Read raw Genicam H.264 data to avlib - h.264

I try to get familiar with libav in order to process a raw H.264 stream from a GenICam supporting camera.
I'd like to receive the raw data via the GenICam provided interfaces (API), and then forward that data into libav in order to produce a container file that then is streamed to a playing device like VLC or (later) to an own implemented display.
So far, I played around with the GenICam sample code, which transferres the raw H.264 data into a "sample.h264" file. This file, I have put through the command line tool ffmpeg, in order to produce an mp4 container file that I can open and watch in VLC
command: ffmpeg -i "sample.h264" -c:v copy -f mp4 "out.mp4"
Currently, I dig through examples and documentations for each H.264, ffmpeg, libav and video processing in general. I have to admit, as total beginner, it confuses me a lot.
I'm at the point where I think I have found the according libav functions that would help my undertaking:
I think, basically, I need the functions avcodec_send_packet() and avcodec_receive_packet() (since avcodec_decode_video2() is deprecated).
Before that, I set up an avCodedContext structure and open (or combine?!?) it with the H.264 codec (AV_CODEC_ID_H264).
So far, my code looks like this (omitting error checking and other stuff):
...
AVCodecContext* avCodecContext = nullptr;
AVCodec *avCodec = nullptr;
AVPacket *avPacket = av_packet_alloc();
AVFrame *avFrame = nullptr;
...
avCodec = avcodec_find_decoder(AV_CODEC_ID_H264);
avCodecContext = avcodec_alloc_context3(avCodec);
avcodec_open2 ( avCodecContext, avCodec, NULL );
av_init_packet(avPacket);
...
while(receivingRawDataFromCamera)
{
...
// receive raw data via GenICam
DSGetBufferInfo<void*>(hDS, sBuffer.BufferHandle, BUFFER_INFO_BASE, NULL, pPtr)
// libav action
avPacket->data =static_cast<uint8_t*>(pPtr);
avErr = avcodec_send_packet(avCodecContext, avPacket);
avFrame = av_frame_alloc();
avErr = avcodec_receive_frame( avCodecContext, avFrame);
// pack frame in container? (not implemented yet)
..
}
The result of the code above is, that both calls to send_packet() and receive_frame() return error codes (-22 and -11), which I'm not able to decrypt via av_strerror() (it only says, these are error codes 22 and 11).
Edit: Maybe as an additional information for those who wonder if
avPacket->data = static_cast<uint8_t*>(pPtr);
is a valid operation...
After the very first call to this operation, the content of avPacket->data is
{0x0, 0x0, 0x0, 0x1, 0x67, 0x64, 0x0, 0x28, 0xad, 0x84, 0x5,
0x45, 0x62, 0xb8, 0xac, 0x54, 0x74, 0x20, 0x2a, 0x2b, 0x15, 0xc5,
0x62}
which somehow looks as something to be expected becaus of the NAL marker and number in the beginning?
I don't know, since I'm really a total beginner....
The question now is, am I on the right path? What is missing and what do the codes 22 and 11 mean?
The next question would be, what to do afterwards, in order to get a container that I can stream (realtime) to a player?
Thanks in advance,
Maik

At least for the initally asked question I found the solution for myself:
In order to get rid of the errors on calling the functions
avcodec_send_packet(avCodecContext, avPacket);
...
avcodec_receive_frame( avCodecContext, avFrame);
I had to manually fill some parameters of 'avCodecContext' and 'avPacket':
avCodecContext->bit_rate = 8000000;
avCodecContext->width = 1920;
avCodecContext->height = 1080;
avCodecContext->time_base.num = 1;
avCodecContext->time_base.den = 25;
...
avPacket->data = static_cast<uint8_t*>(pPtr);
avPacket->size = datasize;
avPacket->pts = frameid;
whereas 'datasize' and 'frameid' are received via GenICam, and may not be the appropriate parameters for the fields, but at least I do not get any errors anymore.
Since this answers my initial question on how I get the raw data into the structures of libav, I think, the question is answered.
The discussion and suggestions with/from Vencat in the commenst section lead to additional questions I have, but which should be discussed in a new question, I guess.

Related

Puzzling MediaCodec.BufferInfo.size values

I use MediaCodec to decode H.264/H.265 video streams. I don't think the code of thousands of lines is relevant to my question, so please allow me to avoid pasting the code here. Let me emphasize that the decoding works flawlessly. I am asking this question primarily out of curiosity, not for solving a problem.
MediaCodec.BufferInfo bi = new MediaCodec.BufferInfo();
...
int iOutputBufferIndex = myMediaCodec.dequeueOutputBuffer(bi, TIMEOUT_USEC);
logd("debug", "Output buffer size: " + bi.size);
The official document says that MediaCodec.BufferInfo.size is "The amount of data (in bytes) in the buffer." I have tested three video streams decoded by the code. MediaCodec.BufferInfo.size is 1 for an H.265 video and 8 for two H.264 video streams. These numbers do not look like "The amount of data (in bytes) in the buffer.". My understanding is that the buffer has decoded video frame.
Could anyone shed some light on this (i.e. the exact meaning of MediaCodec.BufferInfo.size) ?

How to edit a binary at specific address locations using Nim?

I'm trying to understand how to modify a Windows executable using Nim to write NOPs in specific locations of the file. In Python I do this as follows:
with open('file.exe', 'r+b') as f:
f.seek(0x0014bc0f)
f.write(binascii.unhexlify('9090'))
f.seek(0x0014bc18)
f.write(binascii.unhexlify('9090'))
But in Nim I can't find the necessary methods to accomplish the seeking to a specific address and how to input these op-codes properly. I'm pretty new to Nim and the different file operation possibilities are a little confusing.
the docs for io are quite good, but to start you off:
opening a file for reading and writing (equivalent of 'r+') is:
let f = open("file.exe",fmReadWriteExisting)
the equivalent of with here, to make sure to close the file on any error:
defer: f.close
seeking to an offset (by default bytes relative to the start):
f.setFilePos(0x014bc0f)
writing two bytes at that position (nop nop)
f.write("\x90\x90")
check out the documentation for each proc for more options

Use libpcap to capture multiple interfaces to same file

I would like to use libpcap to capture on multiple specific interfaces (not 'any') to the same file
I have the following code (error handling and some args removed):
static gpointer pkt_tracing_thread(gpointer data)
{
while (1)
{
pcap_dispatch(g_capture_device1, .., dump_file1);
pcap_dispatch(g_capture_device2, .., dump_file2);
}
}
fp1 = calloc(1, sizeof(struct bpf_program));
fp2 = calloc(1, sizeof(struct bpf_program));
cap_dev1 = pcap_open_live(interface1,...
cap_dev2 = pcap_open_live(interface2,...
pcap_compile(cap_dev1, fp1, ...
pcap_compile(cap_dev2, fp2, ...
pcap_setfilter(cap_dev1, fp1);
pcap_setfilter(cap_dev2, fp2);
dump_file1 = pcap_dump_open(g_capture_device1, filename);
dump_file2 = pcap_dump_open(g_capture_device2, filename);
g_thread_create_full(pkt_tracing_thread, (gpointer)fp1, ...
g_thread_create_full(pkt_tracing_thread, (gpointer)fp2, ...
This does not work. What I see in filename is just packets on one of the interfaces. I'm guessing there could be threading issues in the above code.
I've read https://seclists.org/tcpdump/2012/q2/18 but I'm still not clear.
I've read that libpcap does not support writing in pcapng format, which would be required for the above to work, although I'm not clear about why.
Is there any way to capture multiple interfaces and write them to the same file?
Is there any way to capture multiple interfaces and write them to the same file?
Yes, but 1) you have to open the output file only once, with one call to pcap_dump_open() (otherwise, as with your program, you may have two threads writing to the same file independently and stepping on each other) and 2) you would need to have some form of mutex to prevent both threads from writing to the file at the same time.
Also, you should have one thread reading only from one capture device and the other thread reading from the other capture device, rather than having both threads reading from both devices.
As user9065877, you have to open the output file only once and write to it only from one thread at a time.
However, since you'd be serializing everything anyway, you may prefer to ask libpcap for pollable file descriptors for the interfaces and poll in a round-robin fashion for packets, using a single thread and no mutexes.

How to successfully parse the output of FFMpeg in NodeJS

So I have seen a lot of topics on FFMPeg and it's a great tool I learnt about today, but I have spent the day perfecting the command and now am a little stuck with the NodeJS part.
In essence the command does the following: take input from a Mac OSX webcam, and then stream it to a web-socket. Now I looked at a lot of the NodeJS libraries but I couldn't find one that did what I need; or did not understand how to. Here is an example of the command that I am using:
ffmpeg -f avfoundation -framerate 30 -video_size 640x480 -pix_fmt uyvy422 -i "0:1" -f mpegts -codec:v mpeg1video -s 640x480 -b:v 1000k -bf 0 http://localhost:8081/stream
This does everything I need for the streaming side of things, but I wish to call it via NodeJS, and then be able to monitor the log, and parse the data that comes back for example:
frame= 4852 fps= 30 q=6.8 size= 30506kB time=00:02:41.74 bitrate=1545.1kbits/s speed= 1x \r
and use it to get a JSON array back for me to output to a webpage.
Now all I am doing is working on ways of actually parsing the data, and I have looked at lots of other answers for things like this, but I can't seem to split/replace/regex it. I can't get anything but a long string from it.
Here is the code I am using (NodeJS):
var ffmpeg = require('child_process').spawn('/usr/local/Cellar/ffmpeg/3.3.1/bin/ffmpeg', ['-f', 'avfoundation', '-framerate', '30', '-video_size', '640x480', '-pix_fmt', 'uyvy422', '-i', '0:1', '-f', 'mpegts', '-codec:v', 'mpeg1video', '-s', '640x480', '-b:v', '1000k', '-bf', '0', 'http://localhost:8081/test']);
ffmpeg.on('error', function (err) {
console.log(err);
});
ffmpeg.on('close', function (code) {
console.log('ffmpeg exited with code ' + code);
});
ffmpeg.stderr.on('data', function (data) {
// console.log('stderr: ' + data);
var tData = data.toString('utf8');
// var a = tData.split('[\\s\\xA0]+');
var a = tData.split('\n');
console.log(a);
});
ffmpeg.stdout.on('data', function (data) {
var frame = new Buffer(data).toString('base64');
// console.log(frame);
});
I have tried splitting with new lines, carridge return, spaces, tabs, but I just can't seem to get a basic array of bits, that I can work with.
Another thing to note, is you will notice the log comes back via stderr, I have seen this online and apparently it does it for a lot of people? So I am not sure what the deal is with that? but the code is is the sdterr callback.
Any help is very appreciated as I am truly confused on what I am doing wrong.
Thanks.
An update on this, I worked with one of the guys off the IRC channel: #ffmpeg on FreeNode. The answer was to send the output via pipe to stdout.
For example I appended the following to the FFMpeg command:
-progress pipe:1
The progress flag is used to give an output every second with information about the stream, so this is pretty much everything you get from the stderr stream every second, but piped to the stdout stream in a format that I can parse. Below is taken from the documentation.
-progress url (global) Send program-friendly progress information to url. Progress information is written approximately every second and at the end of the encoding process. It is made of "key=value" lines. key consists of only alphanumeric characters. The last key of a sequence of progress information is always "progress".
Here is an example of the code I used to parse the stream information:
ffmpeg.stdout.on('data', function (data) {
var tLines = data.toString().split('\n');
var progress = {};
for (var i = 0; i < tLines.length; i++) {
var item = tLines[i].split('=');
if (typeof item[0] != 'undefined' && typeof item[1] != 'undefined') {
progress[item[0]] = item[1];
}
}
// The 'progress' variable contains a key value array of the data
console.log(progress);
});
Thanks to all that commented!
In the spirit of not reinventing the wheel, you might want to try using fluent-ffmpeg. It dispatches a progress event with a number of useful fields
'progress': transcoding progress information
The progress event is emitted every time ffmpeg reports progress
information. It is emitted with an object argument with the following
keys:
frames: total processed frame count
currentFps: framerate at which FFmpeg is currently processing
currentKbps: throughput at which FFmpeg is currently processing
targetSize: current size of the target file in kilobytes
timemark: the timestamp of the current frame in seconds
percent: an estimation of the progress percentage
If you're curious about how they do this, you can read the source, starting from here and here
Ffmpeg uses stderr to output log info because stdout is used for piping the output to other processes. The stuff in stderr is actually just debug information, and not the actual output of the process.
BONUS ROUND
I've seen some hacky video players that use websockets to stream videos, but that approach has a number of issues with it. I'm not going to go over those, but I will explain why I think you should use hls.js.
Support is pretty good; basically works everywhere except old IE. It uses MSE to upgrade the standard video element, so you don't have to wrestle with building a custom player.
Here are the docs for the hls format flag
Here's some code that I'm using to stream from an IPTV box to a web page.
this.ffmpeg = new FFFmpeg()
this.ffmpeg.input(request(this.http_stream))
.videoCodec('copy')
.audioCodec('copy')
.outputOptions([
'-f hls',
'-hls_list_size 6',
'-hls_flags delete_segments'
])
.output( path.join(this.out_dir, 'video.m3u8') )
.run()
It generates a .m3u8 manifest file along with segmented mpeg-ts video files. All you need to do after that is load the m3u8 file into the hls.js player and you have a live stream!
If you're going to re-encode the stream, you will probably see some low fps and glitchiness. I'm lucky since my source stream is already encoded as mpeg-ts.

How can I avoid error #2030: End of file was encountered. - AS3

I am trying to make this game in Flash / AS3 which goes on as long as the player doesn't lose. The longer the player goes for, the higher his/her score is.
I also need to save the high score, so I decided to save this using a text file. I open the file with mode UPDATE so that if the file doesn't exist it is created and I can read/write. But obviously, when the file is first created it is empty so when I try to read the data in the file it returns error #2030. It then doesn't execute any of the code that involves the file stream. Also even if there is a number in the file, I can't read it because the program doesn't know how many digits the high score is going to be. This is the code that I am using:
var file:File = File.documentsDirectory;
file = file.resolvePath("hscore.txt");
var fileStream:FileStream = new FileStream();
fileStream.openAsync(file, FileMode.UPDATE);
var buffer:Number = fileStream.readInt();
if (buffer > 0){
if (score > buffer){
buffer = score;
fileStream.writeInt(buffer);
}
else{
highScore.text = String(buffer);
}
}
else{
fileStream.writeInt(score);
highScore.text = String(score);
}
fileStream.close();
I have tried using readUTF/writeUTF and readUTFBytes/writeUTFBytes instead of readInt/writeInt, but with readUTF/writeUTF it just gives me the same error and with readUTFBytes/writeUTFBytes the program would need to know how many digits the high score is. If anyone knows how to get around a 2030 error or knows how to read a file without reaching the end or knows how to see how many digits there are in the file please let me know.
I could have added a line to the file telling the program how many digits there are but when the file is new the file stream would reach the end of the file while trying to read the number of digits. I am also open to other methods of storing the high score and reading it later. Thanks in advance for any help.
P.S. I am trying to do this with Adobe AIR for Android
Answering my own question based on help from comments:
Open and close the file without making any changes so that if file doesn't exist the file is created.
Check the file size, 1 byte = 1 digit (stored in UTF-8).
Read appropriate number of bytes, make changes as needed, etc.
]
Error 2030 avoided! :D
Thanks to #Vesper and #Craig
*****Edit*****
Realised you don't need to close the file in the first step...
*****Edit2*****
Not UTF-8, ASCII