I advice you to go to Nvidia website and download their VideoSDK. All the
answers to your questions are there ))
On Mon, May 28, 2018 at 4:51 PM, Sergio Basurco <
sergi...@coherentsynchro.com> wrote:
> I have a nice H264 decoder based on this one:
>
> http://roxlu.com/2014/039/decoding-h264-and-
Not sure if this is the right place, but Nvidia Video SDK page contains
wonderful examples with runnable source code which shows you how to decode
and encode. That's not FFMPEG, but from what you're trying to do it sounds
like you can do most of it with Nvidia NVENC and NVDEC and just use AV lib
to
Hi guys, I am muxing video and audio into mp4. The final file looks ok,but
during the muxing session I am getting this error all the time. Can just
discard it or there is an issue I need to solve here?
*[mp4 @ 023c51dc7880] Delay between the first packet and last packet in
the muxing queue is
Hi. I will ask again,as nobody answered. I am performing muxing,based on
muxing.c
example.But in my case the audio comes from a file,which sometimes is
longer (in terms of duration) then the video stream. For some reason,even
that I am checking and comparing current times of both streams,at the
end
AV_CODEC_ID_PCM_S16LE
Yeah ,probably this is it. I write PCM 16bit. Thanks.
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user
You're right.Let me rephrase:
I need to encode the data so that the encoded frame contains PCM16 . What
codec ID do I need to use for that?
I see in the ID list all sorts of PCM and I am not sure which to choose.
Thanks.
___
Libav-user mailing list
Liba
Hi, what is the codec id for wav enoding? I see in
encode_audio.c AV_CODEC_ID_MP2
is used. So is this the correct id to write wav files?
Thanks.
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/libav-user
Hi all. I have a nasty bug. I have written a muxer,which is based on
muxing.c sample.
The only different is that in my case audio stream comes from AAC file.And
if the sound track's
time is longer than video total time,then the overall resulting video time
becomes larger than expected. Here is the
Hi, I am rewriting some messy FFMPEG code I wrote a while ago.
My reference is muxing.c
The difference I have is that my audio stream comes from file (AAC) ,while
video packets
come from generated pixel frame.
I mostly have it working,but my problem is interleaved packets write.If I
do it like the
Hi .
I am muxing a video stream. Setting 60 fps. But when inspecting codec info
in VLC it shows me FPS: 60.20668
Is it normal? In ffprob everything looks fine. VLC plays the video also
fine.
But Adobe aftereffects,fails to read the whole video fine.
What can be the issue here? I do all exaclty as
*AVCodecParameters is for information that codecs need to do their workbut
is stored in a structured way in the container format rather thanencoded in
the bitstream. Codecs vary in what they need and what theyput in the
bitstream, and formats vary in what they are able to encode,but there is
enough
en you need to
> tell the encoder to put its details in the header also.
>
> Salsaman.
>
>
> http://lives-video.com
> https://www.openhub.net/accounts/salsaman
>
> On Wed, Mar 14, 2018 at 5:31 PM, Michael IV wrote:
>
>> Hi Anton. I see what you mean, but I actually use
Hi Anton. I see what you mean, but I actually use that internal context,and
it works perfectly.
For video I am getting h264 NALs from somewhere,so I don't perform encoding
at all.
For audio, I am opening existing audio files/ streams with data in the
format I need,so I don't
transcode,but just pass
xing of input inStream into mAudioOutStream... So I don't get it.
On Wed, Mar 14, 2018 at 9:52 PM Michael IV wrote:
> Exactly:
>
>
> *For muxing/encoding:avcodec_parameters_from_context(stream->codecpar,
> codec_ctx); *
>
> That's what I am talking about.In
y I have to
create codec context just to get rid of those warnings.
On Wed, Mar 14, 2018 at 9:44 PM Philippe Gorley <
philippe.gor...@savoirfairelinux.com> wrote:
> On 2018-03-14 03:26 PM, Michael IV wrote:
> > Hi. I am trying to get rid of deprecation warning in the API. I am
>
Hi. I am trying to get rid of deprecation warning in the API. I am running
with
ffmpeg -20170711 version. I have a code where I don't explicitly create
AVCodecContext
because I multiplex already existing h264 NALS. But I still have to setup
AVStream which has AVCodecContext which it creates interna
Thanks for the tip.Will look into it.
On Wed, Mar 14, 2018 at 6:48 PM Gonzalo Garramuño
wrote:
>
>
> El 14/03/18 a las 09:03, Michael IV escribió:
> > Hi.I have the following case:
> > I am receiving audio stream which consist
> > of 2 channel float 32 (non plan
Well I cracked this one. The audio encoding example is not a good fit for
formats like aac. One has to use also output format and write container for
this thing to work as expected. Raw aac stream is useless.
On Mar 14, 2018 15:42, "Michael IV" wrote:
> Just a dumb quest
t; 2018-03-14 13:13 GMT+01:00, Michael IV :
> > So I am using that lib as you can see.
>
> Yes, sorry...
>
> Carl Eugen
> ___
> Libav-user mailing list
> Libav-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/libav-user
&g
So I am using that lib as you can see. Or you mean something else?
On Wed, Mar 14, 2018 at 2:10 PM Carl Eugen Hoyos wrote:
> 2018-03-14 13:03 GMT+01:00, Michael IV :
>
> > I resample from non planar FLT to FLTP as follows:
>
> While I didn't check, I would expect
Hi.I have the following case:
I am receiving audio stream which consist
of 2 channel float 32 (non planar) audio frames. Then I am trying to
convert those into
AV_SAMPLE_FMT_FLTP in order to encode with AAC codec. The problem is that I
receive that data as packets of size different from what my AV
Hi. I have a PCM 16 bit (signed) stream,which I want to encode using AAC
codec.
Now I am following the setup of audio encoding example and where I check
for
SAMPLE FMT support by the codec,the check fails:
while (*p != AV_SAMPLE_FMT_NONE)
{
if (*p == c->sample_fmt)
{
break;
}
p++;
}
Beca
I am streaming fragmented MP4 into browser video tag via
websocket.On the browser side I use MSE to accept and process fragments.
Now,as long as I stream only video,without adding AAC audio stream,it works
fine.
But it doesn;t work with audio at all.No video and and the sound playback
gets playe
Hi!
I am trying to reuse AVIOContext.
I call avio_flush() after finihsing muxing and before starting next session.
But in this case the output video is corrupted.The corruption is in the
header.The AVCC info is missing,which leads me to think that the first
thing the avio does at the
beginning of
Hi. I am trying to reuse already created instances of
AVFormatContext,AVCodecContext
and AVIOContext. My question is related to avformat_write_header() with
parameters.
At the end of the session I usually call avcodec_parameters_free to free
internal buffers.
Do I have also to call this method
Hi. I am trying to understand something regarding AVCC info when muxing
elementary h264 stream into mp4.SO far
I was adding extra data which is SPS and PPS that I retrieve from h264
encoder. But when I analized the mp4 file I found that all the AVC stuff
like
AVC profile
AVC level
AVC NALU length s
Btw, I turned off the auto spspps insertion on my h264 bitstream encoder
and put this info only into mp4 container header when muxing. It doesn't
work. The muxer produces invalid data.
___
Libav-user mailing list
Libav-user@ffmpeg.org
http://ffmpeg.org/ma
Which probably means this is not an important one? I am asking these maybe
"silly " question because I am streaming fragmented MP4 to browser. Firefox
has no problems, but Chrome has.It fails on the two first nals. That's why
I am trying to see maybe it's because I have spspps header embedded both
Thanks John. Can you tell me where I set that "sample entry "? Is it a part
of sps pps bits?
On Nov 14, 2017 6:07 PM, "John Stebbins" wrote:
On 11/14/2017 03:10 AM, Michael IV wrote:
> Hi. I am not sure about the following setup.
> I create elementary h264 bitst
Hi. I am not sure about the following setup.
I create elementary h264 bitstream using h264 encoder. I append SPS PPS
nals to the start of the stream because I stream this raw bitstream to some
presentation endpoint. But now I also have a use case where I wrap that
stream into mp4 container. Should
HI All.
If I need to stream fragmented mp4 chunks, what are the optimal flags?
I see in the web something like this:
"frag_keyframe+empty_moov"
or something like that:
empty_moov+faststart-default_base_moof
What effect does "default_base_moof" flsg have ?
Thanks.
___
Does ffplay.c shed some light? I am interested to learn this too.
On Nov 11, 2017 11:36 PM, "Patrick Cusack" wrote:
> I am looking for example code on synchronizing audio and video playback of
> a simple .mov container file (the streams could be comprised of ppm/aac and
> dnhd/prores/h264). I h
Guys, you didn't follow my last question. I don't need a help with the API
usage. I am quite fine with that. I tried to attract your attention to the
fact that after I retrieve the info my streams contain uninitialized
codecpar . Which looks weird. The ffmpeg version is from chrome. It's
hacked. A
re looking for.
>
> Good luck,
> Dan
>
>
> On Nov 9, 2017 at 9:38 AM, > wrote:
>
> But now I see in vlc that the MPEG4 codec is actually named "mp4v" .So
> that's the name I have to specificy in --enable-decoder when doing FFMPEG
> config
But now I see in vlc that the MPEG4 codec is actually named "mp4v" .So
that's the name I have to specificy in --enable-decoder when doing FFMPEG
config?
On Thu, Nov 9, 2017 at 7:26 PM, Michael IV wrote:
> And testing the actual codecId of the video stream in the file I am a
st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
* where ec_ctx->codec_id == AV_CODEC_ID_MPEG4*
On Thu, Nov 9, 2017 at 7:22 PM, Michael IV wrote:
> Hi, thanks for the input. Now I am confused even more. All I need is to
> start demuxing an mp4 file. Those codecs test abov
ing
location:
*pFormatCtx->streams[i]->codecpar->codec_type==AVMEDIA_TYPE_VIDEO*
Because my *codecpar *is not initialized. That's what I don't understand
here.
On Nov 9, 2017 8:14 PM, "Carl Eugen Hoyos" wrote:
> Hi!
>
> > Am 09.11.2017 um 17:13 schri
Hi All. What maybe a reason that avcodec_find_decoder returns NULL ?
I enabled in my FFMPEG build MOV and h264 demuxer. Configure report clearly
states that.
But doing this test:
av_register_all();
avcodec_register_all();
avformat_network_init();
AVCodec* h264codc =
Hi guys.
I have a weird error.
I want to open input file with avformat_open_input.
And I am getting the error "protocol not found"
Here is the code:
pCodecCtx = NULL;
pFormatCtx = NULL;
avcodec_register_all();
av_register_all();
avformat_network_init();
pFormatCtx = avform
I never did 1:1comparison, but it's more than satisfactory. And it's
blazing fast.
On Oct 29, 2017 3:52 PM, "Carl Eugen Hoyos" wrote:
> 2017-10-29 9:14 GMT+01:00 Michael IV :
> > I would recommend to invest into good Nvidia GPU and use
> > NVENC which o
I would recommend to invest into good Nvidia GPU and use NVENC which
outperforms libx264 and reduces CPU consumption dramatically.
On Oct 28, 2017 10:51 PM, "Carl Eugen Hoyos" wrote:
2017-10-25 22:09 GMT+02:00 Dave :
> Should I be wary of AMD though or do the more modern CPUs support
> the sam
Hy guys,need a little help here.
I am reading audio buffer from WASAPI which is in PCM float 32 format. I am
using
libswrescale to convert it to PCM S16
Getting a lot of noise along with the original sound .
I asked a related question on SO:
https://stackoverflow.com/questions/46645109/windowswa
I have tried with dash.js and with VLC. In fact vlc doesn't play even dash
samples which I downloaded from the web. What is CLI?
On Sep 29, 2017 12:40 AM, "Carl Eugen Hoyos" wrote:
2017-09-28 18:40 GMT+02:00 Michael IV :
> Hi all.
>
> I have two questions:
>
>
Hi all.
I have two questions:
1) is there any example how to encode video (h264) into MPEG DASH?
I tried doing it with setting output file name to be with extension ".mpd"
and
avformat_alloc_output_context2(&mOutputFormatContext, NULL, "dash",
filename.c_str())
This setup produces single .mpd fi
like uncompleted refactoring of the api.
On Sep 28, 2017 5:03 PM, "Power Pan" wrote:
On Thu, Sep 28, 2017 at 7:30 PM, Michael IV wrote:
> Hi!
> I am getting compiler warning for avcodec_parameters_copy being deprecated.
>
> But I also can't figure out how to use avc
Hi!
I am getting compiler warning for avcodec_parameters_copy being deprecated.
But I also can't figure out how to use avcodec_parameters_copy when I have
AVStream as destination and AVCodecContext as src .AVCodecContext doesn't
have AVCodecParamters property.
___
Hi!
I want to accumulate results of av_interleaved_write_frame() in a buffer
instead of a file.
avformat_alloc_output_context2() file name parameter can be null. That
probably means I should be
able to access some internal buffer with the write frames?Or can I supply a
custom buffer to write the
Ok,I think I got it.In my case I have to use av_parser_parse2() to get h264
frame into packet,then mux into interleaved frame.
On Wed, Sep 27, 2017 at 1:46 PM, Michael IV wrote:
> What I understood from doc/examples/muxing.c is that it operates on YUV
> frames.
> My data is raw h
What I understood from doc/examples/muxing.c is that it operates on YUV
frames.
My data is raw h264 NALUs stream from NVENC and I want mux those directly
into TS adding audio frames .
On Wed, Sep 27, 2017 at 1:36 PM, Carl Eugen Hoyos
wrote:
> 2017-09-27 11:19 GMT+02:00 Michael IV :
>
I join Ajay's question. I found this very old and incomplete example:
https://lists.libav.org/pipermail/libav-user/2009-May/003034.html
What it does it first decodes h264 from a file and writes it into stream.
The example has some code that attempts to setup audio frame but it
doesn't mux it int
I am trying to open input stream,and I found an example which seems to be pretty
old : https://lists.libav.org/pipermail/libav-user/2009-May/003034.html
So I am moving step by step converting the old API to the new one.
Now I have this part:
AVFormatParameters params, *ap = ¶ms;
memset(ap,
As far as I know h264 doesn't support alpha. But VP8 does.
On Sep 12, 2017 2:06 PM, "Davood Falahati"
wrote:
> Dear community,
>
> I have a opevCV Mat frame, call it mask. I decode frames into AVFrame,
> convert AVFrame to openCV Mat and encode them into AVPacket again.
>
> Now, I want to add
, how do I flip the frame vertically on decoding ?
On Jul 31, 2017 2:07 PM, "jing zhang" wrote:
You should use decoded AVFrame->format to check decoded yuv format.
Maybe H.264 bitstream encoded by NVENC is corrupted?
2017-07-31 16:27 GMT+08:00 Michael IV :
> So you basically say
aming from NVENC.
https://stackoverflow.com/questions/33185966/how-to-stream-h-264-video-over-udp-using-the-nvidia-nvenc-hardware-encoder
And I currently don't set those params for encoder.Do I have to?
On Mon, Jul 31, 2017 at 11:04 AM, Hendrik Leppkes
wrote:
> On Mon, Jul 31, 2017 at 10
Hi! I am using NVENC encoder to create h264 stream.The YUV format is NV12.
Now, I need also to decode that stream back to YUV.I wrote a module for
that,based on
this example:
https://gist.github.com/roxlu/9329339
But, because my YUV format is not 420P, but NV12 I am trying to force it on
AVCodecC
55 matches
Mail list logo