My application is streaming live video, I want to alert them for after every 100MB download.
Any help will be appreciated.
Not easily because MPMoviePlayer internals doesn't reveal metadata about the stream.
If you actually knew for certain the data FPS and resolution (or bitrate for audio), using the elapsed time you could do an adhoc calculation by dividing them.
Related
I am attempting to stream live audio from an iOS device to a web browser. The iOS device sends small, mono wav files (as they are recorded) through a web socket. Once the client receives the wav files, I have the Web Audio API decode and schedule them accordingly.
This gets me about 99% of the way there, except I can hear clicks between each audio chunk. After some reading around, I have realized the likely source of my problem: the audio is being recorded at a sample rate of only 4k and this cannot be changed. It appears that the Web Audio API's decodeAudioData() function does not handle sample rates other than 44.1k with exact precision resulting in gaps between chunks.
I have tried literally everything I could find about this problem (scriptProcessorNodes, adjusting the timing, creating new buffers, even manually upsampling) and none of them have worked. At this point I am about to abandon the Web Audio API.
Is the Web Audio API appropriate for this?
Is there a better alternative for what I am trying to accomplish?
Any help/suggestions are appreciated, thanks!
Alas! AudioFeeder.js works wonders. I just specify the sampling rate of 4k, feed it raw 32 bit pcm data and it outputs a consistent stream of seamless audio! Even has built in buffer handling events, so no need to set any loops or timeouts to schedule chunk playback. I did have to tweak it a bit, though, to connect it to the rest of my web audio nodes and not just context.destination.
Note: AudioFeeder does automatically upsample to the audio context sampling rate. Going from 4k to 44.1k did introduce some pretty gnarly sounding artifacts in the highend, but a 48db lowpass filter (4 x 12db's) at 2khz got rid of them. I chose 2khz because, thanks to Harry Nyquist, I know that a sampling rate of 4k couldn't have possibly produced frequencies above 2khz in the original file.
All hail Brion Vibbers
I am working on an iPhone app that would capture realtime video from the iPhone camera over long periods of time. That part is fairly straightforward - the catch is that I only want the device to retain the last 2 minutes of recorded video at any time, discarding all images prior to this time period. I'm having trouble conceptualizing how this functionality might work. The only idea that comes to mind is to retain a stream of still images for the last 2 minutes, and discard outdated images. Then when the user stopped the recording, these images would be compiled into a video. It just seems really inefficient to hold on to almost 3000 images at a time.
I would love to hear any ideas for how to achieve this goal in a reasonably efficient manner. Thank you all in advance for your input!
Best,
James
Skip the idea with still images. You'd loose all the efficency that video codecs have to offer. Plus, I don't think the iPhone can handle that amount of data properly.
But maybe there's a compromise - it may be possible to record, say, ten seconds at a time, then start a new recording seamlessly. Dump the old recordings once they become old then two minutes.
I'm not quite sure if this is possible without loosing a few frames between the recordings though.
I can get individual frames from the iPhone's cameras just fine. what I need is a way to package them up with sound for streaming to the server. Sending the files once I have them isn't much of an issue. Its the generation of the files for streaming that I am having problems with. I've been trying to get FFMpeg to work without much luck.
Anyone have any ideas on how I can pull this off? I would like a known working API or instructions on getting FFMpeg to compile properly in an iPhone app.
You could divide your recording to separate files with a length of say, 10sec, then send them separately. If you use AVCaptureSession's beginConfiguration and commitConfiguration methods to batch your output change you shouldn't drop any frames between the files. This has many advantages over frame by frame upload:
The files can be directly used for HTTP live streaming without any server side processing.
The gap between data transfers allow the antennas to sleep in between if the connection is fast enough, saving battery life.
Conversely, if the connection is slow so upload is slower than recording, managing delayed upload of a set of files is much easier than a stream of bytes.
What I need to do is be able to mix 4 channels of audio (not from a live source, just prerecorded audio files in the app bundle), and change their volumes individually, in real time, preferably with MP3s. What's the best/correct road for me to take, regarding all the various sound APIs for the iPhone?
Thanks!
Storm Sim does this with AVAudioPlayer, which is certainly the simplest methdod. You can call prepareToPlay on each of the player objects then kick them off with play later so there won't be any delay. I also use a blank 1-second audio player on eternal loop to keep the deviceTime counting down, so you can use playAfter to give a specific deviceTime in the future to make all the samples play in-sync or offset relative to each other (deviceTime only ticks if there is some sort of audio playing). The AVAudioPlayerDelegate has interrupted/resumed events and finishedPlaying so you can get notification of what is happening.
However there is only one hardware MP3/AAC decoder, so the other three will use up CPU (and thus battery) doing the decoding. If you want to maximize battery life, use CAF files in IMA4#44100. It is about 1/4 the size of the raw WAV files so it isn't as good as MP3 but the performance is much better, especially if you are using a lot of small audio tracks. If you are using voice you can get away with much less fidelity and smash the files even more. afconvert in terminal can help you getting your source files in the CAF format (you should use CAF files no matter what the encoding).
I have read a lot of posts about live streaming in iPhone, but none of them really works.
The project I want to work out is as follow:
There is a MUTE movie streaming in a movie theater. I want to get the time code (the position it is playing) through wifi and makes iPhone/iPod Touch to play/stream an audio track at the same time code.
May I ask how to achieve it?
UPDATE: Latency is expected and will be taken into consideration. Small time difference is acceptable in this case.
The variable nature of a wireless connection and the latency involved will completely obliterate the video/audio sync you are trying to achieve.