I'm working on a personal project where the iPhone connects to a server-type application running on a Mac. The iPhone send and receives textual/ASCII data via standard sockets. I now need to stream the microphone from the Mac to the iPhone. I've done some work with AudioServices before but wanted to check my thoughts here before getting too deep.
I'm thinking I can:
1. Create an Audio Queue in the standard Cocoa application on the Mac.
2. In my Audio Queue Callback function, rather than writing it to a file, write it to another socket I open for audio streaming.
3. On the iPhone, receive the raw sampled/encoded audio data from the TCP stream and dump it into an Audio Queue Player which outputs to headphone/speaker.
I know this is no small task and I've greatly simplified what I need to do but could it be as easy as that?
Thanks for any help you can provide,
Stateful
This looks broadly sensible, but you'll almost certainly need to do a few more things:
Buffering. On the "recording" end, you probably don't want to block the audio queue if the buffer is full. On the "playback" end, I don't think you can just pass buffers into the queue (IIRC you'll need to buffer it until you get a callback).
Concurrency. I'm pretty sure AQ callbacks happen on their own thread, so you'll need some sort of locking/barriers around your buffer accesses.
Buffer pools, if memory allocation ends up being a big overhead.
Compression. AQ might be able to give you "IMA4" frames (IMA ADPCM 4:1, or so); I'm not sure if it does hardware MP3 decompression on the iPhone.
Packetization, if e.g. you need to interleave voice chat with text chat.
EDIT: Playback sync (or whatever you're supposed to call it). You need to be able to handle different effective audio clock rates, whether it's due to a change in latency or something else. Skype does it by changing playback speed (with pitch-correction).
EDIT: Packet loss. You might be able to get away with using TCP over a short link, but that depends a lot on the quality of your wireless network. UDP is a minor pain to get right (especially if you have to detect an MTU hole).
Depending on your data rates, it might be worthwhile going for the lower-level (BSD) socket API and potentially even using readv()/writev().
If all you want is an "online radio" service and you don't care about the protocol used, it might be easier to use AVPlayer/MPMoviePlayer to play audio from a URL instead. This involves implementing a server which speaks Apple's HTTP streaming protocol; I believe Apple has some sample code that does this.
Related
I'm developing a kind of proxy for video streaming and I'm now dealing with an issue related to packets received out-of-order (without losses). This issue (maybe) is the reason why there are frequent noises in the video playback.
Do you know by chance if VLC is able to reorder packets? If so, it would mean that the reason why there are some noises in the playback is something else, if not, I should just develop an additional layer that ensure the reception with the correct order.
Thanks.
Assuming that you are talking about RTP over UDP, AFAIK VLC uses live555 libraries for client-side RTSP/RTP functionality and live555 has a built in jitter buffer that should take care of re-ordering for you. I can't recall the size of the jitter buffer of hand but 100ms seems to ring a bell.
In case you didn't know: When developing media streaming applications (esp. over UDP) it is important to increase the size of the receiver buffer. If it is full and packets get dropped, which could explain your artifacts.
Also, UDP being unreliable means that you will experience artifacts if packets are lost/corrupted and you have no suitable mechanism to deal with it.
I'd like to get real-time video from the iPhone to another device (either desktop browser or another iPhone, e.g. point-to-point).
NOTE: It's not one-to-many, just one-to-one at the moment. Audio can be part of stream or via telephone call on iphone.
There are four ways I can think of...
Capture frames on iPhone, send
frames to mediaserver, have
mediaserver publish realtime video
using host webserver.
Capture frames on iPhone, convert to
images, send to httpserver, have
javascript/AJAX in browser reload
images from server as fast as
possible.
Run httpServer on iPhone, Capture 1 second duration movies on
iPhone, create M3U8 files on iPhone, have the other
user connect directly to httpServer on iPhone for
liveStreaming.
Capture 1 second duration movies on
iPhone, create M3U8 files on iPhone,
send to httpServer, have the other
user connected to the httpServer
for liveStreaming. This is a good answer, has anyone gotten it to work?
Is there a better, more efficient option?
What's the fastest way to get data off the iPhone? Is it ASIHTTPRequest?
Thanks, everyone.
Sending raw frames or individual images will never work well enough for you (because of the amount of data and number of frames). Nor can you reasonably serve anything from the phone (WWAN networks have all sorts of firewalls). You'll need to encode the video, and stream it to a server, most likely over a standard streaming format (RTSP, RTMP). There is an H.264 encoder chip on the iPhone >= 3GS. The problem is that it is not stream oriented. That is, it outputs the metadata required to parse the video last. This leaves you with a few options.
Get the raw data and use FFmpeg to encode on the phone (will use a ton of CPU and battery).
Write your own parser for the H.264/AAC output (very hard)
Record and process in chunks (will add latency equal to the length of the chunks, and drop around 1/4 second of video between each chunk as you start and stop the sessions).
"Record and process in chunks (will add latency equal to the length of the chunks, and drop around 1/4 second of video between each chunk as you start and stop the sessions)."
I have just wrote such a code, but it is quite possible to eliminate such a gap by overlapping two AVAssetWriters. Since it uses the hardware encoder, I strongly recommend this approach.
We have similar needs; to be more specific, we want to implement streaming video & audio between an iOS device and a web UI. The goal is to enable high-quality video discussions between participants using these platforms. We did some research on how to implement this:
We decided to use OpenTok and managed to pretty quickly implement a proof-of-concept style video chat between an iPad and a website using the OpenTok getting started guide. There's also a PhoneGap plugin for OpenTok, which is handy for us as we are not doing native iOS.
Liblinphone also seemed to be a potential solution, but we didn't investigate further.
iDoubs also came up, but again, we felt OpenTok was the most promising one for our needs and thus didn't look at iDoubs in more detail.
I would like to stream video from an iPhone camera to an app running on a Mac. Think sorta like video chat but only one way, from the device to a receiver app (and it's not video chat).
My basic understanding so far:
You can use AVFoundation to get 'live' video camera data without saving to a file but it is uncompressed data and thus I'd have to handle compression on my own.
There's no built in AVCaptureOutput support for sending to a network location, I'd have to work this bit out on my own.
Am I right about the above or am I already off-track?
Apple Tech Q&A 1702 provides some info on saving off individual frames as images - is this the best way to go about this? Just saving off 30fps and then something like ffmpeg to compress 'em?
There's a lot of discussion of live streaming to the iPhone but far less info on people that are sending live video out. I'm hoping for some broad strokes to get me pointed in the right direction.
You can use AVCaptureVideoDataOutput and a sampleBufferDelegate to capture raw compressed frames, then you just need to stream them over the network. AVFoundation provides an API to encode frames to local video files, but doesn't provide any for streaming to the network. Your best bet is to find a library that streams raw frames over the network. I'd start with ffmpeg; I believe libavformat supports RTSP, look at the ffserver code.
Note that you should configure AVCaptureVideoDataOutput to give you compressed frames, so you avoid having to compress raw video frames without the benefit of hardware encoding.
This depends a lot on your target resolution and what type of frame rate performance you are targeting.
From an abstract point of view, I would probably have a capture thread to fill a buffer directly from AVCaptureOutput, and a communications thread to send and rezero the buffer (padded if need be) to a previously specified host every x milliseconds.
After you accomplish initial data transfer, I would work on achieving 15fps at the lowest resolution, and work my way up until the buffer overflows before the communication thread can transmit which would require balancing image resolution, buffer size (probably dependent on GSM, and soon to be CDMA frame sizes), and finally the maximum rate at which you can transmit that buffer.
Using FFMPEG, Live555, JSON
Not sure how it works but if you look at the source files at http://github.com/dropcam/dropcam_for_iphone you can see that they are using a combination of open source projects like FFMPEG, Live555, JSON etc. Using Wireshark to sniff the packets sent from one of the public cameras that's available to view with the free "Dropcam For Iphone App" at the App Store, I was able to confirm that the iphone was receiving H264 video via RTP/RTSP/RTCP and even RTMPT which looks like maybe some of the stream is tunneled?
Maybe someone could take a look at the open source files and explain how they got RTSP to work on the iphone.
Thanks for the info TinC0ils. After digging a little deeper I'v read that they have modified the Axis camera with custom firmware to limit the streaming to just a single 320x240 H264 feed, to better provide a consistent quality video over different networks and, as you point out, be less of a draw on the phone's hardware etc. My interest was driven by a desire to use my iphone to view live video and audio from a couple of IP cameras that I own without the jerkiness of MJPEG or the inherent latency that is involved with "http live streaming". I think Dropcam have done an excellent job with their hardware/software combo, I just don't need any new hardware at the moment.
Oh yeah, I almost forgot the reason of this post RTSP PROTOCOL DOES WORK ON THE IPHONE!
They are using open source projects to receive the frames and decoding in software instead of using hardware decoders. This will work, however, this runs counter to Apple's requirement that you use their HTTP Streaming. It will also require greater CPU resources such that it doesn't decode video at the desired fps/resolution on older devices and/or decrease battery life compared to HTTP streaming.
I'm wondering if there are any examples atomic examples out there for streaming audio FROM the iPhone to a server. I'm not interested in telephony or SIP style solutions, just a simple socket stream to send an audio clip, in .wav format, as it is being recorded. I haven't had much luck with the google or other obvious avenues, although there seem to be many examples of doing this the other way around.
i cant figure out how to register the unregistered account i initially posted with.
anyway, I'm not really interested in the audio format at present, just the streaming aspect. i want to take the microphone input, and stream it from the iphone to a server. i dont presently care about the transfer rate as ill initially just test from a wifi connection, not the 3g setup. the reason i cant cache it is because im interested in trying out some open source speech recognition stuffs for my undergraduate thesis. caching and then sending the recording is possible but then it takes considerably longer to get the voice data to the server. if i can start sending the data as soon as i start recording, then the response time is considerably improved because most of the data will have already reached the server by the time i let go of the record button. furthermore, if i can get this streaming functionality to work from the iphone then on the server side of things i can also start the speech recognizer as soon as the first bit of audio comes through. again this should considerably speech up the final amount of time that the transaction takes from the user perspective.
colin barrett mentions the phones and phone networks, but these are actually a pretty suboptimal solution for asr, mainly because they provide no good way to recover from errors - doing so over a voip dialogue is a horrible experience. however, the iphone and in particular the touch screen provide a great way to do that, through use of an ime or nbest lists for the other recognition candidates.
if i can figure out the basic architecture for streaming the audio, then i can start thinking about doing flac encoding or something to reduce the required transfer rate. maybe even feature extraction, although that limits the later ability to retrain the system with the recordings.