I am building an application which needs to do real time audio recording. I am using Swift for the project - so unable to use Novocaine library (as it has some Obj-C++ code).
What I need is get small chunks of the audio recording (real-time) which I can process or send to my websocket. Is there a Swift library that I can use to achieve this?
In addition to getting the live audio from the microphone, I also need to show a real time waveform.
Start recording
Get an event every for few bytes of recorded data, where I can send these bytes to my websocket.
Showing a waveform for the audio.
Let me know.
You do not need any of 3-rd party tools for getting audio from mic. It can be set up easily using AVAudioEngine. However, for minimising network traffic I suggest to use lame for compressing raw PCM audio stream into mp3.
Here you can find project with minimal functionality for getting mic input and compressing into mp3. In this example project mp3 stores into Documents folder, so you can try and listen to make sure it works.
From this point you can take mp3 buffer and send via socket. You can also play with lame settings to change quality, etc.
There is another branch called no-lame where same functionality implemented without lame encoding. Look here
Related
Is it possible to record output audio in an app using Swift? So, for example, say I'm listening to a podcast, and I want to, within a separate app, record a small segment of the podcast's audio. Is there any way to do that?
I've looked around but have only been able to find information on recording microphone recording and such.
It depends on how you are producing the audio. If the production of the audio is within your control, you can put a tap on the output and record to a file as it plays. The easiest way is with the new AVAudioEngine feature (there are other ways, but AVAudioEngine is basically an easy front end for them).
Of course, if the real problem is to take a copy of a podcast, then obviously all you have to do is download the podcast as opposed to listening to it. Similarly, you could buffer and save streaming audio to a file. There are many apps that do this. But this is not because the device's output is being hijacked; it is, again, because we have control of the sound data itself.
I believe you'll have to write a kernel extension to do that
https://developer.apple.com/library/mac/documentation/Darwin/Conceptual/KEXTConcept/KEXTConceptIOKit/iokit_tutorial.html
You'd have to make your own audio driver to record it
It appears as though
That is how softonic made soundflowerbed.
http://features.en.softonic.com/how-to-record-internal-sound-on-a-mac
Is this possible to access the raw audio PCM data that is being played when using XAudio2 to play file?
I've been searching for several ways to access a decoded version of audio files being played in SL4/Windows Phone, without success.
According to this post someone had success writing a custom XAPO that just grabs samples and is enabled on a Submix Voice. http://social.msdn.microsoft.com/Forums/windowsapps/en-US/05593fad-dfd8-4c77-983b-8c84cd4a324b/xaudio2-saving-output-custom-xapos-slow-down-audio-play-backwards
Please note that if you just want to do this for audio processing this approach is not optimal because you are limited to the speed of audio playback.
I've got experience with building iOS apps but don't have experience with video. I want to build an iPhone app that streams real time video to a server. Once on the server I will deliver that video to consumers in real time.
I've read quite a bit of material. Can someone let me know if the following is correct and fill in the blanks for me.
To record video on the iPhone I should use the AVFoundation classes. When using the AVCaptureSession the delegate method captureOutput:didOutputSampleBuffer::fromConnection I can get access to each frame of video. Now that I have the video frame I need to encode the frame
I know that the Foundation classes only offer H264 encoding via AVAssetWriter and not via a class that easily supports streaming to a web server. Therefore, I am left with writing the video to a file.
I've read other posts that say they can use two AssetWritters to write 10 second blocks then NSStream those 10 second blocks to the server. Can someone explain how to code the use of two AVAssetWriters working together to achieve this. If anyone has code could they please share.
You are correct that the only way to use the hardware encoders on the iPhone is by using the AVAssetWriter class to write the encoded video to a file. Unfortunately the AVAssetWriter does not write the moov atom to the file (which is required to decode the encoded video) until the file is closed.
Thus one way to stream the encoded video to a server would be to write 10 second blocks of video to a file, close it, and send that file to the server. I have read that this method can be used with no gaps in playback caused by the closing and opening of files, though I have not attempted this myself.
I found another way to stream video here.
This example opens 2 AVAssetWriters. Then on the first frame it writes to two files but immediately closes one of the files so the moov atom gets written. Then with the moov atom data the second file can be used as a pipe to get a stream of encoded video data. This example only works for sending video data but it is very clean and easy to understand code that helped me figure out how to deal with many issues with video on the iPhone.
i am developing one streaming application for iOS and i am getting all audio packets correctly and decoded also but now i am totally confused about how to play it on iPhone.. I have decoded packets using ffmpeg.. All codes i get so far are playing audio from a file but i my case i have to play audio packets which i am getting from server in an order they are coming.. I dont want to save all packets to a file so any code that will help me to solve my problem is appreciated..
Thnak you...
You'll need to use Audio Services to do this. Either AudioQueue or AudioUnit. AudioQueue is better for streaming type applications.
The classic sample - for AudioQueue - is Apple's SpeakHere.
Matt Gallagher also has some superb tutorials with sample code for streaming.
See Streaming MP3/AAC audio again.
If you want to go the AudioUnit route, see Using RemoteIO audio unit.
By basing your code on Matt Gallagher's sample, possibly also using SpeakHere, you should be able to play your decoded packets. See my other answers for how to play using a buffer rather than from a file.
Don't forget that this is quite advanced stuff. You'll need to be comfortable with buffers, pointers, etc. Make sure you understand frames, packets, etc. as well. Some pain in getting your audio out there is to be expected.
Long story short, I am trying to implement a naive solution for streaming video from the iOS camera/microphone to a server.
I am using AVCaptureSession with audio and video AVCaptureOutputs, and then using AVAssetWriter/AVAssetWriterInput to capture video and audio in the captureOutput:didOutputSampleBuffer:fromConnection method and write the resulting video to a file.
To make this a stream, I am using an NSTimer to break the video files into 1 second chunks (by hot-swapping in a different AVAssetWriter that has a different outputURL) and upload these to a server over HTTP.
This is working, but the issue I'm running into is this: the beginning of the .mp4 files appear to always be missing audio in the first frame, so when the video files are concatenated on the server (running ffmpeg) there is a noticeable audio skip at the intersections of these files. The video is just fine - no skipping.
I tried many ways of making sure there were no CMSampleBuffers dropped and checked their timestamps to make sure they were going to the right AVAssetWriter, but to no avail.
Checking the AVCam example with AVCaptureMovieFileOutput and AVCaptureLocation example with AVAssetWriter and it appears the files they generate do the same thing.
Maybe there is something fundamental I am misunderstanding here about the nature of audio/video files, as I'm new to video/audio capture - but thought I'd check before I tried to workaround this by learning to use ffmpeg as some seem to do to fragment the stream (if you have any tips on this, too, let me know!). Thanks in advance!
I had the same problem and solved it by recording audio with a different API, Audio Queue. This seems to solve it, just need to take care of timing in order to avoid sound delay.