In order to minimize read/write traffic in an application, we worked out a strategy that recovers the ftyp, uuid, and moov boxes of mp4s stored in AWS S3, writes them to a local file, then ffprobes that data to recover the movie duration.
This has been working without problems for a long time, but since June 1, 2016, I've been noticing more files hitting ffprobe errors during this process. At first I thought it was because of the ftyp these files provided. While they were indeed not any of the ftyps we are spec'd to accept, I nevertheless found that ffprobe did work on the original (unabridged) mp4 file.
The errors are similar to this one:
[mov,mp4,m4a,3gp,3g2,mj2 # 0x228c020] stream 0, offset 0x6b8: partial file
[mov,mp4,m4a,3gp,3g2,mj2 # 0x228c020] Could not find codec parameters for stream 0 (Video: h264 (avc1 / 0x31637661), 1504x1504, 678 kb/s): unspecified pixel format
Consider increasing the value for the 'analyzeduration' and 'probesize' options
tmp_9NdLYBkm8sNw5TBNGAz24g_0.mp4: Invalid data found when processing input
I read several questions mentioning the analyzeduration and probesize options, and I tried with extremely values, but this did not affect the outcome.
I've also triple checked that the ftyp, moov and uuid boxes that were recovered were in fact the same dimensions as those in the original file, so I don't believe them to be at fault for being incomplete.
So, I'm led to believe that we are not recovering everything we need from the mp4. Really, we're hoping not to have to recover the entire mdat box. Is there any reason to expect that would be indispensable in some cases?
Related
I'm trying to build simple app that would stream video from camera using browser to the remote server.
For the camera access from browser I've found a wonderful WebRTC API: getUserMedia.
Now for the streaming it to the server IIUC the best way would be to use some of the WebRTC_API for transporting and then use some server side library to deal with it.
However, at first I went with a bit different approach:
I've user MediaRecorder based on the stream from camera. And then I was setting the timeslice for the MediaRecorder.start() to be few hundred Ms, e.g. 200. And I had some assumptions in wrt MediaRecorder which are not in sync with what I was observing:
I've observed weird behaviour(wrt to my assumptions about MediaRecorder):
If there was only 1 chunk uploaded to server -> it opens just fine.
If there are multiple chunks -> none of them loads correctly, they give errors: Could not determine type of stream. But then if I use ffmpeg to concat all the chunks - resulting file is fine. Same happens if I'm concatenating the blobs from MediaRecorder.ondataavailable on the client.
Thus the question:
Can the chunks in theory be independent video files? Or it is not what MediaRecorder was designed for? If it is not - then why do we even have the option to give timeslice parameter in its start() method?
Bonus question
If we're setting timeslice comparatively small, e.g. 10ms -> lots of data blobs that are sent to MediaRecorder.ondataavailable are of size 0. Where we can find some sort of guarantees/specs on the minimal timeslice that we can use, so that the data blobs are meaningful?
In the documentation there are the following:
If timeslice is not undefined, then once a minimum of timeslice milliseconds of data have been collected, or some minimum time slice imposed by the UA, whichever is greater, start gathering data into a new Blob blob, and queue a task, using the DOM manipulation task source, that fires a blob event named dataavailable at recorder with blob.
So, my guess is that it is somehow related to some data blobs being of 0 size. What does it "some minimum time slice imposed by the UA" mean?
PS
Happy to provide code if needed. But the question is not about some specific code. It is to get understanding of the assumptions behind the MediaRecorder API and why they are there.
The timeslice parameter does not allow to create independent media chunks; instead, it gives an opportunity to save data (e.g. on the filesystem, or uploaded to a server) on a regular basis, rather than holding potentially large media content in memory.
Can the ext4 filesystem detect data corruption of file contents? If yes, is it enabled by default and how can I check for corrupted data?
I have read that ext4 maintains checksums for file metadata and its journal, but I was unable to find any information on checksums for the actual file contents.
For clarity: I want to know if a file has changed since the last write operation.
No, ext4 doesn't and can't detect file content corruption.
Well known file systems implementing silent data corruption detection and therefore able to correct it when enough redundancy is available are ZFS and btrfs.
They do it by computing and storing a CRC for every data block written and by checking the CRC or each data block read. Should the CRC doesn't match the data, the latter is not provided to the caller and either RAID allows for an alternate block to be used instead, or an I/O error is reported.
The reading process will never receive corrupted data, either it is correct or the read fails.
"Can the ext4 filesystem detect data corruption of file contents?"
Not in the sense you are expecting. It performs journaling, creating a boolean {before vs after} copy to ensure io completion.
A CRC / checksum is a test for modification from a known state and although the CRC or checksum may not compare to the original, that does not imply that the file is then "corrupt" (aka invalid) - - it only says it has been changed. Strictly speaking, one form of "corruption" would be to alter the 'magic number' at the beginning of a file, like changing %PDF to %xYz - - that would make the content unusable to any program.
"... to know if a file has changed since the last write operation".
Systems that track mtime() will do so uniformly, so every write will modify mtime() making your request impossible.
The only way mtime() would not reflect last write io would be media degredation.
The question may looks duplicate. But i am not getting the answer which i am looking.
The problem is, in unix, one of the 4GL binary is fetching data from the table using cursor and writing the data in .txt file.
The table contains around 50 Million records.
The binary took lot of time and not completing. the .txt file is also 0 byte.
I want to know the possibilities why the records are not written in the .txt file.
Note: There is enough disk space available.
Also, for 30 Million records, i can get the data in the .txt file as i expected.
The information you provide is insufficient to tell for sure why the file is not written.
In UNIX, a text file is just like any another file - a collection of bytes. No specific limit (or structure) is enforced on "row size" or "row count," although obviously, some programs might have certain limits on maximum supported line sizes and such (depending on their implementation).
When a program starts writing data to a file (i.e. once the internal buffer is flushed for the first time) the file will no longer be zero size, so clearly your binqary is doing something else all that time (unless it wipes out the file as part of the cleanup).
Try running your executable via strace to see the file I/O activity - that would give some clues as to what is going on.
Try closing the writer if you are using one to write to the file. It achieves the dual purpose of closing the resource along with flushing the remaining contents of the buffer.
CPU calculated output needs to be flushed if you are using any mechanism of buffered writer. I have encountered such situations a few times and in almost all cases, the issue was that of flushing the output.
In java specifically, usually the best practice of writing data involves buffers. So when the buffer limit is reached, it gets written to the file but doesn't get written to the file when the end of buffer has not been reached yet. This happens when program closes without flushing the buffered writer.
So, in your case, if the processing time that it takes is reasonable and still the output is not on the file, it may mean that the output has been calculated and put on the RAM but could not be written to the file (which represents the disk) due to the output not being flushed.
You can also consider the answers to this question.
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.
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