I am trying to record audio from a microphone/iSight camera from Mac to a NSData object.
I have tried to do it using QTKit, but I found out that you could only save it as a .mov file.
But the fact is that I want to recode the audio into a FLAC file. Is that posible, or I'll need to use another framework?.
Thanks.
Grab the source for VLC (if you can deal w/GPL -- it has limitations on use that many find onerous) and have a read. It does transcoding, amongst other things.
Beyond that, one dead simple approach is to save as AIFF and then use a command line tool (via NSTask) to do the conversion.
Or you could just go with Apple Lossless -- it is open source these days.
Of course, this also begs the question; why do you need lossless compression when recording voice [low bandwidth in the first place] via a relatively sub-par microphone?
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Do you know any application that will display me all the headers/parameters of a single H264 frame? I don't need to decode it, I just want to see how it is built up.
Three ways come to my mind (if you are looking for something free, otherwise google "h264 analysis" for paid options):
Download the h.264 parser from (from this thread # doom9 forums)
Download the h.264 reference software
libh264bitstream provides h.264 bitstream reading/writing
This should get you started. By the way, the h.264 bitstream is described in Annex. B. in the ITU specs.
I've created a Web version - https://mradionov.github.io/h264-bitstream-viewer/
Based on h264bitstream and inspired by H264Naked. Done by compiling h264bitstream into WebAssembly and building a simple UI on top of it. Output information for NAL units is taken from H264Naked at the moment. Also supports files of any size, just will take some time initially to load the file, but navigation throughout the stream should be seamless.
I had the same question. I tried h264 analysis, but it only supports windows. So I made a similar tool with Qt to support different platforms.Download H264Naked. This tool is essentially a wrapper around libh264bitstream
New to Mac OS X, familiar with Windows. Windows has DirectShow, a good number of built-in filters, COM programming, and GraphEdit for very fast prototyping and snooping on the graphs you've constructed in code.
I'm now about to go to the Mac to work with cameras, webcams, microphones, color spaces, files, splitting, synchronization, rendering, file reading, file saving, and many of things I've come to take for granted with DirecShow when putting together applications for live performance. On the Mac side, so far I've found ... nothing! Either I don't know where to look or I'm having the toughest time tying the Mac's reputation for its ease of handling media with a coherent programmatic ability to get in there and start messin' with media manipulatin' building blocks.
I've seen some weak suggestions to use gstreamer or some library for QT but I can't bring myself to believe that this is the Apple way to go. And I've come across some QuickTime documentation but I'm not looking to do transitions, sprites, broadcasting, ...
Having a brain trained on DirectShow means I don't even know how Apple thinks about providing DirectShow-like functionality. That means I don't know the right keywords and don't even know where to look. Books? Bought a few. Now I might be able to write some code that can edit your sister's wedding video (if I can't make decent headway on this topic I may next be asking what that'd be worth to you), but for identifying what filters are available and how to string them together ... nothing. Suggestions?
Video handling is going through a huge transition on the Mac at the moment. QuickTime is very old, but also big and powerful, so it's been undergoing an incremental replacement process for the past 5 years or so.
That said, QTKit is the QuickTime subset (capture, playback, format conversion and basic video editing) which is supported going forward. The legacy QuickTime APIs are still there for the moment, and probably will remain at least until its major features are available elsewhere, but are 32-bit only. For some involved video stuff you may end up needing to use it in places.
At the moment, iOS is ahead of the Mac because it could start from scratch with AV Foundation. The future of the Mac media frameworks will probably either be AV Foundation directly (with QTKit being a lightweight shim over the top) or an extension of QTKit that looks very similar.
For audio there's Core Audio which is on Mac and iOS and isn't going away any time soon. It's quite powerful but somewhat obtuse in places. Luckily online support is very good; the mailing list is an essential resource.
For filters and frame-level processing you've got Core Video as someone else mentioned, as well as Core Image. For motion graphics there's Quartz Composer which includes a graphical editor and a plugin architecture to add your own patches. For programmatic procedural animation and easily mixing rendering models (OpenGL, Quartz, video, etc.) there's Core Animation.
In addition to all of these, of course there's no reason you can't use open source libraries where the built-in stuff doesn't do what you want.
To address your comment below:
In QuickTime (and QTKit), individual data types like audio and video are represented as tracks. It may not be immediately clear that QuickTime can open audio as well as video file formats. A common way to combine audio and video would be:
Create a QTMovie with your video file.
Create a QTMovie with your audio file.
Take the QTTrack object representing the audio and add it to the QTMovie with the video in it.
Flatten the movie, so it doesn't simply contain a reference to the other movie but actually contains the audio data.
Write the movie to disk.
Here's an example from Blender. You'll see how the A/V muxing is done in the end_qt function. There's also some use of Core Audio in there (AudioConverter*). (There's some classic QuickTime export code in quicktime_export.c but it doesn't seem to do audio.)
I'm trying to play a MP4-Stream. The stream is send from my android phone. The problem is, that the moov atom, which is needed to play the mp4, is only written if the phone is done with the recording progress. So at the moment I'm only to play the streamed data, after the recording has finished.
My idea was to write the ftyp and moov atom by myself, so that the streamed data can be played, while the phone is recording.
I tried to use the moov atom from another videofile but this didn't work. I also have read, that normally it's impossible to build a moov atom, if only the mdat atom is given.
But in my case I know the recording conditions, like framerate, etc..
So my question is, is it possible to generate a valid/useable moov atom for the incoming stream if I know the recording parameters?
It's possible. I've done it 4 years ago to implement "live streaming" to original iPhone. Just fill STSZ and STCO atoms with constant size frames, then pad each frame with zeros. Yeah, size will be huge, but you'll get real live streaming :-)
It seems to be rather impossible to stream not yet finished mp4 file because player would need special tables with chunks and offsets to locate every data sample. You can fake FTYP, MOOV and other atoms, but you can't generate all tables without having file complete. Better strategy would be to generate many short mp4 files and send them file by file...
Hi I have an adobe air 2 project which records some short sounds via the microphone. I am able to save the streams as wav files but require them to be saved as mp3 (For replaying in the flash player).
Does anyone know if this is possible?
If it isn't is there anyway to get the flash player to player audio in wav form?
Any hints appreciated.
If you are using Adobe AIR v2 then you should be looking for a Non-AS3 commandline tool to convert the wav file to mp3. The process is fairly CPU intensive and would take a LONG time in Actionscript even if there was a library out there that accomplished the task (which I haven't heard of).
My suggestion is to include a tool like LAME with you application and pass your wave file to it (essentially running the process in another thread in C). The only downside is providing an executable for each potential OS you'll be deploying to, if you intend on support Windows, Mac, and Linux that could be up to three different Wave->Mp3 commandline tools.
Link for LAME: http://lame.sourceforge.net/
Seems to be possible: http://www.jordansthings.com/blog/?p=5. No ready source, but libraries listed there should help. (I would just decompile it if need be.)
I have a multimedia application that among other things converts video using FFMpeg. Video conversion being the pain that it is, I have in my test suits some tests that check our ability to convert various video formats, with emphasis on sample videos known not to work.
A common problem we've noticed from users is that some videos end up with their audio desynched after being processed, and I am looking for a way to check this in my tests.
Extracting the audio portion of the resulting videos is not a problem.
My best idea so far would be to check the offset of the first non-silence at both the beginning and end and compare each between the two videos, but I'm hoping someone smart has a better idea.
The application language/environment is Java, but since this is for testing, I'm free to use any toolset.
The basic problem is likely that the video and audio are different lengths. Extract the audio and test its length vs. the video length. If they are significantly different (more than maybe .05 sec, I'm not really sure what is detectable as "off"), then there's a problem.
To fix it, re-encode the audio to match the video length, and then put the audio and video back into a container format.