/libavcodec/exr.c
index 084025a..59f7dad 100644
--- a/libavcodec/exr.c
+++
On 07/04/14 16:49, Gonzalo Garramuno wrote:
I am adding muxing to my video player and I am running into a wall.
If I use mp3 (libmp3lame) as the audio encoding method, the resulting
file has perfect audio that ends up short (like 256 frames). If I use
PCM_S16LE, the audio is complete albeit
I am adding muxing to my video player and I am running into a wall.
If I use mp3 (libmp3lame) as the audio encoding method, the resulting
file has perfect audio that ends up short (like 256 frames). If I use
PCM_S16LE, the audio is complete albeit I end up with some stutters in
the audio
Title says it all. I am wondering what are the best (or some of the
best) codecs that are LGPL and that can be used from the ffmpeg API.
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On 03/04/14 12:52, Gonzalo Garramuno wrote:
Title says it all. I am wondering what are the best (or some of the
best) codecs that are LGPL and that can be used from the ffmpeg API.
This is for both audio (flt) and video (hd).
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On 27/03/14 14:23, wm4 wrote:
Then you should see the same error when seeking with ffplay. (AFAIK you
have to click with the mouse to seek or something.)
No, I don't see the error on ffplay. I also don't see the error on other h264
movies. Only in one I am seeing this error. It would be
After upgrading to 2.1, I am getting in my player for a h264 movie.
illegal short term buffer state detected
whenever I stop the playback and resume it immediately. When I stop the
playback I flush all caches.
I am wondering if someone can explain what might be causing it.
ffplay seems
On 27/03/14 11:55, wm4 wrote:
If by flushing the caches you mean dropping packet data and/or
flushing the decoder just for pausing, you can't do that.
I drop the packets and flush the decoder.
Either you don't drop/flush anything, or you have to issue a seek to
resume playback properly.
I do
On 12/03/14 22:23, YIRAN LI wrote:
Hi all,
I'm looking for a way to generate ffmpeg static libs for Windows (not
import .libs + dll) so that my application no longer needs dlls.
If I configure with --enable-static --disable-shared, then I can only get
.a files (compiled under both MinGW and
On 13/03/14 03:57, Carl Eugen Hoyos wrote:
Gonzalo Garramuno ggarra13@... writes:
I can open a context and see the return value. However,
this returns true for image formats like jpeg, png, and
others and I find those handled better with other libraries.
Then remove image2 from your
On 13/03/14 12:15, wm4 wrote:
Personally I just use av_probe_input_format2() and accept the result
only if it's at least AVPROBE_SCORE_MAX / 4 + 1, which is documented
as the recommended limit. I also don't set the filename, so no
extension matching is performed. (Although I do a second pass
I am wondering, in code, what's the most reliable way to probe a file to
see if ffmpeg will open it. I can open a context and see the return
value. However, this returns true for image formats like jpeg, png, and
others and I find those handled better with other libraries. Is there a
way to
On 15/02/14 04:56, Paul B Mahol wrote:
On 2/15/14, Paul B Mahol one...@gmail.com wrote:
Also the layer option is not used at all
Yes, I know. I tried just printing it with av_log and it crashes with
segfault. The layer option works fine command-line (it asks for a
string), but the
On 15/02/14 04:56, Paul B Mahol wrote:
Also the layer option is not used at all
Here's a patch with layer being printed that segfaults. It seems the
field is never initialized nor parsed, even though command-line works.
diff --git a/libavcodec/exr.c b/libavcodec/exr.c
index
On 15/02/14 18:09, Paul B Mahol wrote:
On 2/15/14, Gonzalo Garramuno ggarr...@gmail.com wrote:
On 15/02/14 04:56, Paul B Mahol wrote:
Also the layer option is not used at all
Here's a patch with layer being printed that segfaults. It seems the
field is never initialized nor parsed, even
Here's the final patch that works out the layers for exr. This patch
works fine if run under valgrind but segfaults otherwise if run
normally. The problem appears to be the parsing of options (and thus in
my options definition I guess). Commenting out: priv_class =
exr_class, makes it work
On 15/02/14 20:02, Paul B Mahol wrote:
I'm not clear enough: AVClass is missing in EXRContext struct.
Thanks for the explanation. Now it all works fine. What's the proper
way to submit a patch to get it added; this being my first contribution
to ffmpeg. Should I send it here or at
I was wondering if ffmpeg ( thru the exr reader ) has a way to specify
channels different than the standard rgba channels. If it can, can
someone provide an example of the command-line.
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On 14/02/14 14:48, Paul B Mahol wrote:
No, it can not.
Thanks. Looking at the exr code it seems it would be easy to modify it
to support other channels. However I am unfamiliar how to pass options
down from ffmpeg_opt.c to the libavcodec/exr.c file.
Maybe someone can point to a good
On 14/02/14 19:17, Paul B Mahol wrote:
See for example libx264 encoder, by using AVOption ...
Okay, I tinkered a bit but I cannot get the option to work. Here's my
patch against latest head.
diff --git a/libavcodec/exr.c b/libavcodec/exr.c
index f231b70..93a0622 100644
--- a/libavcodec/exr.c
On 09/02/14 03:52, Carl Eugen Hoyos wrote:
Gonzalo Garramuno ggarra13@... writes:
Duplicate of ticket #3662, thank you for the sample!
http://thread.gmane.org/gmane.comp.video.ffmpeg.trac/20273
Is it a duplicate?
What did your regression tests show?
Did you test the sample from that ticket
On 09/02/14 13:47, Carl Eugen Hoyos wrote:
Gonzalo Garramuno ggarra13@... writes:
On 09/02/14 03:52, Carl Eugen Hoyos wrote:
What did your regression tests show?
Did you test the sample from that ticket?
Please do so if you believe my findings are wrong.
I tried the sample from #3362
On 09/02/14 15:39, Carl Eugen Hoyos wrote:
Both samples fail horribly if you try to seek
backwards.
Carl Eugen
Okay. Thanks for all the help, Carl! Any possible ETA on when there
might be a fix for it?
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I have an mp4 clip that has sound and it disappears on seek. That is,
even in ffplay, a seek causes audio to go mute.
I've uploaded the file at:
http://www.datafilehost.com/d/885d526c
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On 08/02/14 20:08, Gonzalo Garramuno wrote:
I have an mp4 clip that has sound and it disappears on seek. That is,
even in ffplay, a seek causes audio to go mute.
I've uploaded the file at:
http://www.datafilehost.com/d/885d526c
More info. The sound disapppears in a seek after the clip
On 08/02/14 20:31, Carl Eugen Hoyos wrote:
Gonzalo Garramuno ggarra13@... writes:
I have an mp4 clip that has sound and it disappears on seek.
That is, even in ffplay, a seek causes audio to go mute.
Duplicate of ticket #3662, thank you for the sample!
http://thread.gmane.org
On 23/01/14 08:04, anshul wrote:
Hi
Shell32 is an qt library, I would like to know what part of ffmpeg is
using shell32 library
No, it isn't. Shell32 is a windows library. Probably cygwin uses it.
Use mingw to remove those dependencies.
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On 24/12/13 02:53, Carl Eugen Hoyos wrote:
Should be fixed, thank you for the sample!
Merry Christmas, Carl Eugen
I can verify the problem is solved. Thanks for the rapid response.
Merry Christmas to you and your family.
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On 22/12/13 20:49, Carl Eugen Hoyos wrote:
Does iTunes (and WMP) really show the image?
Thank you, Carl Eugen
I cannot speak for iTunes, but vlc shows it properly.
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Here's the log of some mp3's with embedded PNG files that are failing to
display.
I can upload the mp3 if someone points me where to ftp it to.
ffplay version N-58229-g459c7cb Copyright (c) 2003-2013 the FFmpeg
developers
built on Nov 17 2013 11:30:41 with gcc 4.6 (Ubuntu/Linaro
On 22/12/13 17:57, Carl Eugen Hoyos wrote:
Gonzalo Garramuno ggarra13@... writes:
I can upload the mp3 if someone points me where to ftp it to.
Either use http://www.datafilehost.com/ or read
http://ffmpeg.org/bugreports.html (there is no
hard filesize limit).
I've uploaded into:
http
I would like to probe a file to see if my player can play it (it is a
valid mov, rm, wmv, avi, etc), but I don't know how to do it with the
ffmpeg libraries. I don't care if ffmpeg knows the codec at that point.
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On 03/08/13 00:14, Anshul maheshwari wrote:
Mai
On Aug 3, 2013 2:56 AM, Gonzalo Garramuno ggarr...@gmail.com
mailto:ggarr...@gmail.com wrote:
I copied the latest docs/example muxing.c and decoding_enconding.c
and modified it to my needs.
I am able to save a movie with synced sound if I
I copied the latest docs/example muxing.c and decoding_enconding.c and
modified it to my needs.
I am able to save a movie with synced sound if I use audio codec
pcm_s16le and convert the float values into s16 values using
swresample. However, when I want to save ac3 to keep the data in float
On 27/07/13 13:35, Liang Zhang wrote:
Hello,
I tested the example program of demuxing for decoding audio signals
and have received the information regarding unsupported audio format
for ftlp when the input data has the extension of mp4. I checked the
program and found that the example
I have coded an encoder in my viewer that writes out new movie files.
My problem is trying to use AC3 as the audio codec. When I use it, the
files created are out of sync (audio plays much faster). When I write
out the movie using the PCM_16LE codec, all is fine (but of course, I
don't have a
I have an application that creates a movie file with sound. However,
when the video is played in my application I get a bunch of errors like:
[ffmpeg] freeing incomplete packet size 3224, new 9738
and the audio is wrong. I am wondering what does the above error mean.
On 02/07/13 08:49, Carl Eugen Hoyos wrote:
There is a patch that you could work on:
http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/93750
Carl Eugen
Carl, that set of patches is 3 years old and does not compile due to
deprecation of URLContext.
On 03/07/13 14:07, Stefano Sabatini wrote:
No if you see my more updated patches. That said, the code is still
far from being production-ready, so it may require a significant
effort and a good understanding of involved libraries/standards.
___
I've gotten to the point in my player in that I want to add support to
play DVDs.
However, I don't know where to begin. ffplay plays /dev/sr0 (scene
menu) and VOBs, but more is needed.
I was hoping I could borrow a couple of calls from dvdread and be on my way.
However, dvdread seems to return
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,
I am wondering what is the best method to use in reverse playback.
Currently I seek to some frame and decode all images/sounds that are
smaller than that frame (keeping a cache of frames).
This works rather well for most movies, except with the internal cached
buffers of ffmpeg. I've tried
On 13/06/13 17:26, Gonzalo Garramuno wrote:
I am wondering what is the best method to use in reverse playback.
Currently I seek to some frame and decode all images/sounds that are
smaller than that frame (keeping a cache of frames).
This works rather well for most movies, except
I am wondering what's the difference between the two formats (Dolby 5.1)
vs (Dolby 5.1 (side)).
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On 29/05/13 12:57, Carl Eugen Hoyos wrote:
I did not look at your code, but did you already test the following? $
ffmpeg -i input51 out.wav If out.wav shows your problem, please report
how to reproduce the bug.
I digged into what's happening. LibSDLaudio does some swizzling of
channels to
I am using ffmpeg and libswresample to play back movie trailers. For
stereo playback everything is good, but for Dolby 5.1, I get correct
playback of all ambient noises but the audio track for voices is missing
or too low of a sound, like a missing channel. My only weird
implementation if
On 29/05/13 12:57, Carl Eugen Hoyos wrote:
I did not look at your code, but did you already test the following? $
ffmpeg -i input51 out.wav If out.wav shows your problem, please report
how to reproduce the bug. Carl Eugen
Thanks, Carl. I already tested the problem with ffmpeg and ffplay. Both
On 21/05/13 02:29, Stéphane wrote:
Hi Gonzalo,
I use the binaries from Zeranoe (64 bits dll) without problem. What I
just did, was to build my own lib files using lib.exe from MS SDK tools.
First, I modified the def files from Zeranoe by adding LIBRARY ...
statement at the begining of the
I saw this codeccontext variable ctx-request_sample_fmt and I want to
set it to AV_SAMPLE_FMT_S16. I did so, but I get FLTP samples in return.
Is this supposed to work at all?
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On 29/04/13 10:19, Steffen wrote:
I want to provide a list of codesc that is supported by a file format. I try
to iterate over the codec_tag element in avoutputformat . But this is
impossible because it's private. What can I do?
--
View this message in context:
On 05/05/13 21:40, Gonzalo Garramuno wrote:
On 03/05/13 01:30, Kalileo wrote:
If you decode aac, then you might get hit by the change to planar
format, which happened November 16, 2012 in the ffmpeg sources
(although the effect is usually not a 'frying pan noise
On 06/05/13 11:51, Gonzalo Garramuno wrote:
On 05/05/13 21:40, Gonzalo Garramuno wrote:
On 03/05/13 01:30, Kalileo wrote:
If you decode aac, then you might get hit by the change to planar
format, which happened November 16, 2012 in the ffmpeg sources
(although the effect is usually
I placed a print statement with:
const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
and the return value for 1.0.6 is s16, while for 1.1.4 is fltp, which
I assume is float planar.
Does this mean I need to use the swresample library to resample the
sound to s16 which the
The call avcodec_decode_audio3 is deprecated. However it remains
functional. After the change in code in ffmpeg, avcodec_decode_audio3
fails returning valid audio and returns a 'frying pan' noise.
avcodec_decode_audio3 should be updated so that it works once again
transparently. This would
On 03/05/13 01:30, Kalileo wrote:
If you decode aac, then you might get hit by the change to planar format, which
happened November 16, 2012 in the ffmpeg sources (although the effect is
usually not a 'frying pan noise').
_
The problem also shows with wmav2.
I have a problem with the latest releases of ffmpeg. I was using code
that mimicked the code in avcodec_decode_audio3 calling
avcodec_decode_audio4. However, when I use that code now I get a frying
pan noise when playing the audio in my viewer. I reverted to
avcodec_decode_audio3 again and
make: *** No rule to make target
`libavcodec/x86/dsputil_rnd_template.c', needed by
`libavcodec/x86/dsputil_mmx.o'
After I manually added the file, the compilation later failed with:
make: *** No rule to make target `doc/all-components.texi', needed by
`doc/ffmpeg.pod'. Stop.
I am looking for the final word on pkt.pts/dts for movies that are not
streaming and how to hold a valid packet queue.
int got_pict = 0;
while( pkt.size 0 )
{
int err = avcodec_decode_video2( stream-codec, av_frame, got_pict,
pkt );
if ( err 0 ) {
I am looking for the final word on pkt.pts/dts for movies that are not
streaming and how to hold a valid packet queue.
int got_pict = 0;
while( pkt.size 0 )
{
int err = avcodec_decode_video2( stream-codec, av_frame, got_pict,
pkt );
if ( err 0 ) {
I looked at the latest ffmpeg and the demuxing example has a section for
cached frames, which get decoded with an empty pkt.
I was wondering how ffplay does without this or how a player is supposed
to be coded to support these cached frames.
___
I have written an open source viewer (mrViewer under sourceforge) that
works with frames. All is fine except playback of AVI and MOV files.
WMV movies play just fine.
With avis or mov formats seeking to the beginning of the movie will
offset all frames by two frames and will not allow to
In my viewer, I have a problem playing avi and mov files in that the
second and third frame are not displayed and the second frame becomes
the fourth continuing until end.
The problem, however, does not show with wmv files.
I'm guessing I have a pts/dts problem in my viewer but for the life of
I have a routine that saves a movie from sequential frames.
However, when the filename is a .wmv file, av_guess_format
seems to return a corrupt context. The result of saving is
a movie file with just 3 frames.
I also tried muxing.c from head but it did not compile. After a change
in defines I
On 28/09/12 09:01, Stephane Debusne wrote:
Hi all,
i'm developing a graphical application using libav libraries. The
application typically takes one MJPEG avi file and a wav file as input
and output a muxed video file.
I Know that ffmpeg.exe does it well but i have to do it inside my
I am trying the muxing example from docs/examples in recent git and I get
two problems:
With audio on, audio responds:
[NULL @ 0x1e4eee0] Codec is experimental but experimental codecs are not
enabled, try -strict -2
could not open codec
The second issue is that when I comment the audio and leave
2012/9/7 Arash Cordi arash.co...@gmail.com
most video formats only support yuv. there's nothing you can do about it.
why would you want to do that anyway?
I have my movie data in RGBA and would like to avoid translation to encode
faster and to keep the data the same.
I am trying to encode a movie file (avi or quicktime) as rgba. But I lack
source code with examples, except ffmpeg which is too complex.
Can anyone help me? Thanks in advance.
--
Gonzalo Garramuño
ggarr...@gmail.com
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TItle says it all. I'm looking for a tutorial or for code that shows how
to encode a movie or avi file with rgba as the color space.
--
Gonzalo Garramuño
ggarr...@gmail.com
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Libav-user@ffmpeg.org
I have some VOBs which show with no subtitles when in fact they have
them. The problem shows up in ffplay and my viewer, but not in VLC
(which adds the titles after the movie is running). Is anyone aware of
this problem? Sometimes the first VOB will not show subtitles, but the
second VOB in
I've created and released a video and flipbook player that relies on
ffmpeg. This was my first attempt at video work, so it is rather crude
imho. The program can be downloaded from sourceforge.net at:
http://sourceforge.net/projects/mrviewer/files/
Source code is in git at:
git clone
I am using ffmpeg in an application and some DVDs refuse to play back
their VOBs. This happens only with some DVDs and those that happen play
perfectly in Windows with Media Player.
I also tried with VLC and Dragon Player and both also fail in the playback.
Is there anything I can try or do to
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