I have to play a single channel from an ogg file under Android. After a lot of searching, I think I found out a strategy.
Using OpenSL, I decode the file to PCM using something like this.
Then I should copy the selected channel from the PCM buffer into another buffer linked to the OutputMix, using something like this.
Is this the best option? Is there something already available to look at?
Thank you.
I solved using non-native code based on OpenMXPlayer.
Update: The general idea is to use MediaCodec to decode the file to memory, than modify the data in memory, than send it to an AudioTrack.
Related
I'd like to build something a little like the LiveEffect sample in the Google/Oboe library.
However, I want to affect the audio selected from a file chosen by the user rather than the microphone input. I do not need the input.
There's no example in the Google/Oboe repo of how to operate on a file.
Does anyone have an example or guidance so I can let the user choose a file from their local storage, then (and this is the bit I'm missing) pass the audio across the JNI bridge to my oboe app?
I do the need low-latency capability of Oboe as I'm going to affect the audio in response to motion data.
Any guidance gratefully recieved.
For anyone passing by in search of a similar solution, here's how I solved this
On the Java/Kotlin side pick up the audio file (i used a WAV in this
case) with an intent
Use a contentResolver on the audio file to create an inputStream
read the data from the inputStream into a byteArray
pass the byteArray over the JNI bridge to the native code
wrap the byteArray in a MemInputStream from the PARSELIB example
wrap the MemInputStream in a WavStreamReader, also from the PARSELIB example
create a SampleBuffer, from the IOLIB example, and load the WavStreamReader into it
create SampleSource, from the IOLIB example, and give it the SampleBuffer
give the SampleSource and SampleBuffer to the SimplePlayer from the IOLIB example
do the processing in the SampleSource's mixAudio() method bearing in mind all the rules for real-time processing in Oboe.
I also needed to do this on the block because I have a fixed window operation. to do this, I adapted the SampleBuffer class to add a method that would pull block data into mixAudio(), but that's only specific to some cases.
I hope that helps someone in the future.
I have seen many questions related to this. Nevertheless there is not an answer for mine I think.
I would like to use an already coded RTSP Client on Android to use with MediaCodec in order to capture a RTSP stream in H264 to then decode and display it. I have used VideoView and MediaPlayer which are well-known to support RTSP streaming in the .setDataSource method (file or rtsp/http path) (unlike MediaExtractor which only supports file or http), but the latency is to high for my purposes.
I would like to use MediaExtractor, but because of that limitation on the setDataSource method it seems to be not an option. Given this, I am searching for some help or examples (tutorial?) that I could use as RTSP Client on Android, or if someone has used MediaExtractor in some way to capture the RTSP stream its help is more than welcome as well.
Thank you so much guys!
rojiark
You can try https://github.com/fyhertz/libstreaming
You should know though that is LGPL, which means the rest of your project will become LGPL and if you distribute the application you must also distribute the source code if requested.
I am trying to stream incoming AMR_NB. I can't use MediaPlayer directly because it requires a seekable file. I would like to use MediaCodec, but to use MediaCodec I need (I think... please correct me!) MediaExtractor to give me things like the presentationTime. Is that true? Can I use MediaCodec without MediaExtractor?
MediaExtractor seems to require seekable files. The documentation only specifically says so for one of the setDataSource operations but when I tried to use any of the others it failed due to failed seek attempts.
So, what can I do to get my incoming AMR stream to play? I am aware of a scheme where by you save incoming data to a file and periodically make a copy of that file to feed to MediaPlayer but I'd really prefer to find a real honest streaming solution.
Is it possible to use MediaCodec without using MediaExtractor? If so how do I find presentation time and the string to pass to MediaCodec.createDecoderByType? The documentation SAYS that "audio/3gpp" is what I want but when I attempt to use that I get the following error:
codec = MediaCodec.createDecoderByType("audio/3gpp");
01-02 03:59:36.980: E/OMXMaster(21605): A component of name 'OMX.qcom.audio.decoder.aac' already exists, ignoring this one.
So I'm not sure how to get at MediaCodec either.
"I can't use MediaPlayer directly because it requires a seekable file" This is not generally true. I would like you to try it on your stream and report exactly what happens.
"Can I use MediaCodec without MediaExtractor?" I doubt it: I believe they are designed to be used together.
I have used these components to play streams. However, the MediaExtractor has limitations that are not documented ( as far as I know ). So use a little proxy server to feed it things it can digest. And I have 1 thread to run the MediaExtractor and another to take output from the the MediaCodec. Then i have to avoid deadlocks and cope with snchronization. But it is not that bad provided you just want to play forwards only. Then you have only the problem of how to stop!
I advise that you try MediaPlayer first. Otherwise, if you are keen enough to try the MediaExtractor, we could share our discoveries about what it will and wont digest. Don't take anything for granted. For example it seems it will play my MP3 files, but cannot discover their duration, or seek on them!
I have audio data in uncompressed raw format as java arrays.
short[] or byte[]
I wish to play them in Android.
SoundPool looks like what I need but I can't find way to load data from memory. It loads from files only and I am not sure how do I specify format that it is raw data.
The AudioTrack class has write(...) methods which accept byte[] or short[] as the source parameters. It's the only Android sound class I know of that does.
A library called libpd can give you many audio features lacking in Android. The learning curve is a bit steep, but it's powerful stuff.
My application want to read each frame from a movie file and used to other.
How can I do it?
I don't know that there's a way to do this using the built-in decoders (but would be glad to hear of it if there is). You might have to make something based on modifications to the alternative software decoder included in the android sources, or an external alternative such as ffmpeg or mplayer.