I have a video stream coming in on rtp to ffmpeg and I want to pipe this to my OpenCV tools for live streaming processing. The rtp linkage is working because I am able to send the incoming data to a file and play it (or play if via ffplay). My OpenCV implementation is functional as well because I am able to capture video from a file and also a webcam.
The problem is the streaming to OpenCV. I have heard that this may be done using a named pipe. First I could stream the ffmpeg output to the pipe and then have OpenCV open this pipe and begin processing.
What I've tried:
I make a named-pipe in my cygwin bash by:
$ mkfifo stream_pipe
Next I use my ffmpeg command to pull the stream from rtp and send it to the pipe:
$ ffmpeg -f avi -i rtp://xxx.xxx.xxx.xxx:1234 -f avi -y out.avi > stream_pipe
I am not sure if this is the right way to go about sending the stream to the named pipe but it seems to be accepting the command and work because of the output from ffmpeg gives me bitrates, fps, and such.
Next I use the named pipe in my OpenCV capture function:
$ ./cvcap.exe stream_pipe
where the code for cvcap.cpp boils down to this:
cv::VideoCapture *pIns = new cv::VideoCapture(argv[1]);
The program seems to hang when reaching this one line, so, I am wondering if this is the right way of going about this. I have never used named pipes before and I am not sure if this is the correct usage. In addition, I don't know if I need to handle the named pipe differently in OpenCV--change code around to accept this kind of input. Like I said, my code already accepts files and camera inputs, I am just hung up on a stream coming in. I have only heard that named pipes can be used for OpenCV--I haven't seen any actual code or commands!
Any help or insights are appreciated!
UPDATE :
I believe named pipes may not be working in the way I intended. As seen on this cygwin forum post:
The problem is that Cygwin's implementation of fifos is very buggy. I wouldn't recommend using fifos for anything but the simplest of applications.
I may need to find another way to do this. I have tried to pipe the ffmpeg output into a normal file and then have OpenCV read it at the same time. This works to some extent, but I imagine in can be dangerous to read and write from a file concurrently--who knows what would happen!
hope it's not too late to answer, but I have tried the same thing some time ago, and here is how I did it.
The video-decoding backend for OpenCV is actually ffmpeg, so all its facitilites are available in OpenCV as well. Not all the interface is exposed, and that adds some difficulties, but you can send the rtp stream address to OpenCV.
cap.open("rtp://xxx.xxx.xxx.xxx:1234");
Important: OpenCV is not able to access password-protected rtp streams. To do that, you would need to provide the username and the password, there is no API exposed for it.
Related
TL;DR
I want to convert fMP4 fragments to TS segments (for HLS) as the fragments are being written using FFmpeg on an iOS device.
Why?
I'm trying to achieve live uploading on iOS while maintaining a seamless, HD copy locally.
What I've tried
Rolling AVAssetWriters where each writes for 8 seconds, then concatenating the MP4s together via FFmpeg.
What went wrong - There are blips in the audio and video at times. I've identified 3 reasons for this.
1) Priming frames for audio written by the AAC encoder creating gaps.
2) Since video frames are 33.33ms long, and audio frames 0.022ms long, it's possible for them to not line up at the end of a file.
3) The lack of frame accurate encoding present on Mac OS, but not available for iOS Details Here
FFmpeg muxing a large video only MP4 file with raw audio into TS segments. The work was based on the Kickflip SDK
What Went Wrong - Every once in a while an audio only file would get uploaded, with no video whatsoever. Never able to reproduce it in-house, but it was pretty upsetting to our users when they didn't record what they thought they did. There were also issues with accurate seeking on the final segments, almost like the TS segments were incorrectly time stamped.
What I'm thinking now
Apple was pushing fMP4 at WWDC this year (2016) and I hadn't looked into it much at all before that. Since an fMP4 file can be read, and played while it's being written, I thought that it would be possible for FFmpeg to transcode the file as it's being written as well, as long as we hold off sending the bytes to FFmpeg until each fragment within the file is finished.
However, I'm not familiar enough with the FFmpeg C API, I only used it briefly within attempt #2.
What I need from you
Is this a feasible solution? Is anybody familiar enough with fMP4 to know if I can actually accomplish this?
How will I know that AVFoundation has finished writing a fragment within the file so that I can pipe it into FFmpeg?
How can I take data from a file on disk, chunk at a time, pass it into FFmpeg and have it spit out TS segments?
Strictly speaking you don't need to transcode the fmp4 if it contains h264+aac, you just need to repackage the sample data as TS. (using ffmpeg -codec copy or gpac)
Wrt. alignment (1.2) I suppose this all depends on your encoder settings (frame rate, sample rate and GOP size). It is certainly possible to make sure that audio and video align exactly at fragment boundaries (see for example: this table). If you're targeting iOS, I would recommend using HLS protocol version 3 (or 4) allowing timing to be represented more accurately. This also allows you to stream audio and video separately (non-multiplexed).
I believe ffmpeg should be capable of pushing a live fmp4 stream (ie. using a long-running HTTP POST), but playout requires origin software to do something meaningful with it (ie. stream to HLS).
Context
Most RTP streams (from e.g. an IP camera) need some information from a SDP to be able to decode them.
SDP is usually fetched just in time, usually from a RTSP URL but other means are possible (e.g. HTTP).
Specific case
We have a situation where an RTP stream (from a camera, UDP sent at all time whether anyone listens or not) will be played using VLC, but providing VLC an RTSP URL to fetch SDP just in time is not an option.
There is a RTSP service yet we need to query it in advance and dump the resulting SDP file to feed it to VLC later. Doing a RTSP query just-in-time is useless anyway since the stream exists at all times.
How to do that with VLC?
Search before you post
Of course I've been searching Google, videolan wiki and StackExchange.
Information is difficult to find because when people talk about streaming, RTSP, RTP, they are generally usig VLC to generate a RTP stream, or output a SDP that VLC generates because it does the encoding, etc.
It's not the case here. The SDP to dump comes from the serveur with a single RTSP query.
Question
Basically, I'm looking for a command-line like:
vlc --sout...something...rtsp://sourceIP:Port/...something...out...myfile.sdp
That would dump the SDP in myfile.sdp.
Then, later, running vlc with the myfile.sdp as argument is expected to play the stream.
We did not find a solution using VLC alone (I even looked a little at the VLC source code). So we used a somehow "brute force" solution but hey, it works.
What we do at configure time is ask VLC to play stream once, while Wireshark captures packets with filter rtsp and sdp. One packet appears containing the SDP data we want. We select it and use "extract selected bytes to ..." and save to a file with name ending with .sdp.
That gives us a file containing the SDP information we want. Job done.
I develop an iOS application where i should call web service and return audio file ( as byte array ) then play it to user .
I have problem with audio file format as it has 8 Kbps bit rate and no player inside app can play it. when I convert it to any other bit rate for example (13 Kbps) from server side to test it works properly . However i have a huge number of file where converting them manually is impossible . is there any way to convert file inside iOS app code ?
To address your comment about converting the files manually: Is it a plausible solution for you to do it automatically, server side? If so, you probably have a few options.
One would be to install Audacity, which handles all of these bit rates, and use the Chains feature for batch processing. If it's a one-off conversion you could initiate it manually, otherwise you may need to find a way of scripting the process (if new files come in from an external source, that is).
As for playing these files in iOS, have you considered embedding libpd (just one example of many, but one that you can get up and running in minutes)? It has a fairly open approach to file playback, and may handle these file formats. If you send me an example I can test right away!
I'm trying to teach myself a bit about video streaming and transcoding, with some Roku app development on the side. I have a number of video files (mostly in FLV format (H.264/AAC)) that I would like to stream to a client, which in this case is a Roku box (that accepts MP4 (H.264/AAC) and HTTP Live Streaming (HLS)). I'm wondering if it is possible to transcode/remux the FLV files and stream them to the client on the fly, perhaps over HLS?
I have tried using ffmpeg to remux the files and serve them immediately during the transcoding process, but they are unplayable until the write process is complete. I can get the Roku to play my completed MP4 files just fine via Apache/Rails.
But I'm wondering... is it possible to set up a server to transcode/remux a file and immediately have the output file (from ffmpeg/whatever tool I'm using) streamed to the client? If so, what tools are required to accomplish this? Is it possible to use a media file segmenter to chop up a file as it's being transcoded or remuxed?
I'm well aware that the transcoding process is CPU intensive, but I'm not so much worried about the practicality of transcoding and streaming on the fly since this is simply a personal education project (and I have an idle system that is capable if handling this).
Apologies if I'm way off base here, just trying to hack my way through this.
Thanks!
The trick to getting HLS served immediately that a TS segment has been completed is getting the playlist to dynamically update as the data arrives on disk.
What you are trying to do is essentially stream a Live event over HLS, which absolutely can be done, it just takes co-ordination between tools.
The opensource segmenter is able to do this, the trick is to have ffmpeg write out a single MPEG-TS stream (Unsegmented) and write this to a named pipe (Or equivalent for your OS), then have segmenter read from this named pipe and write the files to a directory within your shared webspace.
The segmenter repeatedly updates the M3U8 file on disk while processing so it can be used as a "Live" stream until the task is finished.
When ffmpeg closes its output the segmenter puts the end tag in the M3U8 and the file becomes "VOD".
The segmenter can be downloaded here
As of Flash 10.1, they have added the ability to add bytes into the NetStream object via the appendBytes method (described here http://www.bytearray.org/?p=1689). The main reason for this addition is that Adobe is finally supporting HTTP streaming of video. This is great, but it seems that you need to use the Adobe Media Streaming Server (http://www.adobe.com/products/httpdynamicstreaming/) to create the correct video chunks from your existing video to allow for smooth streaming.
I have tried to do a hacked version of HTTP streaming in the past where I swap out the NetStream objects (similar to here http://video.leizhu.com/video.html), but there is always a momentary pause between the chunks. With the new appendBytes, I tried to do a quick mock up with the two sections of video from the preceding site, but even then, the skip still remains.
Does anyone know how the two consecutive .FLV files needs to be formated in order for the appendBytes method on the NetStream object to create a nice smooth video without a noticeable skip between the segments?
I was able to get this working using Adobe's File Packager Tool which Samuel described. I didn't use the NetStream object but I used the OSMF Sample Player which I assume uses this internally. Here's how to do with without using FMS:
Get Adobe's File Packager for Http Dynamic Streaming from http://www.adobe.com/products/httpdynamicstreaming/
Run the File Packager on an existing MP4 file containing H.264/AAC like this:
C:\Program Files\Adobe\Flash Media Server 4\tools\f4fpackager>
f4fpackager.exe --input-file="MyFile.mp4" --segment-duration=30
This will result in 30 second long F4F files, also F4X and a F4M file. The F4F files are your correctly segmented (and fragmented) MP4 files that should play.
If you want to test this using the OSMF Player also do the following:
Get Apache Server
Get Adobe's Http Origin Module for Apache from http://www.adobe.com/products/httpdynamicstreaming/
Install the module according to http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS8d6ed60bd880807c48597a9e1265edd6cc0-8000.html
Put the F4F, F4X and F4M file into the vod directory under httpdocs
Get the “OSMF Sample Player for HTTP Dynamic Streaming” from http://www.osmf.org/downloads/OSFMPlayer_zeri2.zip
Put the Sample Player in the httpdocs directory
Load the html file from the Sample Player in a browser eg http://localhost/OSMFPlayer.html
Press the eject button and put in the URL of your F4M file, it should play
So to answer the original question Adobe's File Packager is the file splitter to use, you don't need to buy FMS to use it and it works for FLV and MP4/F4V files.
You don't need to use their server. Wowza supports Adobe's version of HTTP Streaming and you can implement it yourself by segmenting the videos properly and loading all the segments on a standard HTTP server.
Links to all the specs for Adobe's HTTP Streaming are here:
http://help.adobe.com/en_US/HTTPStreaming/1.0/Using/WS9463dbe8dbe45c4c-1ae425bf126054c4d3f-7fff.html
Trying to hack the client to do some custom style http streaming will be a lot more troublesome.
Note that HTTP Streaming does not support streaming several different videos but streams a single file that was broken off into separate segments.
File Packager
A command-line tool that translates on-demand media files into fragments and writes the fragments to F4F files. The File Packager is an offline tool. You can use the File Packager to encrypt files for use with Flash Access. For more information, see Packaging on-demand media.
The File Packager is available from adobe.com and is installed with Adobe® Flash® Media Server to the rootinstall/tools/f4fpackager folder.
Packager download link is on right here: Download File Packager for HTTP Dynamic Streaming
http://www.adobe.com/products/httpdynamicstreaming/
You could use F4Pack, it's a GUI around the commandline-tool from Adobe, that lets you process your flv/f4v file so they can be used for HTTP Dynamic Streaming.
The place in the OSMF code where this happens is the timer-fired state machine inside of the HTTPNetStream class implementation... might be an informative read. I think I even put some helpful comments in there when I wrote it.
As far as the general question:
If you read an entire FLV file into a ByteArray and pass it to appendBytes, it will play. If you break that FLV file in half, and pass the first half as a byte array and then the second half as a byte array, that will play as well.
If you want to be able to switch around between bitrates without a gap, you need to split up your FLV files at matching keyframe points... and remember that only the first call to appendBytes has the initial FLV file header ('F', 'L', 'V', flags, offset)... the rest just expect a continuation of the FLV byte sequence.
I recently found a similar project for node.js to achieve m3u8 transcoding (https://github.com/andrewschaaf/media-server) but have yet to hear of one besides Wowza doing it outside of Origin module for Apache. Since the payloads are nearly identical you're better off looking for a good mp4 segmenting solution (plenty out there) than looking for f4m segmenting. The problem is moov atoms especially on larger mp4 video are difficult to manage and put in their proper initial (near beginning of file) location. Even using optimal ffmpeg settings and 'qtfaststart' you end up with noticeably slower seeking, inefficient bandwidth usage (usually greedy), and a few minor headaches relating to scrubbing/time that you don't get with flv/f4v playback.
In my player I have or intend to switch between HTTP Dynamic Streaming (HDS) and MP4 based on load and realtime log parsing Apache using awk/cron instead of licensing Adobe's Access product for stream protection .. both have unique 'onmetadata' handlers.. but in the end I receive sequenced time/byte hashes virtually equivalent. Just MP4 is slower. So mod_origin is just a synchronizer / request router for Flash clients (over http). I'm still looking for ways to speed up mp4-container-based playback. One incredible solution I read this recently and was rather awestruck by it http://zehfernando.com/2011/flash-video-frame-time-woes/ where a video editor (guy) and flash developer came up with their own mp4 timecoding solution that literally added (via Adobe Premiere script) about 50 pixels to the bottom of every video frame with a visual 'binary' stamp like a frame barcode.. and those binary values translate into highly-accurate timecode values. So Flash could analyze the video frames as they were painted (realtime) and determine precisely where the player was and what bytes were needed from any kind of mp4 byte-segmenting-friendly webserver. The thing is (and perhaps I'm wrong here) Flash seems to arbitrarily choose when it gets to moov data, especially on large video files (.5-1.5gigs). Even if you make sure to run your mp4 through MP4Box (i.e. MP4Box -frag 10000 -inter 0 movie.mp4) I guess this has been a problem OSMF and HDS have worked on quite well
now, though it is annoying that you need Apache and a proprietary closed-source module to use it imo. Its probably just a matter of time before open source implementations arrive as HDS is only 1-2 years old, and it just needs a little reverse engineering like that Andrew Chaaf guy with node.js + mpegts streaming (live or not).
In the end I may just end up using OSMF exclusively beneath my UI as it seems to have similar virtues to HDS if not more so i.e. Strobe if you need sick extensible HDS or MP4 open player platform to hack from to realize your own custom player.
Adobe's F4F format is based on MP4 files, are you able to use F4V or MP4 instead of FLV files?
There are plenty of MP4 file splitters around but you would need to make sure the timestamps in the files are continuous, maybe the pause happens when it sees a zero timestamp within the audio or video stream inside the file.