I am trying to process a .raw image file captured using vrl2, it's a h264 encoded image with yuv422 color space from a Logitech c920 webcam, dcraw is not working for me however from my previous question this command is working fine with low performance (a 32kb jpg image however using opencv capture I get a 900kb image for the same 640x480 resolution):
ffmpeg -f rawvideo -s 640x480 -pix_fmt yuyv422 -i frame-1.raw
frame-1.jpg
I need a code written in C or the ffmpeg api/OpenCV etc .. to do the same as this command,I don't want to use QProcess in Qt(I am working on a server using Qt where I am trying to send the raw file from a Raspberry PI to the server and process it their), dcraw output is a corrupted image.
http://ffmpeg.org/doxygen/trunk/examples.html
There should be some api samples in there that show how to get the image out with that specific encoding.
When interacting with a RAW file, I have also used IrfanView. If you know the headersize of the file and the width and the height and the bits per pixel per color, you can see what it looks like quickly that way.
EDIT: I tried using Irfanview with your RAW, and I got something close, but not quite. The coloring was always off. I don't think it can handle that particular encoding of a RAW file right now.
Related
Most gif capture software capture screen and then save them one by one single frame picture file on disk,then read them into memory and combine them to gif,makes the whole procdure very slowly.
I got a idea to capture screen with DirectX(so we could also capture directx window faster since it direct operate the screen d3d device)API to got the bitmap，then save them to memory(such as buffer),then passing the memory location to ffmpeg to produce a video so we don't need disk storge as a middle buffer so it could be ten more faster since the disk is now most slowly part on pc now.
the directx capture screen part is already.But I found that ffmpeg using OpenFile to read the picture file,so here may we can simulate the OpenFile?
If answer is yes,how could we do it?
You can open a named pipe and use that as a source.
An example:
ffmpeg -f rawvideo -vcodec ravideo -s $width$x$height$ -r $framerate -pix_fmt $pixelFormat$ -i "\\.\pipe\$pipeName$" Output.gif
You have to fix the format of the frames you are going to feed FFmpeg, hence the -s and the -pix_fmt parameters.
when i open a .m4a or .mp3 file using chrome (or in VLC "open new stream), and skip to a certain time mark, it starts streaming the audio right at that time mark, which means that it downloads a chunk from the audio file starting at that position.
when i check network under chrome dev tools after i click somewhere in the audio player, and copy the cURL request, it contains a range http header, for example:
-H "Range: bytes=26500000-30000000"
when i run that cURL command in the terminal, the output is a corrupt audio file. if i try to convert it using ffmpeg, it logs:
[mov,mp4,m4a,3gp,3g2,mj2 # 00000164d5a3ab40] moov atom not found
the only exception is when i request a byte range starting from 0 (and ending at 300000+) in which case the output file can play.
from what i understand, the headers of the audio file are not downloaded when the byte range is somewhere in the middle or ending of the file.
i tried specifying two ranges, as follows
-H "Range: bytes=0-300000,2000000-4000000"
but the output file is corrupt nevertheless.
how does chrome/vlc deal with this situation, and how can i replicate it? what is the right way to download an audio chunk from a large audio file? i am guessing i would have to make consecutive http requests to build a proper, non-corrupt file, it seems that it is what chrome does.
UPDATE
a "temporary fix" i found to the problem that you might find useful is to use launch VLC through command with a hidden interface, load the audio stream (in this case the audio file url) and transcode the audio to a local file while assigning a start and stop time. an example, with a random podcast:
vlc -Idummy --play-and-exit --sout "#transcode{acodec=s16l,channels=2,samplerate=44100}:std{access=file,mux=wav,dst=C:\audio\output.wav}" --start-time=600 --stop-time=630 "http://traffic.libsyn.com/preview/worldofhardwarestartups/World_of_Hardware_Startups_Podcast_EP01_-_AIR_Ready.mp3"
however, the original issue is still unsolved.
I'm using ESCAPI to capture webcam, it captures the frames in form of RGB pixel data, I've stored the RGB pixel data into a file but the file is huge 200MB for 15s video of 320x240.
I want to encode that pixel data into a video format.
I'm using MinGW on windows.
First use any encoder
I suggest H264 codec for encoding
so find its library for encoding and encode it
Then find any container File formats
I suggest Matroska (.mkv file)
so find its library for muxing encoded h264 in .mkv
Good begining is start with ffmpeg libraries.
Is it possible to dump a raw RTSP stream to file and then later decode the file to something playable?
Currently I'm using FFmpeg to receive and decode the stream, saving it to an mp4 file. This works perfectly, but is CPU intensive, and will severely limit the number of RTSP streams I can receive simultaneously on my server.
I would like to save the stream to file without decoding it, and delay the decoding part to when the file needs to be opened.
Is this possible?
I have tried VLC, which is even more CPU intensive than FFmpeg. I've also looked at this question where the answer says dumping RTSP to file is not useful, and this question, where the comment below the question says "Raw RTSP content is not well suited for save and replay...", which seems to indicate that there is way.
EDIT
Here is the command I'm using for FFmpeg:
ffmpeg -i rtsp://#192.168.241.1:62159 -r 15 C:/DB_Videos/2013-04-30 17_18_34.703.mp4
If you are reencoding in your ffmpeg command line, that may be the reason why it is CPU intensive. You need to simply copy the streams to the single container. Since I do not have your command line I cannot suggest a specific improvement here. Your acodec and vcodec should be set to copy is all I can say.
EDIT: On seeing your command line and given you have already tried it, this is for the benefit of others who come across the same question. The command:
ffmpeg -i rtsp://#192.168.241.1:62156 -acodec copy -vcodec copy c:/abc.mp4
will not do transcoding and dump the file for you in an mp4. Of course this is assuming the streamed contents are compatible with an mp4 (which in all probability they are).
With this command I had poor image quality
ffmpeg -i rtsp://192.168.XXX.XXX:554/live.sdp -vcodec copy -acodec copy -f mp4 -y MyVideoFFmpeg.mp4
With this, almost without delay, I got good image quality.
ffmpeg -i rtsp://192.168.XXX.XXX:554/live.sdp -b 900k -vcodec copy -r 60 -y MyVdeoFFmpeg.avi
You can use mplayer.
mencoder -nocache -rtsp-stream-over-tcp rtsp://192.168.XXX.XXX/test.sdp -oac copy -ovc copy -o test.avi
The "copy" codec is just a dumb copy of the stream. Mencoder adds a header and stuff you probably want.
In the mplayer source file "stream/stream_rtsp.c" is a prebuffer_size setting of 640k and no option to change the size other then recompile. The result is that writing the stream is always delayed, which can be annoying for things like cameras, but besides this, you get an output file, and can play it back most places without a problem.
I am using espeak on BSD to output text as sound. My problem is that I want it to take it as a .mp3 but I am having little luck. I tried piping the output to tee but I guess that only works with stdout not just playing a sound.
Any ideas? My last resort would be recompiling my own version of espeak that allows me to save to a file instead of playing it
you can write it as wave and then convert it with ffmpeg:
espeak "HelloWorld" -w <file>.wav
Or pipe to ffmpeg
espeak "HelloWorld" --stdout | ffmpeg -i pipe:0 output.mp3
From the documentation:
-w < wave file>
Writes the speech output to a file in WAV format, rather than speaking it.
--stdout
Writes the speech output to stdout as it is produced, rather than speaking it. The data starts with a WAV file header which indicates the sample rate and format of the data. The length field is set to zero because the length of the data is unknown when the header is produced.
It looks like both of those options produce WAV files, but you can easily convert those without another program like ffmpeg.