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
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I'm trying to get samples from AudioQueue to show spectrum of music (like in iTunes) on iPhone.
Ive read a lot of posts but almost all asks about get samples when Recording, not playing :(
I'm using AudioQueue Services for streaming audio. Please help to understanding next points:
1/ Where can I get access to samples (PCM, non mp3 (I'm using mp3 stream)
2/ Should I collect samples in my own buffer to apply fft ?
3/ Is it possible get frequencies without fft transformations ?
4/ How can I synchronize my fft shift in buffer with current playing samples ?
thanks,
update:
AudioQueueProcessingTapNew
For iOS6+, this works fine for me. But what about iOS5 ?
For playing audio, the idea is to get at the samples before you feed them to the Audio Queue callback. You may need to convert any compressed audio file format into raw PCM samples beforehand. This can be done using one of the AVFoundation converter or file reader services.
You can then copy frames of data from the same source used to feed the Audio Queue callback buffers, and apply your FFT or other DSP for visualization to them.
You can use either FFTs or a bank of band-pass filters to get frequency info, but the FFT is very efficient at this.
Synchronization needs to done by trial-and-error, as Apple does not specify exact audio and view graphic display latencies, which may differ between iOS devices and OS versions anyway. But short Audio Queue buffers or using the RemoteIO Audio Unit may give you better control of the audio latency, and OpenGL ES will give you better control of the graphic latency.
I'm currently playing a stream on my iOS App but one feature we'd like to add is the visualization of the output wave. I use an output audio queue in order to play the stream, but have found no way to read the output buffer. Can this be achieved using audio queues or shall be done wit a lower level api?
To visualize, you presumably need PCM (uncompressed) data, so if you're pushing some compressed format into the queue like MP3 or AAC, then you never see the data you need. If you were working with PCM (maybe you're uncompressing it yourself with the Audio Conversion APIs), then you could visualize before putting samples into the queue. But then the problem would be latency - you want to visualize samples when they play, not when they go into the queue.
For latency reasons alone, you probably want to be using audio units.
It cannot actually be done. In order to do so, I need audio units to implement the streamer.
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.
I'm looking to create an app that emulates a physical instrument. I've got audio samples but I want to be able to increase the pitch/frequency dynamically so I don't have to load from too many files.
Any idea which audio API will be able to do this? I reckon either OpenAL or Audio Queue Services but am not sure which is suitable. Any links to guides/sample code is also much appreciated.
Thanks in advance.
I went down this road in 2009, trying Audio Toolkit, Audio Queue Services, openAL, and finally settling on the RemoteIO AudioUnit.
Audio Toolbox is fine for basic triggered sound effects, but it wasn't able to change frequencies or loop samples.
Audio Queue Services can loop samples, but the only way I could find to adjust the playback frequency of a sample was to re-read the data from the file -- very painful. Plus, the framework is tremendously cumbersome - I'd only use it if I was trying to stream something off the Internet.
OpenAL was a godsend - was up and running with it in under an hour, after getting my hands on the no-longer-available-from-Apple "CrashLanding" iPhone sample app. I found OpenAL to be ideally suited to games or even a musical instrument -- samples could be pre-loaded, adjusting the frequency was easy, and looping was no problem. The deal-breaker for me was that starting and stopping a looped sample would result in a nasty "pop" almost every time. Also the builtin 3d positional audio mixer was a bit too CPU-intensive for my liking.
If your instrument does not use looped samples, I'd suggest trying the OpenAL route first - the learning curve is much less intimidating. Try to track down "SoundEngine.h", "CrashLanding" or "TouchFighter", or check out the following link:
http://benbritten.com/blog/2008/11/06/openal-sound-on-the-iphone/
Since looped samples was a requirement for me, I finally settled on AudioUnits (which, on the iPhone, is referred to as "RemoteIO" if you want to do input or output). It was tremendously difficult to implement - very similar to Audio Queue Services, in that the core of your implementation will be inside a "buffer callback", being called several times per second to fill a buffer of outbound audio with raw SInt16 values.
Ultimately, I got my instrument working beautifully with multi-note polyphony, looped samples, no popping, and minimal latency.
Unfortunately, RemoteIO is not well documented. Michael Tyson was one of the first in the field to write about RemoteIO at length, and his posts (and the comments) were very useful to me:
http://michael.tyson.id.au/2008/11/04/using-remoteio-audio-unit/
Good luck!
Edited years later: I've open-sourced the RemoteIO/AudioUnits code I alluded to above: https://github.com/glenn-barnett/hexaphone/blob/master/Classes/Instrument.m - apologies for the mess, I hope to get some time to clean up the code and comments.
Try creating an Audio Unit. I'm doing something similar an AU worked well for me.
Initially I used an audio queue as it was simpler (higher level?) and
synchronous, however it was lacking in responsiveness, so I dumped it for
the Audio Unit.
It sounds, a bit, like you're creating essentially the wavetable synthesis method of playing MIDI files. You might be able to find a MIDI synthesizer for the iPhone that you can use, and then use your audio samples to build a wavetable set. Anytime you'd want to play tones, you would simply send the MIDI event into the iPhone MIDI synth with your loaded wavetable set.
Another option now is AUSampler.
http://developer.apple.com/library/mac/#technotes/tn2283/_index.html