I want to receive images from an IP camera over HTTP via GET request. I have written a program that creates TCP socket connection with the camera and sends the following GET request to the camera:
GET /mjpeg?res=full HTTP/1.1\r\nHost: 143.205.116.14\r\n\r\n
After that, I receive images with the following function in a while loop:
while((tmpres = recv(sock,(void *) buf, SIZE, 0)) > 0 && check<10)
.....
where SIZE represents the size of the buffer. I, infact, don't know what size to define here. I am receiving a color image of size 2940x1920. So I define SIZE=2940x1920x3. After receiving the MJPEG image, I decode it with ffmpeg. But I observe that ffmpeg just partially/uncorrectly decodes the image and I just see a half (or even less) of the image. I assume it could be a size problem.
Any help in this regard would be highly appreciated.
Regards,
Khan
Why reinvent the wheel (for the umpteenth time)? Use a ready-made HTTP client library, such as libcurl.
For that matter, perhaps you can even just write your entire solution as a shell script using the curl command line program:
#!/bin/sh
curl -O "http://143.205.116.14/mjpeg?res=full" || echo "Error."
curl -o myfile.jpg "http://143.205.116.14/mjpeg?res=full" || echo "Error."
# ffmpeg ...
Save bytes received as binary file and analyze. May be an incomplete image (image can be encoded as progressive JPEG - is interlaced in fact - that means if you truncate the file you'll see horizontal lines.) or can be a ffmpeg decoding issue. Or something different. What is check < 10 condition ?
Related
I've made a C++ program that lives in gke and takes some videos as input using ffmpeg, then does something with that input using opengl(not relevant), then finally encodes those edited videos as a single output. Normally the program works perfectly fine on my local machine, it encodes just as I want it to with no warnings or valgrind errors whatsoever. Then, after encoding the said video, I want my program to upload that video to the google cloud storage. This is where the problem comes, I have tried 2 methods for this: First, I tried using curl to upload to the cloud using a signed url. Second, I tried mounting the google storage using gcsfuse(I was already mounting the bucket to access the inputs in question). Both of those methods yielded undefined, weird behaviour's ranging from: Outputing a 0byte or 44byte file, (This is the most common one:) encoding in the correct file size ~500mb but the video is 0 seconds long, outputing a 0.4 second video or just encoding the desired output normally (really rare).
From the logs I can't see anything unusual, everything seems to work fine and ffmpeg does not give any errors or warnings, so does valgrind. Everything seems to work normally, even when I use curl to upload the video to the cloud the output is perfectly fine when it first encodes it (before sending it with curl) but the video gets messed up when curl uploads it to the cloud.
I'm using the muxing.c example of ffmpeg to encode my video with the only difference being:
void video_encoder::fill_yuv_image(AVFrame *frame, struct SwsContext *sws_context) {
const int in_linesize[1] = { 4 * width };
//uint8_t* dest[4] = { rgb_data, NULL, NULL, NULL };
sws_context = sws_getContext(
width, height, AV_PIX_FMT_RGBA,
width, height, AV_PIX_FMT_YUV420P,
SWS_BICUBIC, 0, 0, 0);
sws_scale(sws_context, (const uint8_t * const *)&rgb_data, in_linesize, 0,
height, frame->data, frame->linesize);
}
rgb_data is the data I got after editing the inputs. Again, this works fine and I don't think there are any errors here.
I'm not sure where the error is and since the code is huge I can't provide a replicable example. I'm just looking for someone to point me to the right direction.
Running the cloud's output in mplayer wields this result (This is when the video is the right size but is 0 seconds long, the most common one.):
MPlayer 1.4 (Debian), built with gcc-11 (C) 2000-2019 MPlayer Team
do_connect: could not connect to socket
connect: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing /media/c36c2633-d4ee-4d37-825f-88ae54b86100.
libavformat version 58.76.100 (external)
libavformat file format detected.
[mov,mp4,m4a,3gp,3g2,mj2 # 0x7f2cba1168e0]moov atom not found
LAVF_header: av_open_input_stream() failed
libavformat file format detected.
[mov,mp4,m4a,3gp,3g2,mj2 # 0x7f2cba1168e0]moov atom not found
LAVF_header: av_open_input_stream() failed
RAWDV file format detected.
VIDEO: [DVSD] 720x480 24bpp 29.970 fps 0.0 kbps ( 0.0 kbyte/s)
X11 error: BadMatch (invalid parameter attributes)
Failed to open VDPAU backend libvdpau_nvidia.so: cannot open shared object file: No such file or directory
[vdpau] Error when calling vdp_device_create_x11: 1
==========================================================================
Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family
libavcodec version 58.134.100 (external)
[dvvideo # 0x7f2cb987a380]Requested frame threading with a custom get_buffer2() implementation which is not marked as thread safe. This is not supported anymore, make your callback thread-safe.
Selected video codec: [ffdv] vfm: ffmpeg (FFmpeg DV)
==========================================================================
Load subtitles in /media/
==========================================================================
Opening audio decoder: [libdv] Raw DV Audio Decoder
Unknown/missing audio format -> no sound
ADecoder init failed :(
Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
[dvaudio # 0x7f2cb987a380]Decoder requires channel count but channels not set
Could not open codec.
ADecoder init failed :(
ADecoder init failed :(
Cannot find codec for audio format 0x56444152.
Audio: no sound
Starting playback...
[dvvideo # 0x7f2cb987a380]could not find dv frame profile
Error while decoding frame!
[dvvideo # 0x7f2cb987a380]could not find dv frame profile
Error while decoding frame!
V: 0.0 2/ 2 ??% ??% ??,?% 0 0
Exiting... (End of file)
Edit: Since the code runs on a VM, I'm using xvfb-run ro start my application, but again even when using xvfb-run it works completely fine on when not encoding to the cloud.
Apparently, I'm assuming for security reasons, the google cloud storage does not allow us to do multiple continuous operations on a file, just a singular read/write operation. So I found a workaround by encoding my video to a local file inside the pod and then doing a copy operation to the cloud.
I am trying to send audio from a microphone input between a server and client using pyaudio, I only need voice quality sampled at a rate of 8000. Without compression it works fine and I am trying to add zlib compression to reduce the bandwidth.
In my server the stream_callback function is
def callback(in_data, frame_count, time_info, status):
for s in read_list[1:]:
s.send(zlib.compress(in_data))
return (None, pyaudio.paContinue)
In my client I am trying to decompress like this
try:
while True:
data = s.recv(CHUNK)
stream.write(zlib.decompress(data, zlib.MAX_WBITS | 16))
except KeyboardInterrupt:
pass
I have tried various parameters with zlib.MAX_WBITS but all return this error:
zlib.error: Error -3 while decompressing data: incorrect header check
Edit: I have also tried with no second parameter with zlib.decompress
Can someone suggest what I am doing wrong please, TIA
You don't need the second parameter of zlib.decompress at all. What you have in the question would look for a gzip stream instead of a zlib stream.
For compressing audio, you should use an audio compressor. Take a look at this answer.
Using wireshark, I could see the html page I was requesting (segment reconstruction). I was not able to use pyshark to do this task, so I turned around to scapy. Using scapy and sniffing wlan0, I am able to print request headers with this code:
from scapy.all import *
def http_header(packet):
http_packet=str(packet)
if http_packet.find('GET'):
return GET_print(packet)
def GET_print(packet1):
ret = packet1.sprintf("{Raw:%Raw.load%}\n")
return ret
sniff(iface='wlan0', prn=http_header, filter="tcp port 80")
Now, I wish to be able to reconstruct the full request to find images and print the html page requested.
What you are searching for is
IP Packet defragmentation
TCP Stream reassembly
see here
scapy
provides best effort ip.defragmentation via defragment([list_of_packets,]) but does not provide generic tcp stream reassembly. Anyway, here's a very basic TCPStreamReassembler that may work for your usecase but operates on the invalid assumption that a consecutive stream will be split into segments of the max segment size (mss). It will concat segments == mss until a segment < mss is found. it will then spit out a reassembled TCP packet with the full payload.
Note TCP Stream Reassembly is not trivial as you have to take care of Retransmissions, Ordering, ACKs, ...
tshark
according to this answer tshark has a command-line option equivalent to wiresharks "follow tcp stream" that takes a pcap and creates multiple output files for all the tcp sessions/"conversations"
since it looks like pyshark is only an interface to the tshark binary it should be pretty straight forward to implement that functionality if it is not already implemented.
With Scapy 2.4.3+, you can use
sniff([...], session=TCPSession)
to reconstruct the HTTP packets
I am currently working on a project for school and have ran into an issue with a large amount of data being sent via Telnet. If I send a message less than 10KB it is fine. However if I send a message that is above 10KB, I receive the following error "501 Syntax error - line too long" after a few minutes of it running.
Does anyone know of a better way to implement what I am trying to accomplish, that will preferably work with the send()? The data being sent is 5 pages (in Word) of an image in base64.
Thank you, any help is greatly appreciated.
Here is the code portions that I am currently using, which work, with small amounts of data.
char *MailContents = new char[20000000];
std::ifstream in("C:\\test.txt");
std::string MailData((std::istreambuf_iterator<char(in)),std::istreambuf_iterator<char>());
//The following streams in the data into MailData() from a .txt file.
memcpy(MailContents, MailData.c_str(), MailData.length()); //This takes the data and copies it to MailContents
strcat(MailContents, "\r\n");
send(Connection, MailContents, strlen(MailContents), 0); //The following line will take the data in MailContents and echo it to the Telnet data section to be sent.
send(Connection, ".\r\n", strlen(".\r\n"), 0); //This line terminates the data entry and sends it.
I'm working with IP camera, and I have got Jpeg frames and audio data (PCM) from camera.
Now, I want to create video file (both audio and video) under .avi or .mp4 format from above data.
I searched and I knew that ffmpeg library can do it. But I don't know how to using ffmpeg to do this.
Can you suggest me some sample code or the function of ffmpeg to do it?
If your objective is to write a c++ app to do this for you please disregard this answer, I'll just leave it here for future reference. If not, here's how you can do it in bash:
First, make sure your images are in a nice format, easy to handle by ffmpeg. You can copy the images to a different directory:
mkdir tmp
x=1; for i in *jpg; do counter=$(printf %03d $x); cp "$i" tmp/img"$counter".jpg; x=$(($x+1)); done
Copy your audio data to the tmp directory and encode the video. Let's say your camera took a picture every ten seconds:
cd tmp
ffmpeg -i audio.wav -f image2 -i img%03d.jpg -vcodec msmpeg4v2 -r 0.1 -intra out.avi
Where -r 0.1 indicates a framerate of 0.1 which is one frame every 10 seconds.
The possible issues here are:
Your audio/video might go slightly out of sync unless you calculate your desired framerate carefully in advance. You should be able to get the length of the audio (or video) using ffmpeg and some grep magic. Even so the sync might be an issue with longer clips.
if you have more than 999 images the %03d format will not be enough, make sure to change the 3 to the desired length of the index
The video will inherit its length from the longer of the streams, you can restrict it using the -t switch:
-t duration - Restrict the transcoded/captured video sequence to the duration specified in seconds. "hh:mm:ss[.xxx]" syntax is also supported.