I'm receiving a series of UDP packets from a socket containing encoded PCM buffers. After decoding them, I'm left with an int16 * audio buffer, which I'd like to immediately play back.
The intended logic goes something like this:
init(){
initTrack(track, output, channels, sample_rate, ...);
}
onReceiveBufferFromSocket(NSData data){
//Decode the buffer
int16 * buf = handle_data(data);
//Play data
write_to_track(track, buf, length_of_buf, etc);
}
I'm not sure about everything that has to do with playing back the buffers though. On Android, I'm able to achieve this by creating an AudioTrack object, setting it up by specifying a sample rate, a format, channels, etc... and then just calling the "write" method with the buffer (like I wish I could in my pseudo-code above) but on iOS I'm coming up short.
I tried using the Audio File Stream Services, but I'm guessing I'm doing something wrong since no sound ever comes out and I feel like those functions by themselves don't actually do any playback. I also attempted to understand the Audio Queue Services (which I think might be close to what I want), however I was unable to find any simple code samples for its usage.
Any help would be greatly appreciated, specially in the form of example code.
You need to use some type of buffer to hold your incoming UDP data. This is an easy and good circular buffer that I have used.
Then to play back data from the buffer, you can use Audio Unit framework. Here is a good example project.
Note: The first link also shows you how to playback using Audio Unit.
You could use audioQueue services as well, make sure your doing some kind of packet re-ordering, if your using ffmpeg to decode the streams there is an option for this.
otherwise audio queues are easy to set up.
https://github.com/mooncatventures-group/iFrameExtractor/blob/master/Classes/AudioController.m
You could also use AudioUnits, a bit more complicated though.
Related
I checked the video stream displayed well in qml video surface. now I want to get the video frame data to do something not bad thing. but, It seems not doing well until now... I made a simple pipeline like below for focus on a test.
nvarguscamerasrc - appsink
I used QGst::Utils::ApplicationSink to get a frame data. I referenced an example "appsink-src"
/* making pipeline */
QGst::ElementPtr source, sink;
SubClassApplicationSink *appsink;
source = QGst::ElementFactory::make("nvarguscamerasrc");
sink = QGst::ElementFactory::make("appsink");
appsink = new SubClassApplicationSink();
// configure elements
source->setProperty("sensor-id", n);
appsink->setElement(sink);
appsink->enableDrop(true);
appsink->setMaxBuffers(7654321);
m_pipeline->add(source, sink);
source->link(sink);
subclass of ApplicationSink implements some callbacks eos, preroll, sample.
and I prints logs some values in a buffer I got from the new sample.
the same outputs are repeated as callback function is called.
result: [start-end offsets are -1, no flags, memory count 1, memory size 1008]
I don't know why... How do you think?
I solved the issue. the problem was a pipeline's composition. after put a "nvvidconv" element between "nvarguscamerasrc" and "appsink" then I could get video frames successfully.
I don't know why needs a nvvidconv element. but, It seems because of source's video type, "video/x-raw(memory:NVMM)" which means using DMA buffers for performance reasons.
https://forums.developer.nvidia.com/t/what-is-the-meaning-of-memory-nvmm/180522
I am developing MIDI Player by referring to the following Web-Page.
http://twocentstudios.com/2017/02/20/bouncing-midi-to-audio-on-ios/
I don't do any recording, I just want to play the SMF file.
However, when I run setPreload (true), it says "ASSERTION FAILED: Preroll mode set during render" and my app hangs.
I searched for "Preroll mode set during render" but couldn't find any valid information.
Please help someone.
EDIT:
hi, #dspr.
The percussion sounds even if I don't do "AudioUnitSetProperty (kAUMIDISynthProperty_EnablePreload: 1)".
I think this is because the BANK for percussion is automatically assigned to ch.10.
However, in this state, the piano and guitar and others do not sound.
AVAudioUnitMIDI Instrument needs kAUMIDISynthProperty_EnablePreload to analyze which tone is assigned to which track in the SMF file, right?
Which method does AVAudioUnitMIDIInstrument use to preload SMF files?
(1) AudioUnitSetProperty (kAUMIDISynthProperty_EnablePreload: 1) to AVAudioUnitMIDISynth
(2) << How to preload? >>
(3) AudioUnitSetProperty (kAUMIDISynthProperty_EnablePreload: 0) to AVAudioUnitMIDISynth
(4) Start AVAudioSequencer
MIDI Player uses the kAUMIDISynthProperty_EnablePreload property of MIDISynth for that purpose. See the Apple comment about it below. Note the It should only be used prior to MIDI playback and must be set back to 0 before attempting to start playback sentence at the end :
/*!
#constant kAUMIDISynthProperty_EnablePreload
#discussion Scope: Global
Value Type: UInt32
Access: Write
Setting this property to 1 puts the MIDISynth in a mode where it will attempt to load
instruments from the bank or file when it receives a program change message. This
is used internally by the MusicSequence. It should only be used prior to MIDI playback,
and must be set back to 0 before attempting to start playback.
*/
EDIT : frankly, I'm a little bit reserved about your link
One strategy I haven’t tried would be to pitch shift the MIDI up one octave, play it back at 2x, record it at 88.2kHz, then downsample to 44.1kHz. AVAudioSession presumably can’t go past 48kHz though.
Clearly, the person who wrote that has a very poor knowledge about audio and sampling. Playing a MIDI song transposed one octave up at double tempo is really not equivalent than playing the same recorded in audio at double speed whatever you make the recording at 88.2kHz or any other sample rate. As a simple example, what happens is the file contains a drum set ? A snare drum (40) will become a Chinese cymbal (52) played two times slower ?
As I can understand this post, the described hack has for unique purpose to make recording. So if you simply want to play your MIDI file back you can certainly find a simpler and better example.
I want to play the recorded audio directly to speaker when headset is plugged in an iOS device.
What I did is calling AudioUnitRender in AURenderCallback func so that the audio data is writed to AudioBuffer structure.
It works well if the "IO buffer duration" is not set or set to 0.020seconds. If the "IO buffer duration" is set to a small value (0.005 etc.) by calling setPreferredIOBufferDuration, AudioUnitRender() will return an error:
kAudioUnitErr_CannotDoInCurrentContext (-10863).
Any one can help to figure out why and how to resolve it please? Thanks
Just wanted to add that changing the output scope sample rate to match the input scope sample rate of the input to the OSx kAudioUnitSubType_HALOutput Audio Unit that I was using fixed this error for me
The buffer is full so wait until a subsequent render pass or use a larger buffer.
This same error code is used by AudioToolbox, AudioUnit and AUGraph but only documented for AUGraph.
To avoid spinning or waiting in the render thread (a bad idea!), many
of the calls to AUGraph can return:
kAUGraphErr_CannotDoInCurrentContext. This result is only generated
when you call an AUGraph API from its render callback. It means that
the lock that it required was held at that time, by another thread. If
you see this result code, you can generally attempt the action again -
typically the NEXT render cycle (so in the mean time the lock can be
cleared), or you can delegate that call to another thread in your app.
You should not spin or put-to-sleep the render thread.
https://developer.apple.com/reference/audiotoolbox/kaugrapherr_cannotdoincurrentcontext
I recently watched the WWDC2014, Session on AVAudioEngine in practice, I have a question about the concept explained using AVAudioBuffers with NodeTap installed on the InputNode.
The Speaker mentioned that, its possible to notify the App module using Callback.
So my question is instead of Waiting for the callback until the buffer is full, is it possible to notify the app module after certain amount of time in ms. So once when the AVAudioEngine is started, is it possible to configure / register for Callback on this buffer for every 100 milliseconds of Recording. So that the App module gets notified to process this buffer for every 100ms.
Have anyone tried this before. Let me know your suggestions on how to implement this. It would be great if you point out some resource for this logic.
Thanks for your support in advance.
-Suresh
Sadly, the promising bufferSize argument of installTapOnBus which should let you choose a buffer size of 100ms:
input.installTapOnBus(bus, bufferSize: 512, format: input.inputFormatForBus(bus)) { (buffer, time) -> Void in
print("duration: \(buffer.frameLength, buffer.format.sampleRate) -> \((Double)(buffer.frameLength)/buffer.format.sampleRate)s")
}
is free to be ignored,
the requested size of the incoming buffers. The implementation may choose another size.
and is:
duration: (16537, 44100.0) -> 0.374988662131519s
So for more control over your input buffer size/duration, I suggest you use CoreAudio's remote io audio unit.
What I'm trying to do:
Record up to a specified duration of audio/video, where the resulting output file will have a pre-defined background music from external audio-file added - without further encoding/exporting after recording.
As if you were recording video using the iPhones Camera-app, and all the recorded videos in 'Camera Roll' have background-songs. No exporting or loading after ending recording, and not in a separate AudioTrack.
How I'm trying to achieve this:
By using AVCaptureSession, in the delegate-method where the (CMSampleBufferRef)sample buffers are passed through, I'm pushing them to an AVAssetWriter to write to file. As I don't want multiple audio tracks in my output file, I can't pass the background-music through a separate AVAssetWriterInput, which means I have to add the background-music to each sample buffer from the recording while it's recording to avoid having to merge/export after recording.
The background-music is a specific, pre-defined audio file (format/codec: m4a aac), and will need no time-editing, just adding beneath the entire recording, from start to end. The recording will never be longer than the background-music-file.
Before starting the writing to file, I've also made ready an AVAssetReader, reading the specified audio-file.
Some pseudo-code(threading excluded):
-(void)startRecording
{
/*
Initialize writer and reader here: [...]
*/
backgroundAudioTrackOutput = [AVAssetReaderTrackOutput
assetReaderTrackOutputWithTrack:
backgroundAudioTrack
outputSettings:nil];
if([backgroundAudioReader canAddOutput:backgroundAudioTrackOutput])
[backgroundAudioReader addOutput:backgroundAudioTrackOutput];
else
NSLog(#"This doesn't happen");
[backgroundAudioReader startReading];
/* Some more code */
recording = YES;
}
- (void)captureOutput:(AVCaptureOutput *)captureOutput
didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
fromConnection:(AVCaptureConnection *)connection
{
if(!recording)
return;
if(videoConnection)
[self writeVideoSampleBuffer:sampleBuffer];
else if(audioConnection)
[self writeAudioSampleBuffer:sampleBuffer];
}
The AVCaptureSession is already streaming the camera-video and microphone-audio, and is just waiting for the BOOL recording to be set to YES. This isn't exactly how I'm doing this, but a short, somehow equivalent representation. When the delegate-method receives a CMSampleBufferRef of type Audio, I call my own method writeAudioSamplebuffer:sampleBuffer. If this was to be done normally, without a background-track as I'm trying to do, I'd simply put something like this: [assetWriterAudioInput appendSampleBuffer:sampleBuffer]; instead of calling my method. In my case though, I need to overlap two buffers before writing it:
-(void)writeAudioSamplebuffer:(CMSampleBufferRef)recordedSampleBuffer
{
CMSampleBufferRef backgroundSampleBuffer =
[backgroundAudioTrackOutput copyNextSampleBuffer];
/* DO MAGIC HERE */
CMSampleBufferRef resultSampleBuffer =
[self overlapBuffer:recordedSampleBuffer
withBackgroundBuffer:backgroundSampleBuffer];
/* END MAGIC HERE */
[assetWriterAudioInput appendSampleBuffer:resultSampleBuffer];
}
The problem:
I have to add incremental sample buffers from a local file to the live buffers coming in. The method I have created named overlapBuffer:withBackgroundBuffer: isn't doing much right now. I know how to extract AudioBufferList, AudioBuffer and mData etc. from a CMSampleBufferRef, but I'm not sure how to actually add them together - however - I haven't been able to test different ways to do that, because the real problem happens before that. Before the Magic should happen, I am in possession of two CMSampleBufferRefs, one received from microphone, one read from file, and this is the problem:
The sample buffer received from the background-music-file is different than the one I receive from the recording-session. It seems like the call to [self.backgroundAudioTrackOutput copyNextSampleBuffer]; receives a large number of samples. I realize that this might be obvious to some people, but I've never before been at this level of media-technology. I see now that it was wishful thinking to call copyNextSampleBuffer each time I receive a sampleBuffer from the session, but I don't know when/where to put it.
As far as I can tell, the recording-session gives one audio-sample in each sample-buffer, while the file-reader gives multiple samples in each sample-buffer. Can I somehow create a counter to count each received recorded sample/buffers, and then use the first file-sampleBuffer to extract each sample, until the current file-sampleBuffer has no more samples 'to give', and then call [..copyNext..], and do the same to that buffer?
As I'm in full control of both the recording and the file's codecs, formats etc, I am hoping that such a solution wouldn't ruin the 'alignment'/synchronization of the audio. Given that both samples have the same sampleRate, could this still be a problem?
Note
I'm not even sure if this is possible, but I see no immediate reason why it shouldn't.
Also worth mentioning that when I try to use a Video-file instead of an Audio-file, and try to continually pull video-sampleBuffers, they align up perfectly.
I am not familiarized with AVCaptureOutput, since all my sound/music sessions were built using AudioToolbox instead of AVFoundation. However, I guess you should be able to set the size of the recording capturing buffer. If not, and you are still get just one sample, I would recommend you to store each individual data obtained from the capture output in an auxiliar buffer. When the auxiliar buffer reaches the same size as the file-reading buffer, then call [self overlapBuffer:auxiliarSampleBuffer withBackgroundBuffer:backgroundSampleBuffer];
I hope this would help you. If not, I can provide example about how to do this using CoreAudio. Using CoreAudio I have been able to obtain 1024 LCPM samples buffer from both microphone capturing and file reading. So the overlapping is immediate.