HI all
I am using ffmpeg-r25759-swscale-r32562 files and trying to decode the raw
media packets of FLV file haivng the codec type vp6f.
The Initcodec function is returning fine but the avcodec_decode_video2 is
not able to decode the frames.
Is there anything specific we need to do fro vp6f?
i h
our code is sound. The thing is,
> the code to read any one of those files should be relatively the same.
>
> If you are getting segfaults, it's most likely an implementation issue, not
> a library issue.
>
> On Wed, Mar 30, 2011 at 2:29 AM, NITIN GOYAL wrote:
>
>> HI
i have used it on windows after compiling my builds using a cross compiler
on the ubuntu and i have created my own player.\ using ffmpeg.
i am not seeing any kind of this issue.
is the issue persists on all types of windows machine?
On Thu, Apr 28, 2011 at 10:42 AM, Saurabh Gandhi wrote:
> Hell
nd. Can you please download this and check if
> the ffplay.exe is working fine for decoding h.264 based RTSP stream?
>
> --
> Regards,
> Saurabh Gandhi
>
>
>
>
>
> On Thu, Apr 28, 2011 at 12:30 PM, NITIN GOYAL wrote:
>
>> i have used it on windows after compi
Hi
I am not finding any way why someone will need this information for this
project.
You can contact admins of ffmpeg user list for this info guess.
Thanks
Nitin
2011/4/29 Marián Dániel
> *Hi to all on the Mailing list!*
>
>
>
> *We are currently working on a project, wich needs references fr
u wil get the data in YUV format...
On Thu, May 19, 2011 at 6:19 PM, pavan kumar wrote:
> how can we say the decoding is successful?
>
> how to see the decoded data stored in the 2nd variable of the
> avcodec_decode_video2() function
>
>
> ___
> Libav-u
Hi
I am trying to write a parser for RTMP for a video conference solution
having more than 10 participants at a time and then further use ffmpeg
to decode the data received from that traffic.
I am not able to find out he suitable way to associate the different
participants data being displayed by
Hi
I want to decode the raw *nellymoser *samples which i have extracted from a
pcap in a raw file.
Now, this raw file have all the captured Nellymoser samples but i am not
able to figure out how can i decode them or transcode them into some format
using ffmpeg exe.
I mean i am not able to find t
Hi
I have decoded the nellymoser audio coded data using ffmpeg using
avcodec_decode_audio3 function and I got the pointer with the raw data but
the other metrices like sampling rate, bit rate all returning the default
values. Nevertheless, i have saved that raw data into the file.
Now, i want to
Hi
I am trying to decode the H.263 sorenson type of data using
avcodec_decode_video function.
It is able to decode the I and P frames but i have some *D frames
(Disposable inter frames) *in my data and this function is not able to
decode these frames and I am getting zero as return value.
So, ca
Hi
I want to decode the 3gpp content having the video codec mp4v-es codec.
What will the appropriate fourccc for decoding this I guess it should be
mpeg4?
Also, i want to know do we need to send any header information to the
decoder prior to sending the raw video packets like in case of h264 dat
Hi
It seems to be the nice effort and UI looks good.
It sees to be a file conversion app from one format to another and in
general its same as other data conversion tools in the market.
Can you tell me what is the new thing or special thing you have added in it
which is nt commonly available?
A
Hi
I have some audio with G.711 and g.729 codecs.
I am able to decode the g729 but not g711.
So, I ant to know do FFMPEg supports g711 at all? It has two flavours i
guess:- u-law and a-law.
So, if somebody has any idea how to decode g711 codec with ffmpeg do let me
know.
__
Hi
I am trying to decode some Amrnb audio using the avcodec_decode_audio3
function. I want to input frame by frame and save the raw output frame by
frame as well.
*int len = avcodec_decode_audio3(mpCodecContext, (short *)outbuf,
&out_size, &pkt);*
Now, the issue is that when I give the input of
Hi
Do FFMPEG also supports the depacketization of the
Bandwidth Efficient Mode packetized RTP payload for AMR audio content?
in *rtpdec_amr.c*, I am seeing that the algo used seems to be having the
octet aligned format only.
I am looking for the clear algo which can be used to depacketize the R
Hi
Do anybody have idea that how can i convert the RTP packetized AMR audio
content into standalone AMR file?
I have the dump of RTP payload and I want to convert it into the standalone
AMR file. I have followed up RFC 3267 and RFC 4867 and understood the byte
pattern but I havenot found anywhere
Hi
I have used the *avcodec_decode_audio3* function to decode the amr content
in the frame order.
I am getting the output buffer of 640 bytes for each frame and Sample
format is float and I have saved that content as well in a raw output file.
Now, the issue is that I want to validate that this
Hi
Decoding output is generally 32bit float or you can PCM float output.
Why do you mean by mapping the data??
I dont think you will be bale to map up the bytes of of both decoder.
Also, once you have decoded the content, what you want to do with the
ffmpeg using that decoded data? You want to
Hi
I have upgraded my ffmpeg version to the latest commit and Now i can see
that the audio decoding funciton *avcodec_decode_audio3* has been
deprecated and when i use the new function*avcodec_decode_audio4*, as per
the changes required in it, I get the error as
*[amrnb @ 003a5000] get_buffer() f
idea about this error?
Regards
Nitin
On Wed, Feb 22, 2012 at 6:11 PM, NITIN GOYAL wrote:
> Hi
>
> I have upgraded my ffmpeg version to the latest commit and Now i can see
> that the audio decoding funciton *avcodec_decode_audio3* has been
> deprecated and when i use t
Hi
I am seeing that the this example has been updated 10 days ago but that
also have avcodec_encode_audio example.
see this link
https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/decoding_encoding.c
I am not sure when this encode_audio2(0 has been release. If its released
in between th
I want to know the significance of Marker Bit in RTP for Voice packets and
is here any RFC which tell that.
I know that the for the Video packets marker bit means last packet for the
same image and hence, its the last packet with PTS time-stamp corresponding
to image but for the Voice Packets for
i think this disclaimer has been added by the mail exchange sever
automatically so nothing can be done for it and its better to avoid it. :)
On Tue, Mar 27, 2012 at 7:15 PM, Ajita Pandey wrote:
> Hi,
>
> Yes I talking about the disclaimer here.
>
>
>
> -Original Message-
> From: libav-us
Hi
I am trying to decode the mp4a-latm content using ffmpeg. I have used the
CodecID as Codec_ID_AAC_LATM and extracted the config parameter from the
SDP and further extracted the audio content from the RTP by removing the
rtp header and sent it to decodeAudio4 function but the decode function
alw
Read the dranger docs on how to decode the NALs or see the example program
from libav source to start with.
And if you have already done that, please share the issue you are finding.
On Mon, Apr 9, 2012 at 3:21 PM, Kalileo wrote:
>
> On Apr 9, 2012, at 16:13 , srikanta mondal wrote:
>
> > Hi al
Hey
do all ur raw payload u r sending to the decoding function are in NAL
format like 0001xRawpayload format??
I have done the same thing with the RTP packets. Removed RTP headers
inserted NAL start prefix headers i.e. 0001 before each NAL and then
entered the data to decoding functions.
And if
Do ffmpeg supports stream with IPsec and GPRS tunneling (GTP)?
I have not found any reference of any of them?
I have a RTP stream which is GTP and protected by IPSec so, will i able to
decode or depacketize the content using ffmpeg?
Any idea on this.
If someone have idea about any other tool fo
I think it is always not necessary have to both audio and video with same
size. I have seen the videos which are greater in length that audio and
there was silence at the end of the video. If there is no silence in the
end of your video then you need to track the PTS values of the audio
and check
Hi
How can we extract the raw AAC frames from an mp4a-latm content packetized
in RTP packets?
I have gone through the RFCs 3016 but nothing is concrete and much clearer.
I have also seen the code of ffmpeg as how they do it. I have understood
that they are using the config data and parsing it bu
Any sort of help on this will be highly appreciated.
Regards
Nitin
On Sun, Apr 15, 2012 at 8:49 PM, NITIN GOYAL wrote:
> Hi
>
> How can we extract the raw AAC frames from an mp4a-latm content packetized
> in RTP packets?
>
> I have gone through the RFCs 3016 but nothing is
There are different branches of thr libavcodec and anyone using libavcodec
can use any of these branches rather than the mian branch as per the
requiremnt.
Some different devopers have created some feautre which is not part of main
branch and may be VLC is using that and thus the code difference i
egards,
>
> Ajita
>
> ** **
>
> *From:* libav-user-boun...@ffmpeg.org [mailto:
> libav-user-boun...@ffmpeg.org] *On Behalf Of *NITIN GOYAL
> *Sent:* Tuesday, April 24, 2012 8:44 PM
> *To:* This list is about using libavcodec, libavformat, libavutil,
> libavdevice an
Hi
I want to decode *Enhanced Variable Rate CODEC* (*EVRC*) and its other
flavors like EVRC0/EVRC1/EVRCB etc. But it seems like EVRC is not being
supported by ffmpeg main trunk.
Is there any way i can decode this content using ffmpeg library of or any
of its other branches if they have its suppor
Hi
I am trying to decode G726 audio codec but when i give string "g726" to
initcodec funciton, but ffmpeg is not able to detect the codec and return
-22 error.
Can someone help me what -22 error says?
If i will use CODEC_ID_ADPCM_G726 will it work?
I have the G726-16 codec over a stream.
Regar
I am also working on the Video quality analyzer tool.
But mine is a No reference one where i don't need to compare my video
quality to the source content.
But for measuring the quality, it depends on the parameters you want to
take into consideration and if you can provide a MOS values in last ba
wht do you mean by sometimes?
if its not consistent then i wonder the data frame you are sending are not
correct.
if its consistent and always results zero then either the NAL formation
have some issue or you are missing some important configuration.
On Fri, May 25, 2012 at 10:21 AM, wrote:
>
Hi
Is there any way i can modify or use the ffmpeg library to detect the
black/dark/green frames from the content?
Theoretically, we can do it by calculating the luminance of each frame and
finding with higher values but i am thinking if that can be
done through ffmpeg?
Any sort of advice on thi
i think theoretically its possible but it might be bit complex..
the following link might help you:
http://libav-users.943685.n4.nabble.com/Output-mpeg-ts-to-rtp-td2234066.html
Regards
Nitin
On Wed, Jun 6, 2012 at 8:45 PM, Brad O'Hearne wrote:
> Any ideas on publishing / sending MPEG-TS over
Hi
I have a requirement that i need to create a Video file with multiple
resolutions as the raw content I am getting on the stream is of multiple
resolutions and keeps on switching the resolutions from time to time.
So, i will capture that content and transcode to an avi file with the
multiple re
Hi
Playing with stream means?
You can use mplayer for playing it or can create your own player like
mplayer using ffmpeg. Yeah all the feature you have mentioned are very well
supported by libav.
VLC is also using libav for many codecs.
Regards
Nitin
On Tue, Jun 26, 2012 at 1:45 PM, Naresh San
please mention the libav version you are currently using then only people
will be able to tell you the differences.
Also, please check the source code from GITHUB and compare the differences
from the previous version to check out.
Regards
Nitin
On Thu, Aug 16, 2012 at 7:22 PM, salsaman wrote:
I think if you can explain your requirement it will be much better and what
kind of support you are actually looking for?
On Thu, Dec 13, 2012 at 9:12 PM, Mike Versteeg wrote:
> Anyone willing & able to offer me (paid) support?
>
> Thanks,
>
> Mike
>
I think the error you are mentioning is related to installation and
not at all related to the compilation as you have mentioned in your
first mail.
Please check why the installation got failed. May be you can check the
content of config.log and try to see.
BTW, there are many other ways to instal
installation Failed :( , please visit the forum^[[0m
>
>
>
> Please help me for the same.
>
> On Fri, Jan 11, 2013 at 4:50 PM, NITIN GOYAL
> wrote:
>>
>> log
>
>
>
>
> ___
> Libav-user mailing list
> L
On Wed, May 14, 2014 at 10:30 AM, Ken Bass wrote:
>
> Question: can I simply write out each frame as I receive it, with an
> appropriate PTS? This would most likely skip expected frames. Also, the
> captured frame most likely wouldn't be written at the specified fps
> interval.
>
>
I think I am n
are you sure that the input source audio dont have any noise?
The code looks ok as you are just sending te buffered content to the decode
func.
On Tue, May 13, 2014 at 1:11 AM, Ankush wrote:
> This is my sample audio code for reading a fixed size of buffer.
> everything works just fine in it
It should have some threshold and shd not wait for infinite time.
If this a bug, please report on the bug tracker.
On Wed, May 7, 2014 at 8:13 PM, baneyue wrote:
> Hi:
>
> I'm working with http video stream using ffmpeg library, when i use
> `avformat_open_input` to open the st
Cn you please be more specific?
Like in case of mp4, for subtitles there can be separate atoms which can be
inserted and played as the offset for a/v works there.?
If you can explain what you have tried with subtitles and what is causing
issues, it will more helpful for anyone.
On Wed, Nov 5, 20
On Tue, Nov 18, 2014 at 9:19 AM, user wrote:
> other PES streams I want to
you have to encode them in TS if you want to stream with Mpeg-TS over UDp
directly or within RTP pkts. PES can also be in that with all other values
for sync etc
___
Libav-use
On Tue, Dec 30, 2014 at 11:04 PM, Clément Champetier
wrote:
> *Invalid data found when processing input*
check the debug info from ffplay and see if its doing some error
concealment etc. I think it should play straight forward if all the data
for NAL is being available there in it.
___
Please check out this link which explains in detail:
http://jiasi.blogspot.in/2011/05/motion-vector-extraction.html
if you face any other issue, please post the specific question.
On Tue, Dec 30, 2014 at 8:52 AM, Shuliang Zheng wrote:
> Hi all,
>
> I am working on an experiment which needs MV
On Thu, Dec 25, 2014 at 12:43 AM, nizar aloui wrote:
> I managed to make a local patch in order to reconnect the socket but I do
> have another issue with the same hls stream and is that some times the
> connection drops with the latest displayed message print:
Can you please check the wiresha
It looks odd coz as per my understanding GOP with B frames generally ends
with B frames and not P frame as you are getting. Can yu check if its open
GOP or closed GOP or if you are missing to get the last B frames while
interpreting?
On Fri, Dec 19, 2014 at 8:38 PM, Valentin Noël wrote:
> Hi to
On Wed, Mar 11, 2015 at 8:10 PM, Rafael Lúcio wrote:
> Its very simple... everytime I call av_read_frame( input_context,
> input_packet ) I save the pts into the last_pts variable...
One way is to make a buffer and a queue and reorganize the PTS in
increasing order before processing.
I think there will be multiple mp4 files in there. You need to extract them
and merge them in order.
On Thu, Mar 12, 2015 at 8:46 PM, Marco Nespeca wrote:
> Hello,
>
> Does anyone know how to convert popcorn.js player files into a video file
> format? MP4
>
> Can anyone point me in the right dir
you need the sequence and the timestamps to do that for both jpeg and wav
file.
What kind of sample you are looking for?
On Thu, Mar 12, 2015 at 8:33 PM, Pierrick Chabi
wrote:
> Hello,
>
> Is there a recent sample that illustrates how to programmatically convert
> a sequence of jpeg files and a
yup GOP size could be one of the param.
On Sat, Mar 14, 2015 at 4:23 PM, Rafael Lúcio wrote:
> what about the size of this buffer before processing?
>
> gop_size ?
>
> 2015-03-13 23:48 GMT-03:00 NITIN GOYAL :
>
>>
>> On Wed, Mar 11, 2015 at 8:10 PM, Rafael Lúcio
In this case only PTS matters if the simple raw data is in RTP packets.
If there is MPEG then you have to use PTS and PCR values as well.
Generally, data sync happens during the playback and the simple stream are
stored in the file.
On Wed, Mar 25, 2015 at 4:15 AM, Alessio Volpe
wrote:
> Thanks
On Thu, Mar 26, 2015 at 9:29 PM, Reddi, Praveen wrote:
> .
>
> Then I prepended 4 bytes *[0x00, 0x00 , 0x00, 0x01]* to the byte buffer
> before sending to the decoder, because I didnt receive any SPS, PPS NAL
> bytes.
>
> When I send this byte buffer to the decoder, decoder fails when it tries to
yes and they will get decoded only as per DTS otherwise will get queued up..
On Fri, Apr 17, 2015 at 6:40 PM, Ran Shalit wrote:
> Hello,
>
> This is a general question only for help in understanding libavformat API.
> I am trying to understand the demuxing example in FFmpeg wiki.
>
> while (av_r
I dont think there is some specific need but may be some user can do it and
make it available for specific needs as FFMPEG mainly uses soft decoding.
On Thu, Apr 16, 2015 at 6:11 PM, Mert Gedik wrote:
> Hello,
>
> I just wonder that if there is any implementation or any plan to use hw
> decoding
RTP payload 96,97 are dynamic payloads and can be assigned as needed. RFC
2833
in general when we have both video and audio, Video goes for 96 while audio
goes for 97.
On Wed, Oct 26, 2016 at 7:08 AM, Yu Ang Tan wrote:
> On Thu, Sep 22, 2016 at 11:46 AM Yu Ang Tan wrote:
>
>> I want to crea
As its a dynamic payload profile, you have to assign it somewhere.. ffplay
won't be able to decode the sdp untill its specifically defined about the
profile type...
On Sunday, October 30, 2016, Yu Ang Tan wrote:
> On Sun, 30 Oct 2016 18:33 NITIN GOYAL > wrote:
>
>> R
I guess you can only get the frame rate but calculating the max/min needs
to be done by you explicitly by keeping a separate variable to monitor
those values and keep on updating them while traversing the complete stream.
On Tue, Nov 1, 2016 at 7:37 AM, YIRAN LI wrote:
> AVFormatContext
__
On Tue, Mar 21, 2017 at 3:16 PM, Prakash Rokade wrote:
> Is there any example that will encoding the live sources.
>
https://ffmpeg.org/doxygen/trunk/encoding-example_8c-source.html
This example have audio encoding.
___
Libav-user mailing list
Libav-u
You can easily transcode rtmp to rtp using ffmpeg.
Use - I url - c copy - f rtp_mpegts rtp://127.0.0.1:10001
For iOS, you might need hls stream with m3u8 to make it work I guess.
On Friday, December 14, 2018, UTKARSH AGARWAL wrote:
> I am working on a screen sharing iOS application . Getting
LinkedIn
Nitin Goyal requested to add you as a connection on LinkedIn:
--
Libavcodec,
I'd like to add you to my professional network on LinkedIn.
- Nitin Goyal
Accept invitation from Nitin Goyal
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