On Jul 15, 2014, at 07:54 , Michael Albanese wrote:
> I'm trying to modify dranger's tutorial02 in order to playback frames at the
> rate which they are meant to be played, I've tried the following:
>
> - (Close, but too slow / fast) Getting the FPS, sleeping calculated_delay -
> processing ti
On Jul 17, 2014, at 19:36 , Marco Sieber wrote:
>
>> Are you manipulating the timebase and the DTS / PTS from the input before
>> you give it to the encoding / muxing functions?
>> Are you passing the proposed DTS / PTS from the ffmpeg encoding functions to
>> the muxer, or are you manipulatin
Correction: correct -> incorrect
On Jul 17, 2014, at 18:08 , Kalileo wrote:
>
> You might also have different settings in the encoded data and in the
> container, and VLC uses the correct one, while QT might use the one with the
> _incorrect_ settings. (I have seen such a se
On Jul 17, 2014, at 17:14 , Marco Sieber wrote:
> >I wrote a Transcoder it works fine so far, in generell its X
> >Container/Format to MP4/H264.
> >We have a 20min File (5GB) its a QuickTime Movie (DNxHD36) the MP4 Output
> >works fine with VLC (no sync issue).
> >But if i open it with Quicktim
On Jun 25, 2014, at 18:29 , Marco Sieber wrote:
> I'm topposting because its my own i cant wait months for solutions, so
> because nobody had some crisp thoughts on that or even solutions
> So this is considered as a "push"
>
Your reply let's me think that you might mean somethin
On Aug 19, 2013, at 16:48 , Andrey Mochenov wrote:
> Hi,
>
> Thank you for your answer.
> I suppose, IDR stands for Instantaneous Decoding Refresh.
No, he means key frames, or i-frames, or however you want to call them, as
opposed to p-frames or b-frames.
> Please, explain: what do you mean
On Aug 8, 2013, at 21:18 , asqz wrote:
> ...
> It seems like I have to stop some buffer to be filled by the udp stream. I
> tried the function av_read_pause without success.
>
> Can you help me fix this problem please ?
>
How about closing the stream and reopen when you're ready to read again?
On Jul 30, 2013, at 22:18 , Vahid Kowsari wrote:
> Kalileo wrote:
>>
>> I think Alex means something like this, here as an example with 2 threads:
>>
>> AVDictionary *opts = NULL;
>> av_dict_set(&opts, "threads", "2", 0)
On Jul 30, 2013, at 21:26 , Adi Shavit wrote:
>>
>> You can also try to use multithreaded decoder. But this is not going
>> to help if you have more video streams in parallel than CPU cores.
>
> I can give it a shot.
> How do I set it up?
I think Alex means something like this, here as an examp
secret what you do. That
is why I asked you this:
On Jun 14, 2013, at 19:56 , Kalileo wrote:
> In order to make it possible to help you you would have to be much more
> specific about what you do, which settings and options you use, and what
> result you get, or want.
__
On May 11, 2013, at 22:14 , Mike Versteeg wrote:
> Is x264 capable of live video streaming, i.e. constant bitrate? I'm
> asking because I cannot get any acceptable quality from it when using
> bitrates < 1 Mbps, not even for static images. Only if I set qcompress
> to a very low value does the qu
On Jun 9, 2013, at 00:19 , Massimo Battistel wrote:
> hello,
> I would like to know if there are some code samples and/or tutorials for
> programmatically output streaming (mpeg-ts over udp) using
> libavformat/avcodec.
> I can't find anything on ffmpeg web site.
Nothing special needed, becaus
On Jun 6, 2013, at 19:26 , Andrey Utkin wrote:
> 2013/6/6 Kalileo :
>> Is there any way to set the PMT pid programmatically ?
>
> AVDictionary *opts = NULL;
> av_dict_set(&opts, "mpegts_pmt_start_pid", "0x2000" /* example value */, 0);
> /* m
The mpegts muxer has the options (in ffmpeg cli)
-mpegts_pmt_start_pid 0x1500
-mpegts_start_pid 0x150
-metadata service_provider="Some provider"
-metadata service_name="Some Channel"
If you use the mpegts muxer indirectly, through the -segment options to create
HLS, thes
On May 30, 2013, at 22:46 , Vahid Kowsari wrote:
> Hi All,
>
> I am working on a piece of code at the moment to receive HD H264 MPEG TS
> streams at around 7.5 mbps from a udp multicast.
> This code is running inside of ubuntu VM remotely.
>
> I am having problems because no matter what I do I
On May 27, 2013, at 22:05 , hamidi wrote:
> what do u mean by 'from source'. i'm a newbie in ubuntu and ffmpeg. do u need
> to install it by using apt-get install?
> about x264, i thought it to be a command line utility.
> after installing x264 by apt-get install, i've to add -lx264 to g++ comma
On May 27, 2013, at 21:30 , hamidi wrote:
> oh ic! so i forgot make. thx
> now, when i make my project i get the following errors:
>
> /hamidi/libav/libav-0.8.6/libavcodec/libx264.c:126: undefined reference to
> `x264_picture_init'
> /hamidi/libav/libav-0.8.6/libavcodec/libx264.c:128: undefined
On May 27, 2013, at 20:49 , hamidi wrote:
> thanx 4 the reply
> i've already tried the option. after downloading and extracting the source, i
> tried the following:
> ./configure --enable-libx264 --enable-gpl
> make install
> but the function still returns NULL.
> what can i do?
> thx
there sho
On May 26, 2013, at 00:31 , Brad O'Hearne wrote:
> On May 25, 2013, at 1:03 AM, Kalileo wrote:
>
>> As Robert said, please, please, please: "Please, take a step back and
>> reality-check your statements. You're so off-track here."
>
> …
> I ge
On May 25, 2013, at 14:36 , Kalileo wrote:
>
> On May 24, 2013, at 04:54 , Brad O'Hearne wrote:
>
>> If this is what you choose to exalt as "Top five...", I'm fine with being
>> considered "off-track". I don't want to share in any ideo
On May 24, 2013, at 04:54 , Brad O'Hearne wrote:
> If this is what you choose to exalt as "Top five...", I'm fine with being
> considered "off-track". I don't want to share in any ideology or number
> myself with those that considers this kind of behavior professional or
> acceptable. It is co
On May 23, 2013, at 04:22 , Brad O'Hearne wrote:
> On May 22, 2013, at 1:29 PM, Kalileo wrote:
>
>> Please don't complain if you don't find many willing to discuss these
>> theories with you, most here are looking for answers or trying to help
>>
On May 22, 2013, at 23:53 , Brad O'Hearne wrote:
> On May 22, 2013, at 9:02 AM, Robert Krüger wrote:
>
>> After all, most of the work is
>> done by people in their spare time and I haven't found many developers
>> who enjoy writing documentation (no matter how important docs are, I
>> think we
On May 21, 2013, at 22:53 , Brad O'Hearne wrote:
> On May 21, 2013, at 7:26 AM, Kalileo wrote:
>
>> Which "Playback" , which player ? Every player can have it's own playback
>> logic, nevertheless I don't see where the player logic would require fps
On May 21, 2013, at 20:35 , Brad O'Hearne wrote:
> On May 21, 2013, at 12:12 AM, Robert Krüger wrote:
>> are you talking about using the ffmpeg as a command line application?
>
> No, using libraries programmatically in an app.
>
>> you elaborate what exactly you mean by "but FFmpeg must have f
On May 21, 2013, at 20:40 , Brad O'Hearne wrote:
> On May 20, 2013, at 8:39 AM, Brad O'Hearne wrote:
>
>> I take it by sound of crickets (no response) to my question above that
>> either I've done a bad job communicating the issue, or it is indeed a real
>> stumper. In the event that it is th
On May 21, 2013, at 20:40 , Brad O'Hearne wrote:
> On May 20, 2013, at 8:39 AM, Brad O'Hearne wrote:
>
>> I take it by sound of crickets (no response) to my question above that
>> either I've done a bad job communicating the issue, or it is indeed a real
>> stumper. In the event that it is th
On May 17, 2013, at 17:47 , Pradeep Karosiya wrote:
> Ok after successively disabling the code I found that memory leak is coming
> from av_interleaved_write_frame(), even though I'm using av_free_packet, it
> is still there.
> I tried to use av_destruct_packet just before av_free_packet but then
On May 15, 2013, at 11:09 , Bjoern Drabeck wrote:
>>>
>>> When I am opening a file I call avformat_open_input, then after that I
>>> call avformat_find_stream_info, and I am timing those calls:
>>>
>>> 'avformat_open_input': 9.7970 seconds
>>> 'avformat_find_stream_info': 0.1080 secon
On May 15, 2013, at 09:36 , Bjoern Drabeck wrote:
>>
>> This is also what I noticed, when playing with probesize and analyzeduration
>> trying to get the player to start more quickly (using mpegts (h264/aac)
>> streams served through a memory buffer).
>>
>> This leads to the question why does
On May 12, 2013, at 22:43 , Carl Eugen Hoyos wrote:
> Kalileo writes:
>
>> This is also what I noticed, when playing with
>> probesize and analyzeduration trying to get the
>> player to start more quickly (using mpegts
>> (h264/aac) streams served through a me
On May 10, 2013, at 14:12 , Bjoern Drabeck wrote:
>>
> I just tested some more, it seems to make no real difference whether I
> set probesize/analyzeduration to 500 or 5000, both take more or
> less the same time (on the test file I was using today it was around
> 10 sec).. if I go much lower
On May 3, 2013, at 07:09 , Gonzalo Garramuno wrote:
> 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 i
On May 1, 2013, at 07:14 , Brad O'Hearne wrote:
> On Apr 30, 2013, at 3:09 PM, Kalileo wrote:
>
>> Brad, that the problem with what you spread here
>
> I spread *nothing*. Here's the exact statement from Paul Mahol, presumably
> one of the developers on the l
On May 1, 2013, at 03:27 , Carl Eugen Hoyos wrote:
> Brad O'Hearne writes:
>
>> As constructive feedback, I recommend removing aac entirely
>> from the codebase
PLEASE N
>
> The native aac encoder is better than most non-experimental
> audio encoders for stereo input, it is not goo
On Apr 22, 2013, at 06:32 , Brad O'Hearne wrote:
> On Apr 20, 2013, at 1:01 PM, Brad O'Hearne wrote:
>
>> Beyond this, I'm pretty much grabbing at straws. I'm suspecting that the
>> answer is probably just a knob or switch somewhere, a setting that will make
>> it work. I read a bit about bit
On Apr 1, 2013, at 11:57 , Brad O'Hearne wrote:
> On Mar 31, 2013, at 9:20 PM, Kalileo wrote:
>>> In my specific use case, I had configured a minimum frame rate of 24 fps on
>>> my QTDecompressedVideoCaptureOutput, and so expecting that frame rate, I
>&
On Apr 1, 2013, at 09:50 , Brad O'Hearne wrote:
> On Mar 31, 2013, at 6:32 PM, Kalileo wrote:
>
> Kalileo -- thanks for the reply. I'm not sure if you've read this thread and
> everything I've written, but based on the questions it appears you may have
&
On Apr 1, 2013, at 06:31 , Brad O'Hearne wrote:
> On Mar 31, 2013, at 1:25 AM, Alex Cohn wrote:
>> I am not sure when "duration" is taken into account, but you could
>> simply set current->pts = prev->pts+2. Note that this was my original
>> proposal.
>
> Alex -- I considered that approach, and
On Mar 30, 2013, at 04:49 , Brad O'Hearne wrote:
> On Mar 29, 2013, at 2:12 PM, Kalileo wrote:
>
> All of the below really helps my understanding...I think a few more things I
> need to know to fill in the gaps:
>>> To the best of what I've been able
on the codec and on the container.
>
> I beg to differ from Kalileo in one aspect: the codec may care about pts, but
> on the other hand it may not.
>
>
We do not differ here.
Codec as in decoder for video or audio, i don't see that, (except rearranging
the order of vide
On Mar 30, 2013, at 03:28 , Brad O'Hearne wrote:
> On Mar 28, 2013, at 11:53 PM, Kalileo wrote:
>> Hi Brad,
>>
>> when you start writing the packets (muxing them), you give each audio and
>> video packet a DTS (and PTS) value. You can start at zero.
>&g
On Mar 29, 2013, at 03:03 , Brad O'Hearne wrote:
> On Mar 27, 2013, at 11:48 PM, Kalileo wrote:
>
>> When you encode audio and video, you'll feed each packet with the dts and
>> pts value. The encoding function for video and the encoding function for
>> aud
values, and that
there is no magic communication between audio and video involved, it is quite
simple.
if you base your dts and pts values on the time when you received the data
after it went through various buffers, you have to take the delay caused by
these buffers into consideration.
Ht
also contains some
av_dict_set() examples which might help you) which I think asks the same
question as you do just in other words.
> From: Kalileo
> Subject: avformat_open_input using custom AVDictionary to set video_size
> Date: January 24, 2013 04:35:58 GMT+07:00
> To: "ffmp
Thanks to Carl-Eugen and Phil for explaining this in words easy to understand
for a non-lawyer like me :) - the first time I feel i know what to do to comply
with the license.
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On Feb 12, 2013, at 20:11 , Phil Turmel wrote:
> On 02/12/2013 06:07 AM, Kalileo wrote:
>>
>> On Feb 12, 2013, at 05:03 , Phil Turmel wrote:
>
> [trim /]
>
>>> If you zip up the source and include it in your CD or download
>>> every time you pu
On Feb 12, 2013, at 05:03 , Phil Turmel wrote:
> On 02/11/2013 04:48 PM, Video Monkey wrote:
>> I realize I'm seeing legal advice on a technical forum, however I'm
>> hoping there is an experienced hand willing to guide me. I'm reading
>> the "License Compliance Checklist" at http://ffmpeg.org/le
;m using the very latest ffmpeg 1.0.3 from 2 days ago.
My Questions are:
How do I pass the video size to avformat_open_input and
avformat_find_stream_info so that it is taken?
Which settings do i have to set to set (and how) to avoid the need for
avformat_find_stream_info?
How do i tell avformat_open_input that there is a video stream and an audio
stream, or help it to find them (possible a change in the encoding parameters)?
Thanks for any hints or help!
Regards,
Kalileo
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On Jan 22, 2013, at 11:37 , Taha Ansari wrote:
> Hi,
>
> I've been looking deeper into this, unfortunately so far I could not find a
> way to detect live stream's bit rate, or any way I could some how 'stuff'
> custom data (which would be bit rate in my case) with the encoder, and
> extract t
On Dec 29, 2012, at 18:54 , Funky Factory Development wrote:
> Hi,
>
> I'm having some issues with AAC decoding with recent version of ffmpeg:
>
> I'm using this code to decode AAC Frames (from a stream).
> I had some troubles with the decoding - everything runs fine but the after a
> few seco
On Jun 29, 2012, at 15:00 , Jonas Elofsson wrote:
> Hi,
>
> Thanks. I guess this means "pts"?
>
> I have no time to check again at the moment, but I did some tests with this
> as well.
>
> The video I tested with reported 25 frames per second and the timebase 1/25.
> Shouldn't this give me
On May 29, 2012, at 20:26 , Christian Brümmer wrote:
> "Cant be that hard - or? "
>
> Okay now im pretty sure it is:
>
> I read AVCodecContex.extradata contains the informations i need. For decoding
> i used that code ->
> http://cgit.lscube.org/cgit.cgi/feng/tree/src/media/parser/h264.c#n218
On May 25, 2012, at 04:35 , Michael Bradshaw wrote:
> On Thu, May 24, 2012 at 2:05 PM, Andrey Utkin
> wrote:
>>> Is there a particular reason FFmpeg streams so slowly?
>>
>> Coz while you decode, you don't fetch, and while you fetch, you don't
>> decode. That's what switching overhead and lack
On May 16, 2012, at 11:44 , Wagner Patriota wrote:
> I can't compile the last version of FFMPEG with libx264... it says:
> "ERROR: libx264 version must be >= 0.118."
>
> I have the latest version from the libx264 website...
>
You don't.
There are still fles from an old libx264 installation in
On May 7, 2012, at 02:28 , Bruce Wheaton wrote:
> Good question. The intent is to seamlessly move from ingesting a movie and
> viewing it to playing back. I can't find a way to access the same movie for
> playback as it's being ingested (that would work fine too), can't tell which
> media cont
On May 4, 2012, at 00:16 , jettoblack wrote:
>
> Kalileo wrote
>>
>> On Apr 30, 2012, at 23:53 , jettoblack wrote:
>>
>>> On Mon, Apr 30, 2012 at 2:51 AM, Kalileo [via libav-users] <
>>> ml-node+s943685n4597541h88@.nabble> wrote:
>>>
&
On Apr 30, 2012, at 23:53 , jettoblack wrote:
> On Mon, Apr 30, 2012 at 2:51 AM, Kalileo [via libav-users] <
> ml-node+s943685n4597541...@n4.nabble.com> wrote:
>
>>
>> The decoder gives you the (decoded) pictures in the correct order.
>>
>>
> That
The decoder gives you the (decoded) pictures in the correct order.
The PTS value _should_ reflect that order.
I would try to attach the metadata in the order as you get it from the demuxer
with the pictures as you get them out of the decoder.
Basically, it is exactly the same syncing you have
On Apr 26, 2012, at 05:53 , Григорий wrote:
>
> So I'm doing the following:
> - Create a buffer
> - Get a video RTMP packet from the queue
> - Read 3 fields at the beginning of it - FLV flags (1 byte), AVC packet type
> (1 byte), and CompositionTime (24-bit integer)
> - If AVC packet type is 0,
On Apr 13, 2012, at 08:16 , ashika umanga wrote:
> thanks all for the tips,
> I noticed that in the "muxing.c" example the "pts" value is only set to the
> AVFrame of video stream.Is it possible to do the same for Audio stream?
> I am sorry but I dont have clear idea on how to use "pts" .I know
On Apr 10, 2012, at 13:19 , srikanta mondal wrote:
> Thank you for reply.
> @kalileo I have simply got the UDP socket and get the byte array from it.
you could also use the ffmpeg libraries to do that.
> Now that byte array is basically the x264_nal_t*(p_payload[0]) from sender
&g
On Apr 10, 2012, at 11:31 , srikanta mondal wrote:
> @kalileo Actually I want to send the encoded frame over the network by UDP
Open the target udp url as output file and then use av_write_frame() to stream.
> and on the client side it will decode. And on the client side I want to
&g
On Apr 10, 2012, at 04:11 , Jeremy Graham wrote:
> I did try using avcodec_open2 as well but I still got a seg fault out of it.
> I set the third parameter to NULL, however. Is that what I am doing wrong?
No, NULL is ok there.
Try setting these 2 values before you open the codec:
c->codec_id
On Apr 9, 2012, at 19:02 , srikanta mondal wrote:
> Thank you for your reply.
> @kalileo I am doing the H.264 compression by opening a file with binary mode
> and write each encoded x264_nal_t* (p_payload[0]) frame into the file.
Then copy the picture - before you encode it - to th
On Apr 9, 2012, at 17:56 , srikanta mondal wrote:
> Dear all, Thank you for reply. I have used x264 library. And I have the
> following code
> [snip]
> This is my brief code sniff.Please guide me how could I able to decode
> directly without using file.
What do you mean with this: "decode
On Apr 9, 2012, at 16:13 , srikanta mondal wrote:
> Hi all,
> I am Srikanta Mondal from India. I am working on video compression and
> sending each compressed frame to network. I have done H.264 compression by
> using libx264 and the function x264_encoder_encode() and I have got the
> x264_n
On Apr 5, 2012, at 16:06 , Andrey Utkin wrote:
> 2012/4/5 huanggh0108 :
>> Of course.
>>
>> If the parametric filename like "udp://?localport=7899", it will receive
>> data from port 7899.
>
> Wrong. udp://127.0.0.1:7899 will do that. Maybe ip part can be
> eliminated, haven't try it.
> localpo
Nice project, thanks for open-sourcing it!
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On Apr 3, 2012, at 15:58 , Li Zhang wrote:
>
> Hi,
>
> Dose anyone has some suggestion for this problem?
>
> I hope I can deal with this problem not just let them popping up and stopping
> decode. In other words, it can resume the decoding again when the continuity
> check failed.
>
>
On Mar 30, 2012, at 21:28 , Andreas Polychronopoulos wrote:
> Hi all,
>
> does anyone know how can I use libav* to stream over rtp/udp frames that I
> have already encode using x264-api ?
> I don't want to use libav* to encode the frames only just to stream them ?
Yes, that is possible.
You
On Mar 29, 2012, at 14:10 , ashika umanga wrote:
> Greetings all,
>
> Is it possible to stream OSS instead of writing to a file?
> I was hoping something like this:
>
> avformat_alloc_output_context2(&oc, NULL, "ogg", "udp://localhost:");
have a look at avio_open() or avio_open2(), there y
On Mar 27, 2012, at 19:59 , Ajita Pandey wrote:
> [snip] but I don't think to put this question in the public mailing list have
> any harm
> The purpose of this mailing list is to share ur quires and sort out other
> quires .As it's a open source code.
Carl Eugen was talking about your disclai
On Mar 26, 2012, at 18:58 , Li Zhang wrote:
>
> Yes, I agree with what you said. But I do not know how to compute the dts
> exactly. I just let the muxer add that for me. But I do not know if muxer
> did that.
>
> I tried just now that starting the audio transcoding when the first video
> tr
o frames are used
to synchronize. You are not setting these values. For example, right before you
write a packet you even use:
output_video_packet.dts = AV_NOPTS_VALUE;
and
pkt.dts = AV_NOPTS_VALUE;
Therefore IMHO there is nothing left for any player
On Mar 21, 2012, at 14:38 , Alex Cohn wrote:
> On Tue, Mar 20, 2012 at 12:32, Li Zhang wrote:
>>
>> The output is stream, which is send to a udp address real time using my
>> customized output function. The encoder is H264 and AAC. Container is TS.
>>
>> Every time, the video frame writing w
On Mar 20, 2012, at 01:52 , Craig Murray wrote:
>
> I need to package the ffmpeg's .dll into another application and call it
> directly.
>
> From the command line, the command need to provide would be:
>
> ffmpeg -i test.avi -f image2 foo-%03d.jpeg
>
>
> But I'm not sure which .dll file I n
On Mar 19, 2012, at 20:00 , Carlo Santos wrote:
>
> Is it possible to make a streaming server that can read frames from a video
> and send it to a client over the network? If so, how is this done?
yes, it is possible.
Have you looked at any example reading frames and writing frames?
In princ
On Mar 9, 2012, at 22:20 , Li Zhang wrote:
>
> Hi everyone,
>
>
> I am a new user to ffmpeg API. I want to transcode mpeg2 video stream to
> H.264+AAC and then put them into TS container again.
>
> I have implemented transcoding from mpeg2 TS to AVI, MKV, MP4 local files.
> Now the problem
On Mar 8, 2012, at 14:05 , Carl Eugen Hoyos wrote:
First of all thanks for your answers and ideas, I learned a lot already while
following your suggestions.
> Kalileo writes:
>
>> In order to run the files though ffmpeg I had ffmpeg
>> converting the files, which works,
On Mar 8, 2012, at 02:50 , Carl Eugen Hoyos wrote:
> Kalileo writes:
>
>> I tested it now with
>>
>> ./configure --cc=clang
>> and with
>> ./configure --cc=clang --disable-optimizations --disable-asm
>> --disable-yasm --enable-shared
>
> Pl
On Mar 7, 2012, at 23:25 , Carl Eugen Hoyos wrote:
> Kalileo writes:
>
>> I downloaded new snapshots of ffmpeg, libx264 and lame
>
> Please first test ./configure --cc=clang && make.
I tested it now with
./configure --cc=clang --enable-shared
and with
./conf
On Mar 7, 2012, at 14:15 , Carl Eugen Hoyos wrote:
> Kalileo writes:
>
>> Works fine on windows (with any of these files), however on
>> OS X Lion audio is broken (with every of these files).
>
> (Complete, uncut console output when failing missing.)
>
> Pl
This is driving me nuts:
I'm decoding mpegts files, with h264 and either mp2, mp3, or aac audio. The
mpegts files are cut out from mpegts streams from several different sources,
and saved as file.
The decoding containsothing unusual, the standard procedure as in any example.
Works fine on windo
On Mar 2, 2012, at 17:54 , ALESSANDRO PAGANELLI wrote:
> Hi Kalileo,
>
> please consider that file as a "test" I used to understand how to properly
> rebuild a mp4 + h264 file, so there are several aspects that have to be fixed.
> One is surely the use of two packets,
On Mar 2, 2012, at 16:00 , ALESSANDRO PAGANELLI wrote:
> Andrey, I finally managed to solve the issue! Thanks a lot :)
>
> Here, I briefly report what I did (the complete code can be found in the
> attached source file).
Alessandro, in the source code which you attached you copy the frame wh
On Feb 23, 2012, at 18:37 , ALESSANDRO PAGANELLI wrote:
> The stream I'm using is coded with h264 without mpegts.
if you have as input an mp4 file and as output an h264 stream, with or without
mpegts, then AFAIK you need the h264_mp4toannexb filter.
On Feb 23, 2012, at 00:59 , ALESSANDRO PAGANELLI wrote:
> I'm working on a project (which uses Libav/Ffmpeg) in which I have to
> transmit an h264 encoded file (with Mp4 file format) over a simulated network
> (I'm using network simulator 3 for this), to "emulate" a real video streaming.
>
> N
On Feb 18, 2012, at 03:00 , Andrey Utkin wrote:
> 2012/2/17 Pavel Sokolov :
>> May be I need to add some extra header to the packets?
>
> You can do it by duplicating AVCodecContext.extradata.
> avcodec_copy_context() can do that for you.
When I read from a file (video.ts) in mpegts format, avc
On Feb 8, 2012, at 06:44 , Scott Brown wrote:
> My player is looking at PTS and DTS values, not just playing them as they
> come. If it's missing samples for a given time, it plays silence (for
> audio). I don't set any flags when I open the transport stream, although I
> did once try setting the
On Feb 8, 2012, at 03:58 , Scott Brown wrote:
> Hey everyone,
>
> I'm using ffmpeg libraries to read from a UDP transport stream, and I've been
> working on handling short term signal loss if ffmpeg can no longer read from
> the UDP stream. Basically the call to av_read_frame() takes a long ti
I have a mpegts file (h264/aac) which was created by dumping a mpegts udp
stream to a file. (the file can be played with vlc etc. perfectly).
Now I need to convert that file again to a stream (without decoding/encoding).
If I simply parse the file and remux the frames to an mpegts udp stream the
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