I'm using libmpg123 as a library from native code (Android NDK) to decode a MP3 file to PCM (signed 16 bit) data. It works great, with the exception that the speakers make a weird sound once or twice every few seconds. To figure out what was wrong, I opened the audio in audacity and found this weird drop in audio sometimes.
The 'rest' of the audio sounds great, by the way. The overall quality is good.
Any ideas how this might happen? I'm totally lost...
Those are unclipped overflows (a 32768, where 32767 is the largest positive 16bit value). Are you doing anything to the data? Is what you opened the actual mp3 or the decoded pcm (just to be sure)?
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I'm using mp3 files to play sounds in my Android game, developed in Libgdx. The sounds play fine when they happen every now and then, but when I play them fast (footsteps in a running animation for example) the game freezes/stutters.
Every time a sound is played, I get this in the logs:
W/AudioTrack: AUDIO_OUTPUT_FLAG_FAST denied by client; transfer 4, track 44100 Hz, output 48000 Hz
I use libktx AssetStorage to store the sounds. I've been searching for this issue for a few days now and haven't gotten luck with any of the following solutions:
Override createAudio in AndroidLauncher and use AsynchronousAndroidAudio
Convert mp3 to ogg (using Audacity)
Convert to 48k rate sample (using Audacity)
Add 1 or seconds of silence to the file
I test it on my own device, Samsung Galaxy S5, which is quite old and has version Android 6.0.1.
What can I do to resolve this error and stuttering?
Decoding compressed audio can be a significant processing load. If it's a short recording (e.g., one footstep that is being repeated), I'd either package the sound file as a .wav or decode it into PCM to be held in memory, and use it that way. IDK if it's possible to output PCM directly with libgdx, though, but I do recall inspecting and tinkering with an ogg utility to have it decode into an array, and outputting it with a SourceDataLine for a non-libgdx Java project. I realize that SourceDataLine output is not an option with Android, but Android does have provisions for playing back raw PCM.
Another idea to explore is raising the priority of the thread that is processing the audio to Thread.MAX_PRIORITY if libgdx allows this. Theoretically, the audio processing thread spends most of its time in a blocked state, so doing this shouldn't hurt the performance, unless you are really going overboard with your audio requests.
I just saw the mismatch of sample rates. It's wasteful to repeatedly do conversions on the fly when you can do the conversion once in Audacity. I'm guessing the difference between outputting at 48000 vs 44100 isn't that big of a load difference. Seems to me 44100 should be fine, but I doubt using 48000 for everything adds much in terms of cpu load (or perceivable audio fidelity). So, whichever one you pick, spend a little time making sure all the assets match the format.
I'm playing mp3 file streamed from the network in my application, some mp3 files has weird behavior: mediaPlayer.getCurrentPosition() is larger than mediaPlayer.getDuration() at the end, for about 3 seconds.
The mp3 files are CBR encoded.
What might be the reason of this?
Finally solved the problem by converting the mp3 files, this is the command I'm using:
lame --mp3input -t -m s -b 128 --cbr input.mp3 output.mp3
There is a few reasons you can get this behavior.
First it appears that people had better results using mp3 files at exactly 44100Hz, because apparently the MediaPlayer class is assuming this value and scale the time accordingly, making strange values for files not using this sampling.
You also need to check the mode of your channels, and try using Joint Stereo or forced L/R Stereo. Joint should be the default, but your files might have been previously bad encoded, so it's worth trying. It's interesting to note that Forced L/R Stereo might loose quality for the same bitrate as Joint.
It would also be useful to check the output of soxi which is part of the sox package (you can also do it with ffmpeg), that will give you the number of channels, Sample rate, Bit Rate and Number of Channels.
Also you might want to check the raw content of the mp3 file if you did some treatment on them using any app for the presence of garbage xml content that might have been inserted during the export.
If you have the possibility to modify the mp3 files you're streaming, (which sounds like you do since you can tell the bitrate) these are what I would try first. If it's more like user-upload kind of stuff, maybe you should have a look to another solution instead, like ExoPlayer which has a few thousands stars and active development. It wraps the MediaPlayer api still, but worth a try.
You also have to consider that it might be a threading problem, where the player would stop playing, but the timer would actually keep going, giving you this result where it's superior to the actual duration of the song. 3 seconds seems a bit too much to explain it by that, but that's just a thought.
So I have an app where mp3 file is being played using the MediaPlayer. On most devices everything is fine but on Samsung and some other (like HTC One S) devices the same mp3 plays "too fast" (skipping gaps): looks like player does not handle sound gaps (silence) correctly. These mp3s are just speech and speech naturally has gaps (silence) between spoken words. And these gaps are not played correctly in terms of time - MediaPlayer just skips them. As result mp3 is played faster by the duration of all gaps it contains.
What could be a reason and solution for this?UPDATEI'd found that its about frequency+VBR. Somehow if mp3 is of 22050/24000/32000 Hz instead of 44100 or 48000 and VBR or ABR is used the issue raises up. Im using LAME for mp3 encoding. If I remove "--resample 22.05" option so the resulting mp3 becomes 44.1kHz there is no issue playing this mp3 on samsung phone. However the resulting size of mp3 becomes twice bigger which is not acceptable for me cuz in this case my apk becomes bigger than 50Mb. So now the question is how to properly compress mp3 as 22kHz/VBR/MONO.
The issue was fixed in the following way: I added a white noise to an original sound and then encoded it to MP3 format. Resulting files became bigger in size but also they become more compatible (with Samsung devices) The original audio file (made at recording studio) is too clean meaning that silence/pauses in speech (between pronounced words) has no waveform if look in sound editor, its like an ideal silence. So on variuos Samsung devices such MP3-encoded files played with described issue. However on most other devices and PCs such MP3 files played just fine. Once again - Samsung "rules"!
You need to Google our for controlling playback speed in your application I mean to say that there must be some sort of 'playback rate' variable which must be a floating point value something between 0 to 1. This might help you in some workarounds for your app hope you find this somewhat helpful in anyway . O by the way here are some useful links that might help you out as well and if not then we have to keep waiting in the waiting queue for Samsung ;-) if its specifically related to them happy coding
http://code.google.com/p/android/issues/detail?id=1961
play an mp3 with MediaPlayer class on Android issues
Regards
Anas.
I need to be able to play two or more (let's say, up to 5) short ogg files simultaneously. And by simultaneously I mean in perfect synchrony. I am able to load them to SoundPool and play, but this sometimes creates a noticeable difference in playback start time, which I want to get rid of.
From my understanding this can be avoided if mixing PCMs into one buffer and playing. But OGG's are not PCMs and need to be somehow efficiently decoded before playing and latency must be very low, ideally as soon as user presses the button. So I figured I need a way to stream OGG into PCM and as I receive buffers I would mix them and feed to AudioTrack. My requirement is Android 2.3.3+, so I cannot use any new codecs provided in Jelly Bean.
Also although OGGs themselves are small, there is a lot of them. So keeping them all decoded in memory (SoundPool or some pre-decoding) may case problems too.
Can someone give me a tip where to dig? Can OpenSL ES do that for me? Or should I think about integrating ffmpeg? And is it even possible to stream simultaneus files with low latency?
Thanks
You can play sounds using AssetPlayers, but this sometimes creates a noticeable difference in playback start time, yeh...
So, i recomend to decode ogg using Ogg Vorbis (like here) and then using this PCM buffer for BufferPlayer.
Btw, check this OpenSL ES wrappers
https://github.com/Suvitruf/Android-ndk/tree/master/OpenSLES
I'm writing an audio streaming app that buffers AAC file chunks, decodes those chunks to PCM byte arrays, and writes the PCM audio data to AudioTrack. Occasionally, I get the following error when I try to either skip to a different song, call AudioTrack.pause(), or AudioTrack.flush():
obtainbuffer timed out -- is cpu pegged?
And then what happens is that a split second of audio continues to play. I've tried reading a set of AAC files from the sdcard and got the same result. The behavior I'm expecting is that the audio stops immediately. Does anyone know why this happens? I wonder if its an Audio latency issue with Android 2.3.
edit: The AAC audio contains an ADTS Header. The header + audio payload constitute what I'm calling ADTSFrame. These are fed to the decoder one frame at a time. The resulting PCM byte array that gets returned from the C layer to the Java Layer gets fed to Android's AudioTrack API.
edit 2: I got my nexus 7 (Android 4.1 OS) today. Loaded the same APP onto the device. Didn't have any of these problems at all.
it is highly possible about sample rate. one of your devices might be supporting the sample rate u used while the other could not. Please check it. I had the same issue, it was about sample rate. use 44.1kHz (44100) and try again please.