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!
Related
I'm going to build an iOS application in which you can download audio files and listen to them in offline mode. so far I found out the following process can be done to build the app: creating an xml file with list of audio files on the server and fetch xml with ASIHTTPRequest and parse it into the app (well I still have problem doing this, but my question is something else!)
Before start coding I want to know
what is the right data format for audio files to choose (I know that .mp3, .aac, .m4a, .mp4, .3gp, CAF, MPEG, WAVE, NeXT,... could be used), but I need to know if there is a best format to choose to have better functionality in the app.
is there special consideration that I should take about 3G and wifi connection to download files( I need to know about limitations or techniques for implementing them)
I ask these questions to be able to estimate the time for creating the app regarding techniques that I should use.
It is possible to implement a feature that allows users to watch videos as they are uploaded to server by others. Is html 5 suitable for this task? But flash? Are there any read to go solutions, don't want to reinvent the wheel. The application will be hosted on a dedicated server.
Thanks.
Of course it is possible, the data is there isnt it?
However it will be very hard to implement.
Also I am not so into python and I am not aware of a library or service suiting your requirements, but I can cover the basics of video streaming.
I assume you are talking about video files that are uploaded and not streams. Because, for that, there are obviously thousands of solutions out there...
In the most simple case the video being uploaded is already ready to be served to your clients and has a so called "faststart atom". They are container format specific and there are sometimes a bunch of them. The most common is the moov-atom. It contains a lot of data and is very complex, however in our use case, in a nutshell, it holds the data that enables the client to begin playing the video right away using the data available from the beginning.
You need that if you have progressive download videos (youtube...), meaning where a file is served from a Webserver. You obviously have not downloaded the full file and the player already can start playing.
If the fastastart atom was not present, that would not be possible.
Sometimes it is, but the player for example cannot display a progress bar, because it doesnt know how long the file is.
Having that covered the file could be uploaded. You will need an upload solution that writes the data directly to a buffer or a file. (file will be easier...).
This is almost always the case, for example PHP creates a file in the tmp_dir. You can also specify it if you want to find the video while its being uploaded.
Well, now you can start reading that file byte by byte and print that data to a connection to another client. Just be sure not to go ahead of what has already been recieved and written. You would probaby initiate your upload with a metadata set in memory that holds the current recieved byte position and location of the file.
Anyone who requests the file after the uploaded has started can just recieve the entire file, or if the upload is not yet finished, get it from your application.
You will have to throttle the data delivery or pause it when the data becomes short. This will appear to the client almost as a "slow connection". However you will have to echo some data from time to time to prevent the connection from closing. But if your upload doesnt stall, and why shoud it?, that shouldnt be a problem.
Now if you want to have someting like on the fly transcoding of various input formats into your desired output format, things get interesting.
AFAIK ffmpeg has neat apis which lets you directly deal with datasterams.
Also handbrake is a very good tool, however you would need to take the long road using external executeables.
I am not really aware of your requirements, however if your clients are already tuned in, for example on a red 5 streaming server, feeding data into a stream should also work fine.
Yes, take a look at Qik, http://qik.com/
"Instant Video Sharing ... Videos can be viewed live (right as they are being recorded) or anytime later."
Qik provides developer APIs, including ones like these:
qik.stream.subscribe_public_recent -- Subscribe to the videos (live and recorded)
qik.user.following -- Provides the list of people the user is following
qik.stream.public_info -- Get public information for a specific video
It is most certainly to do this, but it won't be trivial. And no, I don't think that you will find an "out of the box" solution that will require little effort on your behalf.
You say you want to let:
users watch videos as they are uploaded to server by others
Well, this could be interpreted two different ways:
Do you mean that you don't want a user to have to refresh the page before seeing new videos that other users have just finished uploading?
Or do you mean that you want one user to be able to watch a partially uploaded video (aka another user is still in the process of uploading it and right now the server only contains a partial upload of the video)?
Implementing #1 wouldn't be hard at all whatsoever. You would just need an AJAX script to check for newly uploaded videos, and those videos could then be served to the user in whatever way you choose. HTML5 vs. Flash isn't really a consideration here.
The second scenario, on the other hand, would require quite a bit of effort. I am guessing that HTML5 might not be mature enough to handle this type of situation. If you are not looking
to reinvent the wheel and don't have a lot of time to dedicate to this feature than I would say that you would be out of luck. You may be able to use ffmpeg to parse partial video files and feed them to a Flash player, but I would think of this as a large task.
I'm investigating a straightforward task:
open an audio file from the iPhone's 'iPod audio library'
allows the user to select a chunk by setting two markers: start and end time
time-reverse this chunk
save it as a new file
What are my options?
I will list the results of a couple of hours of research: ( forgive the mess, I will as always tidy pu once I have figured it out )
http://lists.apple.com/archives/coreaudio-api/2005/May/msg00096.html <-- 'I'm currently trying to create a program that plays back audio using an AUAudioFilePlayer
AudioUnit plugin that streams the audio to an output AudioUnit'
AUFilePlayer
http://lists.apple.com/archives/coreaudio-api/2008/Dec/msg00156.html
http://zerokidz.com/audiograph/docs/audiograph.pdf <-- this possibly links to code that does it, but it says it is in beta
When reading audio file with ExtAudioFile read, is it possible to read audio floats not consecutively? <-- this leads to an OS X project that reads an audio file from disk into memory; looking through the code leads us to:
https://developer.apple.com/library/mac/#documentation/MusicAudio/Reference/ExtendedAudioFileServicesReference/Reference/reference.html
as far as I can see the audio Graph project attempts to stream the audio from file in real-time, whereas Stephan's Project just exposes the audio; however it looks like he is using obsolete API calls.
this looks like the right code ( apart from the fact that there seems to be a bug in it ): https://stackoverflow.com/questions/8533143/decoding-mp3-files-by-extaudiofileopenurl
http://cocoadev.com/forums/discussion/499/core-audio/p1
https://developer.apple.com/library/ios/#samplecode/iPhoneExtAudioFileConvertTest/Introduction/Intro.html <-- here is an official Apple sample project that could probably be modified to get what I'm after
I believe you can use IphoneExtAudioFileConvertTest like you said to get what you need, after the user marks what times he wants, first thing you want to do is convert to PCM, next you need to find out which packets are the ones you need, write that to an audio file and then recompress... I wrote an answer here on how to get x amount of seconds from an audio file, ive only tested it with mp3 and m4a, but it can be adapted to PCM (pcm should be easier to do since its linear).
Daniel
A potential client has come to me asking for a an app which will stream a six hour audio file. The user needs to be able to set the "playback head" to any position along the file. Presumably, this means that the app must not be forced to download the entire file before it beings playing back starting at an arbitrary
An added complication -- there are actually four files which need to be streamed and mixed simultaneously.
My questions are:
1) Is there an out-of-the box technology which will allow me random access of streaming audio, on iOS? Can this be done with standard server technology and a single long file, or will it involve some fancy server tech?
2) Which iOS framework is best suited for this. Is there anything high-level that would allow me to easily mix these four audio files?
3) Can this be done entirely with standard browser technology on the client side? (i.e. HTML5)
Have a close look at the MP3 format. It is remarkably easy and efficient to parse, chop up into little bits, and reassemble into a custom stream.
Hence rolling your own server-side code to grab what you want and send to the client will not be as crazy or difficult as it may sound.
MP3 is also widely supported by various clients. I strongly suspect any HTML5 capable browser will be able of play the stream you generate via a long-lived bit-rate regulated HTTP request.
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.