I am using using libsndfile read and manipulate sound files.
I am wandering what whould be the simplest way to play these sounds in real time
in Qt.
I have tried to look at QtAudioOutput, however has to use QtAudioFormat, which is
quite cumbersome to configure.
All i want is to
1) Read data using libsndfile
2) manipulate the data
3) send the buffer to an audio output library.
Try using sndfile-play to play audio files using libsndfile, or a library based on libsndfile such as ALURE
http://kcat.strangesoft.net/alure.html
You could load the audio file yourself manually using libsndfile and play the file using OpenAL. This is essentially what ALURE is doing.
You can output the audio using Phonon. Here is the piece of the docu from Phonon module
The type of the source is set by the media source itself, and is dependent on the constructor used to create it. Note that it is possible to provide data from any source by implementing a QIODevice.
Basically you have to implement your own QIODevice if you want to do it Qt way
Related
I'm trying to understand how I can generate a waveform from an audio (or video) file to display to the user.
I've been googling around for quite a while now and can't determine if this is even possible in Qt without using something like FFmpeg. I've seen all of these classes: QMediaPlayer, QMediaContent, QMediaResource, QAudioProbe and experimented with the Qt Media Player Example but am just not seeing where I can access the actual audio buffer.
So I have 2 questions:
Is what I want to do even possible without 3rd party libraries?
If it is possible, can some kind soul outline what I need to read and understand in order to access the audio data
I have tried the suggestions from this question (Audio visualization with QMediaPlayer) but the result of audioProbe->setSource(player) is always false and the method processBuffer never gets called.
audioProbe = new QAudioProbe(this);
bool success = audioProbe->setSource(player);
qDebug() << success;
connect(audioProbe, SIGNAL(audioBufferProbed(QAudioBuffer)), this, SLOT(processBuffer(QAudioBuffer)));
Update: Adding some additional detail in the hope of clarifying things.
For testing/learning I am using the Media Player Example which ships with Qt, so it is set up correctly with Q_OBJECT etc.
For audio, I tested with both .mp3 and .wav files. FWIW, the player example won't play video for some reason (.mp4, .avi were tested)
The player in the code is QMediaPlayer – which inherits from QMediaObject. The example code for the Player class is here. I added my code (in original comment above) right after the player is instantiated. I also tried adding it once media is loaded.
I tried declaring my slot first as private, then as public – either way, it is never called.
Frustrating that such a simple thing is so hard.
Going the "no external library" route will likely just lead to more of a headache and more work than is necessary. The other advantage of going with an established library is you won't be bound to one file format, as not all formats store their data the same way. If the audio format is uncompressed (wav or other) you can read the header until you get to the data chunk. An answer to this question here details this in C. You should be able to get an idea for the file format from this to apply it to another language.
You will want to understand how many channels are in the wav file, bit depth, and also the sampling rate before you can do anything worthwhile with the data. All this info can be grabbed from the header.
It turns out that QAudioProbe is not supported on OSX – the platform I am working on. Took quite a while (a "Qt while. . .") to ferret that info out so I am posting it here explicitly.
See this document for full details: Qt 5.5.0 Multimedia Backends
i'm trying to perform simple automated test on my Adobe Air Application. I decide to store some test data in external MS Excel file to decrease hardcode quantity. But I discovered, that this is not a trivial task, when you're working with Adobe Air.
Actually, i need to correct initialize an excelApp object, the rest part of logger is already implemented. I tried the following approach, but I can't find a package with ActiveXObject class defined.
excelApp = new ActiveXObject("Excel.Application");
Can you give some ideas, where can I find such a class?
There's a simple as3 xls reader library called as3xls that should provide the functionality you need. However, if you need to write out excel files, it's a little bit rough.
I need to parse a large trace file (up to 200-300 MB) in a Flex application. I started using JSON instead of XML hoping to avoid these problems, but it did not help much. When the file is bigger than 50MB, JSON decoder can't handle it (I am using the as3corelib).
I have doing some research and I found some options:
Try to split the file: I would really like to avoid this; I don't want to change the current format of the trace files and, in addition, it would be very uncomfortable to handle.
Use a database: I was thinking of writing the trace into a SQLite database and then reading from there, but that would force me to modify the program that creates the trace file.
From your experience, what do you think of these options? Are there better options?
The program that writes the trace file is in C++.
Using AMF will give you much smaller data sizes for transfer because it is a binary, not text format. That is the best option. But, you'll need some middleware to translate the C++ program's output into AMF data.
Check out James Ward's census application for more information about benchmarks when sharing data:
http://www.jamesward.com/census/
http://www.jamesward.com/2009/06/17/blazing-fast-data-transfer-in-flex/
Maybe you could parse the file into chunks, without splitting the file itself. That supposes some work on the as3 core lib Json parser, but it should be doable, I think.
I found this library which is a lot faster than the official one: https://github.com/mherkender/actionjson
I am using it now and works perfectly. It also has asynchronous decoder and encoder
I've got a bunch of audio files (let's say ogg or mp3), with metadata.
I wish to read their metadata into R so to create a data.frame with:
file name
file location
file artist
file album
etc
Any way you know of for doing that ?
You take an existing mp3 or ogg client, look at what library it uses and then write a binding for said library to R, using the existing client as guide for that side -- and something like Rcpp as a guide on the other side to show you how to connect C/C++ libraries to R.
No magic bullet.
A cheaper and less reliable way is to use a cmdline tool that does what you want and write little helper functions that use system() to run that tool over the file, re-reading the output in R. Not pretty, not reliable, but possibly less challenging.
Possible, yes, easy, no.
You "could" use a combination of readChar and/or readBin on the file and parse out the contents. This would be highly dependent, though, on parsing the frame tags from the raw bytes of the ID3v2 tag (and mind you it would change if it was a version 1 tag). If would certainly be a lot of work to implement a straight R solution. Take this Python code for example, it's very clean straight python code but a lot of branching and parsing.
You can use exiftool with system command available in R. Optionally, you can create regexp to handle the fields you need... If I were you, I'd stick with Dirk's advice (as usual) =)!
Out here in 2021, I wanted to do this so I did the following...
Create a new playlist while in 'songs' view.
Select all songs and drag to the new playlist. Highlight that playlist
File> Library>Export Playlist. My default file was to save as .txt, if not, designate.
Open Excel to save as csv or read.delim() in r as the txt file is tab-separated
import to R
I'm trying to use Adobe's OggVorbis library. But I can't seem to get the Sound object to loop.
I even tried looping the _sound object inside the AudioDecoder.as in the "com.automatastudios.audio.audiodecoder" package.
Do you really have to reload the file and stream it over and over?
If you're streaming, then yes you'll have to jump back the beginning of the stream. A stream, by definition, is a constant link to the server and does minimal loading of files locally.
But, if you're not really streaming, you should have no problem loading up a file and caching it locally then playing it over and over.
Since you mention Alchemy, there may be other unknown issues if you're trying to use a converted C library, as opposed to native ActionScript.
[Note; I didn't know the OggVorbis library for Flex before now].