I'm using two mediaPlayer instances to send two different streams to each channel. Half the time, it works fine but sometimes there is lag between left and right channel which is clearly audible. Is there any alternative except soundpool in Android to play multiple audio files simultaneously with sync ? soundpool is not suitable for my application since audiofiles are large (approx 20 MiB each). Audiofile format in question is : FLAC.
I found that there're no proper in-built mixing capabilities provided by Android API. I ended up using wav files instead of FLAC and mixing them on the fly as needed. Here is a higher level description how I achieved it.
Read both wav files and saving data part in byte array (Don't forget to strip out header bytes)
Mix them byte by byte to generate a unified wav file
In my use case, I just needed to mix left and right channels, but one can do all sorts of transformations as needed.
Create a temporary file to hold mixed data
Play the temporary file with mediaPlayer
One can also use audioTrack to play without storing resulting byte array to temporary file but I chose to use mediaPlayer due to built in seekTo functionality.
Hope, this approach is helpful.
Related
I am developing and android app for text to speech conversion for regional language and for that I am storing basic audio sounds for pronunciation of words. Whenever a character is read from file the corresponding audio file is chosen for waveform generation. hence a single audio file is fetched so many times even for small amount of content in file. So here my question is where I'll have to store that audio files such that loading can be as fast as possible?
What folder its in won't make a real difference, so long as you unpack it from assets (assets being compressed). What would make a difference is storing it in memory vs on disk. Of course that depends on how many/how big of sound files you actually use- do you have one per word, one per phoneme, or something else? My guess is you'll actually have better performance if you combine multiple phonemes into a single waveform for playback rather than try to play each individually.
How can merge two sounds and save as a new file?. One sound is a loaded mp3 file and the other from the microphone. Then I need to upload this sound into a server. Is this possible?
This all can be done, but if you looking simple example with few methods to call, I'm afraid it's not so easy.
You can extract bytes from sound with Sound.extract(). This data is sound amplitude in 16-bit numbers, right and left channel interleaved. Use ByteArray.readShort() to get them.
Microphone data can be captured with SampleDataEvent.SAMPLE_DATA, see example here. To mix them with song, just add sound amplitudes and write result into third array. The result will be essentially WAV-format (without header,) unpacked sound data. You can upload it raw, or search for "as3 mp3 encoder" (google), but these things are rare and written by entusiasts, so maybe you can get them work. Also, to mix sounds correctly, frequencies of data from mic and sound file must be equal.
And upload part - if this was file on disk, this would be easy - FileReference.upload(). But there's only data in memory. So you can look into Socket class to send it.
I need to save a video file generated by two video streams coming from two different sources. I'm using rtsp over tcp/ip, and the videos are encoded with h264.
I need to first record the video from the first source and than continue with the second source.
So what I tried was to declare two AVFormatContext instances, initialize both with avformat_open_input(&context, "rtsp://......",NULL,&options)
and then read frames with av_read_frame(context,&packet)
and write them in the video file av_write_frame(oc,&packet);
It works fine saving the video from the first source, but if by example I saved y frames from the first context, when I try reading and saving the frames from the second context in the same file, for the first y frames I am tring to save, av_write_frame(oc,&packet2);
would retun -22, and would not add the frame to the file.
I think the problem is that the context variable remembers how many frames were read, and it gives every read packet an identification number, to make sure it isn't written twice. But when I'm using a new context those identification numbers reset, the AVOutputFormat or the AVFormatContext also retain the id of the package they are expecting to receive, and would not write anything until they receive a package with that id.
Now I'm wondering how could I solve this inconvenience. I can't find any setter for that id, or any way to reuse the same context. I thought to modify the ffmpeg sources but they are pretty complex and I couldn't find what I was looking for.
An alternative would be to save the two video in two different files but, I don't know how to append them afterwards, as ffmpeg can only append videos encoded with mpeg and rencoding the video isn't really an option, as it will take to much time. Also I couldn't find any other functional way to append two mp4 videos encoded with h264.
I'll be happy to hear any kind of usable ideea to this problem.
If you are saving raw h.264 streams why not simply store two seperate streams and then concatenate the file chunks on the command line seperately using a system command system("cat file1 file2 > finalfile")
If your output is one of the following you can append directly using cat
Transport stream [ts] with same codecs
.mpg files
raw h.264 files
raw mpeg4 files which have exactly same encoding headers [same dimensions, profile and toolsets mentioned in header]
H.263 streams
You cannot concatenate directly mp4 files or 3gpp files.
Is it possible to change the wave header in an existing mp3 file? I need it to be 8 or 16 bits per sample.
Any tips about how to proceed? I am totally stuck.
Thank you very much
Which wave header are you referring to? Are you talking about the frame header? In any case, it doesn't contain a bits-per-sample field because such information would be meaningless to store in the compressed mp3 stream.
It's up to the decoder/player to decide how many bits to use per sample when generating an uncompressed PCM stream from the mp3.
In case you mean that you've slapped a RIFF header onto an mp3 file to be able to import it into various old programs, you can find the format of that header here.
I need to implement a playback of separate audio files in N channels, files may play sequentially or in parallel. I need to implement it on Android.
Timeline:
|file a...|file d....|file b...|......|file k|....
|.....|file g|file c..|file p.|....
I'm thinking two options, one being FMOD to decompress files and play them simultaneously. I have researched and FMOD seems to fit well and much easier than manually playing this using an AudioTrack. However, I can't understand if FMOD would allow us to save the entire merged output without playing it through.
I know that using solution here we can redirect output to a wav file, but is it possible to just create a final output instantly and save it using FMOD? Or will I have to manually merge PCMS into one stream after all..
Thanks.
An important question here is why you need to save the files out, if it's possible to do this offline then it would be a lot simpler. If you must record the concatenation of several files (including others played in parallel), it is quite possible with FMOD.
One way would be to use wave-writer-nrt output mode, which allows you to output a wav file based on FMOD playsound calls in faster than realtime.
Another way is to use a custom DSP to access the data stream of any submix as it plays, useful if you want other sounds actually playing at the same time.
Another is simply create the sound objects, then use Sound::lock to access the PCM data, which you could concatenate yourself to a destination. Keep in mind all the sounds would need to be the same sample rate and channels, otherwise you would need to do processing. Also keep in mind you cannot do this for parallel sounds unless you want to mix the audio yourself.