I have been scratching my head for the past week to do this effect on the text. http://www.youtube.com/watch?v=gB2PL33DMFs&feature=related
Would be great if someone can give me some tips or guidance or tutorial on how to do this.
thankz for reading and answering =D
If all you want is to display a movie with video and sound, a MediaPlayer can do that easily.
So I assume that you're actually talking about synchronizing some sort of animated display with a sound file being played separately. We did this using a MediaPlayer and polling getCurrentPosition from within an animation loop. This more or less works, but there are serious problems that need to be overcome. (All this deals with playing mp3 files; we didn't try any other audio formats).
First, your mp3 must be recorded at 44,100 Hz sampling rate. Otherwise the value returned by getCurrentPosition is way off. (We think it's scaled by the ratio of the actual sampling rate to 44,100, but we didn't verify this hypothesis.) A bit rate of 128,000 seems to work best.
Second, and more serious, is that the values returned by getCurrentPosition seem to drift away over time from the sound coming out of the device. After about 45 seconds, this starts to be quite noticeable. What's worse is that this drift is significantly different (but always present) in different OS levels, and perhaps from device to device. (We tested this in 2.1 and 2.2 on both emulators and real devices, and 3.0 on an emulator.) We suspected some sort of buffering problem, but couldn't really diagnose it. Our work-around was to break up longer mp3 files into short segments and chain their playback. Lots of bookkeeping aggravation. This is still under test, but so far it seems to have worked.
Ted Hopp: time drifting on MP3 files is likely caused by those MP3 files being VBR. I've been developing Karaoke apps for a while, and pretty much every toolkit - from Qt Phonon to ffmpeg - had problems reporting correct audio position on variable MP3 files. I assume this is because they all try to calculate the current audio position by using the number of decoded frames, which makes it unreliable for VBR MP3s. I described it in a user-friendly way in the Karaoke Lyrics Editor FAQ
Unfortunately the only solution I found is to recode MP3s to CBR. Another was to ditch the current position completely, and rely only on system clocks. That actually produced a better result for VBR MP3s, but still not as good as recoding them into CBR.
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I'm making a really small game for android, and trying to add sounds to the game
I'm using the MediaPlayer class to load the audio file (.ogg or .wav)
I want to use .ogg (or .mp3) to shrink the size of the apk, rather than using .wav files.
I understand why loading (i.e. creating the MediaPlayer from a .ogg would take longer than .wav) (compression)
BUT the problem is that when I put the audio on loop by audio.setLooping(true) each time the audio starts again, it lags the game significantly
Why ? does the audio gets decoded each time it's going to start ? even on a loop ?
Also, on the CPU usage, I see spikes marking the beginning of the audio in the loop. so I'm pretty sure that the loop is really causing the lag..
any explanations/solutions?
(P.s. I'm testing on a really low-end physical phone, but for the game it's fairly enough, the sudden spikes are what's causing the problem not the actual usage)
In my android application I need to capture the user's speech from the microphone and then pass it to the server. Currently, I use the MediaRecorder class. However, it doesn't satisfy my needs, because I want to make glowing effect, based on the current volume of input sound, so I need an AudioStream, or something like that, I guess. Currently, I use the following:
this.recorder = new MediaRecorder();
this.recorder.setAudioSource(MediaRecorder.AudioSource.MIC);
this.recorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
this.recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
this.recorder.setOutputFile(FILENAME);
I am writing using API level 7, so I don't see any other AudioEncoders, but AMR Narrow Band. Maybe that's the reason of awful noise which I hear in my recordings.
The second problem I am facing is poor sound quality, noise, so I want to reduct (cancel, suppress) it, because it is really awful, especially on my noname chinese tablet. This should be server-side, because, as far as I know, requiers a lot of resources, and not all of the modern gadgets (especially noname chinese tablets) can do that as fast as possible. I am free to choose, which platform to use on the server, so it can be ASP.NET, PHP, JSP, or whatever helps me to make the sound better. Speaking about ASP.NET, I have come across a library, called NAudio, may be it can help me in some way. I know, that there is no any noise reduction solution built in the library, but I have found some examples on FFT and auto-corellation using it, so it may help.
To be honest, I have never worked with sound this close before and I have no idea where to start. I have googled a lot about noise reduction techniques, code examples and found nothing. You guys are my last hope.
Thanks in advance.
Have a look at this article.
Long story short, it uses MediaRecorder.AudioSource.VOICE_RECOGNITION instead of AudioSource.MIC, which gave me really good results and noise in the background did reduce very much.
The great thing about this solution is, it can be used with both AudioRecord and MediaRecorder class.
For audio capture you can use the AudioRecord class. This lets you record raw audio, i.e. you are not restricted to "narrow band" and you can also measure the volume.
Many smartphones have two microphones, one is the MIC you are using, the other one is near camera for video shooting, called CAMCORDER. You can get data from both of them to do noise reduction. There are many papers talking about audio noise reduction with multiple microphones.
Ref: http://developer.android.com/reference/android/media/MediaRecorder.AudioSource.html
https://www.google.com/search?q=noise+reduction+algorithm+with+two+mic
I have searched for this online, but am still a bit confused (as I'm sure others will be if they think of something like this). I'd like to preface by saying that this is not for homework and/or profit.
I wanted to create an app that could listen to your microwave as you prepare popcorn. It would work by sounding an alarm when there's a certain time interval between pops (say 5-6 seconds). Again, this is simply a project to keep me occupied - not for a class.
Either way, I'm having trouble trying to figure out how to analyze the audio intake in real-time. That is, I need a way to log the time when a "pop" occurs. So that you guys don't think I didn't do any research into the matter, I've checked out this SO question and have extensively searched the AudioRecord function list.
I'm thinking that I will probably have to do something with one of the versions of read() and then compare the recorded audio every 2 seconds or so to the recorded audio of a "pop" (i.e. if 70% or more of the byte[] audioData array is the same as that of a popping sound, then log the time). Can anyone with Android audio input experience let me know if I'm at least on the right track? This is not a question of me wanting you to code anything for me, but a question as to whether I'm on the correct track, and, if not, which direction I should head instead.
I think I have an easier way.
You could use the MediaRecorder 's getMaxAmplitude method.
Anytime your recorder detects a big jump in amplitude, you have detected a corn pop!
Check out this code (ignore the playback part): Playing back sound coming from microphone in real-time
Basically the idea is that you will have to take the value of each 16-bit sample (which corresponds to the value of the wave at that time). Using the sampling rate, you can calculate the time between peaks in volume. I think that might accomplish what you want.
this may be a bit overkill, but there is a framework from MIT media labs called funf: http://code.google.com/p/funf-open-sensing-framework/
They already created classes for audio input and some analysis (FFT and the like), also saving to files or uploading is implemented as far as I've seen, and they handle most of the sensors available on the phone.
You can also get inspired from the code they wrote, which I think is pretty good.
I was looking at Android's SoundPool as a mechanism to implement sound effects in my generic game development library. It seemed ideal.
But a little bit of research indicates that there all kinds of bugs in SoundPool. Are the bugs in SoundPool still relevant?
Because I'm developing a library, any bugs in SoundPool become bugs in my library, and I want to insulate my users from that.
So my question is basically: what API should I use for audio?
Using AudioTrack and writing my own mixer is not out of the question. But obviously it would be preferable to avoid doing that. And is there any API to provide decoding for me?
I need to be able to play a reasonable number of simultaneous sound effects (at least 16, let's say), and have even more open. Sounds need to start playing with low latency. WAV files need to be supported (MP3/Ogg is unimportant). Sound effects need to support seamless looping and dynamic, individual volume adjustment. The Android app lifecycle needs to be properly supported.
I have heard there is a 1MB limit somewhere for SoundPool, this is probably acceptable for each individual sound effect but not for all buffers/sounds. Can someone tell me exactly what the limit is on?
Finally, I need to be able to play background music as well, in compressed formats, with low CPU load. I assume MediaPlayer is ideal for this. Can it be used in parallel with another API?
I know a few people have been using MediaPlayer to fill in for SoundPool. But does it support the features that I need?
Are there any other audio APIs I've missed?
Just to add some more recent feedback on this issue. I've been using SoundPool for some time in an app with a fairly large user base for key press sounds. Our use case:
Must be played immediately
Up to 3+ sounds in parallel
We make use of the setRate across it's full range [0.5f-2.0f]
I've now experienced two major device specific issue and have decided to cut my losses and switch away from SoundPool
A large number of 4.4 LG devices (mostly the LG G2/G3 line) were having a native crash with their implementation of SoundPool. This was fixed in an update (eventually) but we still have a lot of users with un-upgraded devices
Sony Xperia devices currently have all sorts of issue with SoundPool as reported by others. In my case, I've discovered that if you use setRate with rate > 1.0f the SoundPool with start throwing exceptions until your app quits (and burn through a bunch of battery in the process).
TL;DR; I no longer think it's worth the danger/hassle of debugging SoundPool
Stick with OGG files and SoundPool will do you just fine. It's the nature of the multi-platform beast that is Android that there WILL be hardware configurations that will not work with every significant program, no matter how diligently the programmers try.
If this is a large and well-funded project, add to the funding one of each major phone for testing. It's actually much cheaper than the programmer time spent researching and trying to guess what their performance is.
Sorry. Seems as if this isn't the answer that you were looking for. Good luck!
DISCLAIMER: I have a small amount of experience with MediaPlayer, and no successful experience with the other APIs I mention, and the following information is based on what I've read in the DOCs and what I've read from google searches.
You could use mediaplayer (for the background music) with other audio APIs, since MediaPlayer automatically runs on it's own thread, but I believe it has a high-ish cpu load, and I don't think it would take compressed bits very well, but I'm not too sure.
There's also JetPlayer http://developer.android.com/reference/android/media/JetPlayer.html which seems like a lot of work to use effectively, but it would work very well with playing background music, then playing other sounds as needed in the game. From what I read of the DOCs, it takes a MIDI file (I think?) and you mute and unmute tracks to make it work how you want it to.
I like AudioTrack because it gives you the ability to edit sounds at runtime by changing the frequencies of the sound, and SoundPool can do the same.
Though for your situation, AudioTrack doesn't seem like it would work well, since playing two sounds would require two threads because AudioTrack is blocking (I'm pretty sure).
And with SoundPool, I'm thinking that since you have 16 sounds, maybe take two threads with one SoundPool in each thread and apply 8 sounds to each SoundPool. I don't really know though, as I've never even tried using SoundPool.
And again, my information is not based on experience, just what it appears from what I've read, so I may be completely or maybe just slightly wrong, or heck, who knows.
And I don't really know anything about the SoundPool bugs, since I haven't researched it.
I am aware that SoundPool was intended to handle small fx like sounds and I made sure my 4 sound clips which I want to play one by one in some sequence are small enough.
I used ogg quality 0 and clips are 35kb, 14kb, 21kb and 23kb totaling 92kb of compressed audio. I have no idea how to estimate what the uncompressed size would be, but it should not be a lot, right?
So when I play 4 sounds in sequence, it works well for first 9 times (9 sequences x 4 sounds) but starts to cause memory issues on the nines sequence for one of the sounds. It is always sequence 9 when I start to see error.
What is the best way to handle that? I have a few ideas:
1) compress sounds even more (ogg quality -1 and mono instead of stereo)
2) unload and load sounds constantly using SoundPool.load and SoundPool.unload
3) release and recreate soundPool instance time from time
Is there anything else I can do? It is embarrassing that android api cannot handle so small clips. I wonder how people create games with a lot of sound effects...
Errors look like that:
ERROR/AudioFlinger(35): not enough memory for AudioTrack size=1048640
DEBUG/MemoryDealer(35): AudioTrack (0x25018,size=1048576)
It seems I was able to resolve my issue after:
1) downsampled my sound clips from 44 to 22khz. Uncompressed size was cut in half (I figured how to estimate uncompressed sound size - export your clip to uncompressed wav). I used nice open source tool Audacity
2) trim the sounds more to reduce duration
3) put try/catch around play() just in case (it does catch errors when it try to play the sound but cannot)
That seems odd. SoundPool should be expanding your audio clips into memory when it loads them, and once loaded I wouldn't expect you to run into memory issues later. When I've run into memory issues it was right at the beginning, not later on. You sure you're not loading more sounds later on?
To answer your other question, some games use JET Player and MIDI sounds instead of SoundPool. The JetBoy sample program that comes in the Android SDK is a great example of how to do this type of sound work in an app.