I am creating an app which uses sockets to send data to other devices. I am using Http protocol to send and receive data. Now the problem is, i have to stream a video and i don't know how to send a video(or stream a video).
If the user directly jump to the middle of video then how should i send data.
Thanks...
HTTP wasn't really designed with streaming in mind. Honestly the best protocol is something UDP-based (SCTP is even better in some ways, but support is sketchy). However, I appreciate you may be constrained to HTTP so I'll answer your question as written.
I should also point out that streaming video is actually quite a deep topic and all I can do here is try to touch on some of the approaches that you might want to investigate. If you have control of the end-to-end solution then you have some choices to make - if you only control one end, then your choices are more or less dictated by what's available at the other end.
If you only want to play from the start of the file then it's fairly straightforward - make a standard HTTP request and just start playing as soon as you've buffered up enough video that you can finish downloading the file before you catch up with your download rate. You don't need any special server support for this and any video format will work.
Seeking is trickier. You could take the approach that sites like YouTube used to take which is to simply not allow the user to seek until the file has downloaded enough to reach that point in the video (or just leave them looking at a spinner until that point is reached). This is not the user experience that most people will expect these days, however.
To do better you need to be in control of the streaming client. I would suggest treating the file in chunks and making byte range requests for one chunk at a time. When the user seeks into the middle of the file, you can work out the byte offset into the file and start making byte range requests from that point.
If the video format contains some sort of index at the start then you can use this to work out file offsets - so, your video client would have to request at least enough to get the index before doing any seeking.
If the format doesn't have any form of index but it's encoded at a constant bit rate (CBR) then you can do an initial HEAD request and look at the Content-Length header to find the size of the file. Then, if the use seeks 40% of the way through the video, for example, you just seek to 40% of the way through the encoded frames. This relies on knowing enough about the file format that you can calculate an appropriate seek point so that you can identify framing information and the like (or at least an encoding format which allows you to resynchonise with both the audio and video streams even if you jump in at an arbitrary point in the file). This approach might also work with variable bit rate (VBR) as long as the format is such that you can recover from an arbitrary seek.
It's not ideal but as I said, HTTP wasn't really designed for streaming.
If you have control of the file format and the server, you could make life easier by making each chunk a separate resource. This is how Apple HTTP live streaming and Microsoft smooth streaming both work. They need tool support to pre-process the video, and I don't know if you have control of the server end. Might be worth looking into, however. These also do more clever tricks such as allowing a client to switch between multiple versions of the stream encoded at different bit rates to cope with differences in bandwidth.
Related
I was recently playing a video on YouTube, and a thought came to my mind. As videos are played, a user can skip ahead in the video and the video just resumes at that point without any trouble.
What I can't seem to find is how this works, I know that when I request a file through HTTP it downloads the entire thing, so starting a binary stream halfway through the video doesn't seem possible using HTTP. Is there any RFC or related document on how browsers do this?
Thank you
There are a couple of different technologies but they all essentially allow you specific an offset in the video and then download a 'chunk' from there.
The simple way to do this is with byte ranges and HTTP progressive download 0 there is an RFC which covers this:
https://www.rfc-editor.org/rfc/rfc7233
A similar but slightly more complex mechanism is behind the various adaptive bit rate protocols, such as HLS, MPEG-DASH, Smooth-streaming etc. These protocols break a video into multiple chunks (e..g 10 second long segments) and also create several different encoding of the video each at different bit rates.
The client can then request the next chunk based on current network conditions - if the network is busy or if the client is using a low band witch connection it can request the next chunk from a low bit rate encoding of the video. If network connectivity improves then it can request from progressively higher bit rates until it reaches the maximum.
You can see this in action if you look at the 'stats for nerds' available in YouTube if you right click on the video - look at the connection speed graph.
This mechanism also means the client can request chunks from further ahead (or behind) than the current position in the video - so long as it is not live obviously!
It also allows faster start up if you do jump ahead as the playback can start from a lower bit rate which is faster to download and work up to the higher bit rate again. You can often see this when playing around with services like Netflix - if you jump ahead it may be lower quality for a little time initially.
YouTube stores the videos in several chunks. Once each chunk completes its download, you can play that chunk of video. Think of them as individual split videos.
When you try to jump into the middle, they will start downloading the necessary chunk of video and start playing. Therefore, you can jump to the middle.
I'm using two custom push filters to inject audio and video (uncompressed RGB) into a DirectShow graph. I'm making a video capture application, so I'd like to encode the frames as they come in and store them in a file.
Up until now, I've used the ASF Writer to encode the input to a WMV file, but it appears the renderer is too slow to process high resolution input (such as 1920x1200x32). At least, FillBuffer() seems to only be able to process around 6-15 FPS, which obviously isn't fast enough.
I've tried increasing the cBuffers count in DecideBufferSize(), but that only pushes the problem to a later point, of course.
What are my options to speed up the process? What's the right way to do live high res encoding via DirectShow? I eventually want to end up with a WMV video, but maybe that has to be a post-processing step.
You have great answers posted here to your question: High resolution capture and encoding too slow. The task is too complex for the CPU in your system, which is just not fast enough to perform realtime video encoding in the configuration you set it to work.
I'm writing software which is demonstraiting video on demand service. One of the feature is something similiar to IIS Smooth Streaming - I want to adjust quality to the bandwith of the client. My idea is, to split single movie into many, let's say - 2 seconds parts, in different qualities and then send it to the client and play them. The point is that for example first part can be in very high quality, and second in really poor (if the bandwith seems to be poor). The question is - do you know any software that allows me to cut movies precisly? For example ffmpeg splits movies in a way that join is visible and really annoying (seconds are the measure of precision). I use qt + phonon as a player if it matters. Or maybe you know any better way to provide such feature, without splitting movie into parts?
Are you sure ffmpeg's precision is in seconds? Here's an excerpt from the man page:
-t duration
Restrict the transcoded/captured video sequence to the duration specified in seconds. "hh:mm:ss[.xxx]" syntax is also supported.
-ss position
Seek to given time position in seconds. "hh:mm:ss[.xxx]" syntax is also supported.
-itsoffset offset
Set the input time offset in seconds. "[-]hh:mm:ss[.xxx]" syntax is also supported. This option affects all the input files that follow it. The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by 'offset' seconds.
Looks like it supports up to millisecond precision, and since most video is not +1000 frames per second, this would be more than enough precision to accurately seek through any video stream.
Are you sure this is a good idea? Checking the bandwidth and switching out clips every two seconds seems like it will only allow you to buffer two seconds into the future at any given point, and unless the client has some Godly connection, it will appear extremely jumpy.
And what about playback, if the user replays the video? Would it recalculate the quality as it replays, or do you build the video file while streaming?
I am not experienced in the field of streaming video, but it seems what I see most often is that the provider has several different quality versions of their video (from extremely low to HD), and they test the user's bandwidth and then stream at an appropriate quality.
(I apologize if I misunderstood the question.)
I am using the below jQuery plugin for playing mp3
www.happyworm.com/jquery/jplayer
However, there is a bug in Flash that the total play (track) time won't show up correctly UNTIL AFTER the whole mp3 is completed downloaded.
I wonder if there is a way to work around this to get the correct total time using either javascript / another flash / even backend library in ASP.NET. Any suggestion helps. Thanks
You sure that's a bug? Looking at the header definition for the MP3 format I don't see any values for the length of the file. Generally applications that play MP3s would have to calculate the time, and that may not be doable until the entire file is downloaded. So the behavior you're seeing from Flash might be expected.
Theoretically if it's a fixed bitrate file (as opposed to VBR) then knowing the bitrate (gotten from the header) and the total size of the file should be enough to calculate it. However, the server would have to report the size of the file in the response headers (and that's not guaranteed to be accurate).
My guess is you'd need some service on the server that could calculate the length and report that to you in a separate request.
I intend on writing a small download manager in C++ that supports resuming (and multiple connections per download).
From the info I gathered so far, when sending the http request I need to add a header field with a key of "Range" and the value "bytes=startoff-endoff". Then the server returns a http response with the data between those offsets.
So roughly what I have in mind is to split the file to the number of allowed connections per file and send a http request per splitted part with the appropriate "Range". So if I have a 4mb file and 4 allowed connections, I'd split the file to 4 and have 4 http requests going, each with the appropriate "Range" field. Implementing the resume feature would involve remembering which offsets are already downloaded and simply not request those.
Is this the right way to do this?
What if the web server doesn't support resuming? (my guess is it will ignore the "Range" and just send the entire file)
When sending the http requests, should I specify in the range the entire splitted size? Or maybe ask smaller pieces, say 1024k per request?
When reading the data, should I write it immediately to the file or do some kind of buffering? I guess it could be wasteful to write small chunks.
Should I use a memory mapped file? If I remember correctly, it's recommended for frequent reads rather than writes (I could be wrong). Is it memory wise? What if I have several downloads simultaneously?
If I'm not using a memory mapped file, should I open the file per allowed connection? Or when needing to write to the file simply seek? (if I did use a memory mapped file this would be really easy, since I could simply have several pointers).
Note: I'll probably be using Qt, but this is a general question so I left code out of it.
Regarding the request/response:
for a Range-d request, you could get three different responses:
206 Partial Content - resuming supported and possible; check Content-Range header for size/range of response
200 OK - byte ranges ("resuming") not supported, whole resource ("file") follows
416 Requested Range Not Satisfiable - incorrect range (past EOF etc.)
Content-Range usu. looks like this: Content-Range: bytes 21010-47000/47022, that is bytes start-end/total.
Check the HTTP spec for details, esp. sections 14.5, 14.16 and 14.35
I am not an expert on C++, however, I had once done a .net application which needed similar functionality (download scheduling, resume support, prioritizing downloads)
i used microsoft bits (Background Intelligent Transfer Service) component - which has been developed in c. windows update uses BITS too. I went for this solution because I don't think I am a good enough a programmer to write something of this level myself ;-)
Although I am not sure if you can get the code of BITS - I do think you should just have a look at its documentation which might help you understand how they implemented it, the architecture, interfaces, etc.
Here it is - http://msdn.microsoft.com/en-us/library/aa362708(VS.85).aspx
I can't answer all your questions, but here is my take on two of them.
Chunk size
There are two things you should consider about chunk size:
The smaller they are the more overhead you get form sending the HTTP request.
With larger chunks you run the risk of re-downloading the same data twice, if one download fails.
I'd recommend you go with smaller chunks of data. You'll have to do some test to see what size is best for your purpose though.
In memory vs. files
You should write the data chunks to in memory buffer, and then when it is full write it to the disk. If you are going to download large files, it can be troublesome for your users, if they run out of RAM. If I remember correctly the IIS stores requests smaller than 256kb in memory, anything larger will be written to the disk, you may want to consider a simmilar approach.
Besides keeping track of what were the offsets marking the beginning of your segments and each segment length (unless you want to compute that upon resume, which would involve sort the offset list and calculate the distance between two of them) you will want to check the Accept-Ranges header of the HTTP response sent by the server to make sure it supports the usage of the Range header. The best way to specify the range is "Range: bytes=START_BYTE-END_BYTE" and the range you request includes both START_BYTE and byte END_BYTE, thus consisting of (END_BYTE-START_BYTE)+1 bytes.
Requesting micro chunks is something I'd advise against as you might be blacklisted by a firewall rule to block HTTP flood. In general, I'd suggest you don't make chunks smaller than 1MB and don't make more than 10 chunks.
Depending on what control you plan to have on your download, if you've got socket-level control you can consider writing only once every 32K at least, or writing data asynchronously.
I couldn't comment on the MMF idea, but if the downloaded file is large that's not going to be a good idea as you'll eat up a lot of RAM and eventually even cause the system to swap, which is not efficient.
About handling the chunks, you could just create several files - one per segment, optionally preallocate the disk space filling up the file with as many \x00 as the size of the chunk (preallocating might save you sometime while you write during the download, but will make starting the download slower), and then finally just write all of the chunks sequentially into the final file.
One thing you should beware of is that several servers have a max. concurrent connections limit, and you don't get to know it in advance, so you should be prepared to handle http errors/timeouts and to change the size of the chunks or to create a queue of the chunks in case you created more chunks than max. connections.
Not really an answer to the original questions, but another thing worth mentioning is that a resumable downloader should also check the last modified date on a resource before trying to grab the next chunk of something that may have changed.
It seems to me you would want to limit the size per download chunk. Large chunks could force you to repeat download of data if the connection aborted close to the end of the data part. Specially an issue with slower connections.
for the pause resume support look at this simple example
Simple download manager in Qt with puase/ resume support