I have a encrypted file that packed many mp3 sound tracks, I want to decrypt it while my cocos2d-x game is running and play one specific mp3 as background music without any temp files are generated, but, the api in cocos2d-x can only Accept exsiting .mp3 file on disk, so how do I do to make cocos2d-x read mp3 data that I decrypted in memory?
If you are not playing from a filepath, you will have to edit the cocos2d-x audio engine source codes.
For example for iOS, the bgm is loaded through the -(void) load:(NSString*) filePath function of CDLongAudioSource in CDAudioManager.m
NSString *path = [CDUtilities fullPathFromRelativePath:audioSourceFilePath];
audioSourcePlayer = [(AVAudioPlayer*)[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:&error];
It is calling initWithContentsOfURL for the AVAudioPlayer object. You can either modify this or create a separate function that uses initWithData to suit your needs.
As for Android, it seems more complicated. I have traced it down to this createUrlAudioPlayer(const AudioPlayerProvider::AudioFileInfo &info) function in AudioPlayerProvider.cpp that is performing the loading from filepath.
EDIT:
Another suggestion is not to use their audio engine but write your own class that will implement native platform functions to play audio from data.
Related
HiI'm trying to create a "Speech to text" app that can transcribe any audio/video file. I've created an app based on this post and it works great for WAV files. But if I use an MP3 file, the line hr = cpInputStream->BindToFile(wInputFileName.c_str(), SPFM_OPEN_READONLY, &sInputFormat.FormatId(), sInputFormat.WaveFormatExPtr(), SPFEI_ALL_EVENTS); returns
The Parameter is incorrect
The question is, can I use MP3 files as input for SAPI? and if yes, how do I determine the correct format for the call to hr = sInputFormat.AssignFormat(SPSF_16kHz16BitStereo) because SPSF_16kHz16BitStereo will certainly not be correct and I don't think we should hardcode it.
I am using a nodeJS library naudio —link— to record sound from a 2 microphones (total 4 channel audio with each microphone being stereo). This library spits out a .raw file in the following specs: 16 BIT, 48000Hz Sample Rate, Channel Count 4
// var portAudio = require('../index.js');
var portAudio = require('naudiodon');
var fs = require('fs');
//Create a new instance of Audio Input, which is a ReadableStream
var ai = new portAudio.AudioInput({
channelCount: 4,
sampleFormat: portAudio.SampleFormat16Bit,
sampleRate: 48000,
deviceId: 13
});
ai.on('error', console.error);
//Create a write stream to write out to a raw audio file
var ws = fs.createWriteStream('rawAudio_final.raw');
//Start streaming
ai.pipe(ws);
ai.start();
process.once('SIGINT', ai.quit);
Instead of the .raw file, I am trying to convert this to two individual .wav files. With the above encoding and information, what would be the best way to do so? I tried to dig around for easy ways to deinterleaving and getting .wav but seem to be hitting a wall.
The addon is a wrapper around a C++ library called portaudio which according to its documentation supports writing to a WAV file.
What you could do is extend the addon and bind a NodeJS function to the underlying C++ function that write to WAV.
This will give you a good performance if it is an issue.
If you want something easier you could look up utilities that do the conversion and call them from within your script using ex like this
Look similar to this question.
You may also take a look here to know how to create wav file from javascript.
Thanks for taking some time to read my question.
I'm developping a C++ application using Qt and windows API.
I'm recording the microphone output in small 10s audio files in raw format, and I want to convert them to aac format.
I have tried to read as many things as I could, and thought it would be a great idea to start from windows media foundation transcode API.
Problem is, I can't seem to use a .raw or .pcm file in the "CreateObjectFromUrl" function, and so I'm pretty much stuck here for the moment. It keeps on failing. The hr return code equals 3222091460. I have tried to pass an .mp3 file to the function and of course it works, so no url-human-failure involved.
MF_OBJECT_TYPE ObjectType = MF_OBJECT_INVALID;
IMFSourceResolver* pSourceResolver = NULL;
IUnknown* pUnkSource = NULL;
// Create the source resolver.
hr = MFCreateSourceResolver(&pSourceResolver);
if (FAILED(hr))
{
qDebug() << "Failed !";
}
// Use the source resolver to create the media source.
hr = pSourceResolver->CreateObjectFromURL(
sURL, // URL of the source.
MF_RESOLUTION_MEDIASOURCE, // Create a source object.
NULL, // Optional property store.
&ObjectType, // Receives the created object type.
&pUnkSource // Receives a pointer to the media source.
);
The MFCreateSourceResolver works fine, but CreateObjectFromURL does not succeed :(
So I have two questions for you folks :
Is it possible to encode raw audio files to aac files using windows media foundation ?
If yes, what should I read to accomplish what I want ?
I want to point out that I can't just use ffmpeg or libav because I can't afford any license for my software, and don't want it to be under the GPL license. But if there are alternatives to windows media foundations to encode raw audio files to aac, I would be glad to hear them.
And finally, sorry for my bad english, this is obviously not my native language and I'm sorry if I made your eyes bleed. (and happy if I made you laugh)
Have a nice day
The hr return code equals 3222091460
Those are HRESULT codes. Use this "ShowHresult" tool to have them conveniently decoded for you. The code means 0xC00D36C4 MF_E_UNSUPPORTED_BYTESTREAM_TYPE "The byte stream type of the given URL is unsupported."
The problem is basically that there is no support for these raw files, .WAV is a good source for raw audio - the file holds both format descriptor and the payload.
You can obviously read data from the raw audio file yourself and compress into AAC using Media Foundation's AAC Encoder via its IMFTransform interface. This is reasonably easy and you have AAC data on the output to e.g. write into raw .AAC.
Alternate options to Media Foundation is DirectShow (there are suitable codecs, though I thought it might be not so easy to start), libfaac, FFmpeg's libavcodec (available under LGPL, not GPL).
Below is the code I am using to play a video
QFile* file =new QFile(“C:\\Video\\test.avi”);
media->setCurrentSource(Phonon::MediaSource(file));
media->play();
Using this code the playback fails -what I see is the play bar at the bottom but the video never starts.
If I change the code to the following everything works as expected
media->setCurrentSource(Phonon::MediaSource(“C:\\Video\\test.avi”));
media->play();
Are there additional initialization steps required when using an iodevice? Ultimately my code will be using a custom iodevice which is not working as well.
This is an old post, but I wanted to clear up any confusion out in case it will help someone in the future.
QT does allow you to pass Phonon::MediaSource() a QIODevice. We successfully deployed our solution by creating our own subclass of QIODevice.
The reason it was not working for me was QT was having an issue with the codec I was using. When you use the QIO device you don't get the same format support as you would if you pass a string.
One other thing to note, while this solution works great on windows. On a mac when using the QIO device the entire file will be loaded into memory before it plays. In my case this was a deal breaker. Having an encrypted file is usless if the first thing you do is de-crypt the entire file and load it into memory.
From the Phonon::MediaSource documentation:
Warning: On Windows, we only support QIODevices containing the avi,
mp3, or mpg formats. Use the constructor that takes a file name to
open files (the Qt backend does not use a QFile internally).
I think that the last line should answer your question. Instead of a QFile, you can use a QString, or call the function QFile::fileName like this:
QFile* file =new QFile(“C:\\Video\\test.avi”);
media->setCurrentSource(Phonon::MediaSource(file->fileName()));
media->play();
If you take a careful look in the [Phonon Module docu][1], you will see that MediaSource cannot be constructed with QFile*.
By the way I don't see in your code any phonon paths. At least you should create audio sink and connect it with the mediaobject:
Phonon::AudioOutput *audioOut = new PhononAudioOutpu(Phonon::MusicCategory);//or the category you need
Phonon::createPath(mediaObject, audioOutput);
mediaObject->play();
Works fine with QFile
I am using CFtpConnection class for creating my FTPClient Library using MFC.
I am using GetFile to download file from Server.
MY requirement is like if i am downloading 100 MB video from server when 50-60 MB video is downloaded and in between if i play that while it should play upto that particular location what it has downloaded uptil that time .
Is that way i can do it any additional parameters i need to pass or something like that?
My FTPlibrary download method is as follows:
CFtpConnection* m_pConnect;
bool CFTPClient::Download(LPCTSTR pstrRemoteFile, LPCTSTR pstrLocalFile,
DWORD dwFlags)
{
m_pConnect->GetFile(pstrRemoteFile,pstrLocalFile,dwFlags);
return true;
}
And while calling in my application i am doing like this :
CFTPClient m_objftpclient ;
m_objftpclient.Download("MVI_2884_1.avi","D:\\MVI_2884_1.avi",FTP_TRANSFER_TYPE_BINARY);
You can't do that easily or even do it at all. The GetFile method of CFtpConnection is blocking which means it will exit only when the file is downloaded. So even if you thread it, the only way you can monitor the download is to get the size of the file on disk.
If you're about to implement video streaming, you should go down a level and work at the socket level. If you really want to use CFtpConnection, you should use the method OpenFile which returns a CInternetFile which can be read by chunks allowing you to monitor the download and share the buffer in which the file is downloaded for playback.