I'm using AVPlayer to play streamed network audio. I observer status of the streamed item as this post:
ios avplayer trigger streaming is out of buffer
Work seems well, but I encounter a strange problem. I meet the key "playbackLikelyToKeepUp" before the key "playbackBufferEmpty". I placed a log
NSLog(#"___path: %#", path)
in the first line of the function
observeValueForKeyPath.....
and the log I received is:
...
2012-10-29 17:24:35.412 NhacSo[236:907] ___path: rate
2012-10-29 17:24:35.413 NhacSo[236:907] ___path: playbackLikelyToKeepUp
2012-10-29 17:24:35.415 NhacSo[236:907] ___path: playbackBufferEmpty
2012-10-29 17:24:35.416 NhacSo[236:907] ___path: rate
...
Do you know why I receive "playbackLikelyToKeepUp" before "playbackBufferEmpty"? Thank you!!!
You receive playbackLikelyToKeepUp first because that property changes first. What I believe is confusing you is that it changes from YES to NO and not the other way around - that is, playback will no longer be able to keep up.
Related
IXAudio2SourceVoice has a GetState function which returns an XAUDIO2_VOICE_STATE structure. This structure has a SamplesPlayed member, which is:
Total number of samples processed by this voice since it last started, or since the last audio stream ended (as marked with the XAUDIO2_END_OF_STREAM flag).
What I want to be able to do it stop the source voice, flush all its buffers, and then reset the SamplesPlayed counter to zero. Neither calling Stop nor FlushSourceBuffers will by themselves reset SamplesPlayed. And while flagging the last buffer with XAUDIO2_END_OF_STREAM does correctly reset SamplesPlayed back to zero, this seemingly only works if that last buffer is played to completion; if the buffer is flushed, then SamplesPlayed does not get reset. I have also tried calling Discontinuity both before and after stopping/flushing with no effect.
My current workaround is, after stopping and flushing the source voice, to submit a tiny 1-sample silent buffer with the XAUDIO2_END_OF_STREAM flag set and then let the source voice play to process that buffer and thus reset SamplesPlayed to zero. This works fine-ish for my use case, but it seems pretty hacky/clumsy. Is there a better solution?
Looking at the XAudio2 source, there's no exposed way to do that in the API other than letting a packet play with XAUDIO2_END_OF_STREAM.
Calling Discontinuity sets up the end-of-stream flag on the currently playing buffer, or if there's none playing and a queued buffer it sets it there. You need to call Discontinuity and then let the voice play to completion before you recycle it.
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.
When I use <AudioToolbox/AudioServices.h> to implement a recording function, sometimes there will be unsuccessful recording. The reason is AudioQueueStart returned value -66681. The document says:
The audio queue has encountered a problem and cannot start
I found documents but I have no idea about that.
I'm trying to reproduce audio from an ICY stream. I'm able to reproduce that with AVPlayer and some good open source library but I'm not able to control the stream. I have no idea how I can get the percentage reproduced or how to seek to a specific time in the stream. Is that possible? Is there a good library that can help me?
Actually I'm using AFSoundManager but I'm always receiving negative numbers for percentage and I get invalid time when trying to seek the stream at a specified time.
That's the code that I'm using:
AFSoundManager.sharedManager().startStreamingRemoteAudioFromURL("http://www.abstractpath.com/files/audiosamples/sample.mp3") { (percentage, elapsedTime, timeRemaining, error, poppi) in
if error == nil {
//This block will be fired when the audio progress increases in 1%
if elapsedTime > 0 {
println(elapsedTime)
self.slider.value = Float(elapsedTime*1000)
}
} else {
//Handle the error
println(error)
}
I'm able of course to get the elapsedTime but not the percentage or the remainingTime. I always get negative numbers.
This code works perfectly with remote or local audio file but not with the stream.
This isn't possible.
These streams are live. There is nothing to seek to because what you haven't heard hasn't happened yet. Even streams that playback music end-to-end are still "live" in the sense that the audio you haven't received hasn't been encoded yet. (Small codec and transit buffers aside, of course.)
I'm working on a plugin for Apache Cordova that will allow audio streaming from a remote URL through the Media API.
The issue that I'm experiencing is that I get an EXC_BAD_ACCESS signal whenever I try to access certain properties on the AVPlayer instance. currentTime and isPlaying are the worst offenders. The player will be playing sound through the speakers, but as soon as my code reaches a player.currentTime or [player currentTime] it throws the bad access signal.
[player play];
double position = round([player duration] * 1000) / 1000;
[player currentTime]; //This will cause the signal
I am using ARC, so I'm not releasing anything that shouldn't be.
Edit:
Everything that I've done has been hacking around on the Cordova 3 CDVSound class as a proof of concept for actual streaming on iOS.
The original code can be found here: https://github.com/apache/cordova-plugin-media/tree/master/src/ios
My code can be found here:
CDVSound.h
CDVSound.m
The method startPlayingAudio will trip an exc_bad_access at line 346. Removing line 346 will cause audio to play, but it will trip a bad access later down the road when getCurrentPositionAudio and line 532 is called.
Edit / Solution
So it turns out that the best way to handle this is to use a AVPlayerItem and then access it with player.currentItem.currentTime. The real question then becomes, why isn't this behavior documented with AVPlayer and why does it behave like this?