A few years back, I wrote some util library around DShow/DSound to let me play MP3s in a Windows C++ application. Is that still the normal way to do it in a C++/MFC app, or is that an area of DirectX that has been subsumed into the general Windows APIs?
The motivation is simply we use the standard Windows PlaySound method for WAVs, and would like to be able to play MP3s using a similarly simple API, either provided by Windows or something we write to wrap more complex functionality.
EDIT: this is for a large, commercial, closed-source project. And we only want to play things simply, paying a lot for a library won't fly.
PlaySound() natively supports MP3 as long as it is embedded in a WAV file.
People don't realize that WAV is a container format.
Download the ffmpeg utilities to convert the header and preserve the codec:
ffmpeg -i input.mp3 -c copy -f wav embedded_mp3.wav
You can either use DirectShow but it's not part of DirectX anymore or rely on a third-party library like Bass, FMod, mpg123 or even libwmp3.
If you don't want to use DirectShow anymore (but why change if your existing code keeps working?), you can use MCI:
mciSendString("open la_chenille.mp3 type mpegvideo alias song1", NULL, 0, 0);
mciSendString("play song1", NULL, 0, 0);
mciSendString("close song1", NULL, 0, 0);
IGraphBuilder::RenderFile is an easy way to play any audio file.
Youc could use MCI windows functions,
https://msdn.microsoft.com/en-us/library/ms709626
It can play many of audio file formats including MP3, WAV, MIDI etc.
If I recall correctly it does not require DirectX.
The PlaySound function might also work for you.
If you don't want to pay any licence and wanna do in-house, do the parsing of your mp3 file and pass it to XAudio2.
Its a thing that you can do once (2-3 hours at max) and use always. :P
You could have a look at BASS. It's a simple to use audio library, free for noncommercial use.
Related
I have a question about playing .wav files in C++. I search for a method to play it on Linux and Windows and I want to break up it while a function has been ended, did you know a sample for this?
Thanks for any Answer :)!
I am from Germany, please don't be angry about my grammar or spelling :D.
There are several ways to do this.
The simplest, ugliest and most hackish way to do this is to write directly to your sound device located in /dev/snd/. However this really depends on your linux distribution and some modern ones no longer allows you to do this. In general, direct read / write to /dev/ devices is fading away. Here is a question answering this.
For a very long time, the "official" way was to use alsa library which uses a C style interface. It is usually pretty tricky, full of pitfalls and "workarounds" that depends on your exact audio hardware.
I think that alsa is getting gradually replaced by jack, which I hope is easier to use. I don't have any experience with this one.
The last method I know is with portaudio, which as the name implies, is somewhat portable between at least windows, linux and osx.
All of these library will allow you to control your audio hardware, init / setup / play. What is simple about wav files is that the content is raw pcm, which is usually the format used by those libraries. The wav format is usually like this :
[wav header] [audio data until the end of the file]
If you can manage a few milliseconds of garbage when you start playing, you can "safely" send the header for playback as well and avoid parsing it. You will have to understand PCM formats however and should bring some additional readings.
As an added "trick" which doesn't directly concern C++, I strongly suggest using Audacity. It is a small program which I see like the notepad / vim of audio files. It is small, it loads fast, allows you to copy / paste segments of audio and detect pcm formats. (Actually, you just change pcm settings until the audio "sounds" right, but still useful)
The title mentioned Linux, but then you mentioned Windows and Linux in the post.
For Linux, best is to use gstreamer if you insist on C++. Look through the gstreamer code for gst-launch. It is run as below in a Linux machine.
$ gst-launch filesrc location="location of file" ! wavparse ! alsasink
From, http://sanchayanmaity.github.io/beagleboard-xm/linux/programming/software/2014/08/07/playing-wavmp3-file-using-gstreamer-in-code.html
For windows, or if you want to use OS agnostic code on both Windows and Linux, you can use SDL,
http://lazyfoo.net/SDL_tutorials/lesson11/
Another alternative (cross-platform, Object oriented), is SFML. Check the audio wav file playback example at,
http://www.sfml-dev.org/tutorials/2.0/audio-sounds.php
I've been working on a C++ application that plays a different video depending on what a user did. So Far I've worked with OpenCV to only play the video, since I need to do some changes on the video on real time. Since OpenCV doesn't play sound, I need to use a different library to play the sound independently. So far I tried FFMPEG, but I couldn't get it work, since I'm not experienced on working on C++ (At least in a computer with external libraries, I normally programm it on microcontrollers).
The sound doesn't need to sync with the video and it also only should run on Windows. What is my best option?
Thanks.
If it does only need to work for Windows, the WINAPI function PlaySound should fulfil your requirements.
#include "windows.h"
#include "mmsystem.h"
void play()
{
PlaySound(TEXT("sound.wav"), NULL, SND_FILENAME); // plays sound.wav once
}
To make this work you need to link against winmm.lib.
If your file name is not ASCII you should use PlaySoundA (ANSI) or PlaySoundW (unicode) instead.
See also this question if the sound is not played properly.
I want to process audio online/live where I constantly read audio samples from an audio file, process these (e.g. apply some effect), and forward the processed samples to an audio output device like a soundcard. The input files have common formats such as wav, mp3, perhaps even ogg.
Is there a library available similar to libav/ffmpeg for audio files which simplifies reading various audio formats and provides me a constant stream of raw audio samples? Or is the best solution to use individual libraries for each format?
The libraries should be c/c++ and cross-plattform compatible (Mac, Win, Linux, ARM).
EDIT Thanks for all answers. I have evaluated all libraries and came to the conclusion that it is best to just stick with libav/ffmpeg because most of the libraries require ffmpeg as a backend.
Check out Juce. It is a big library that has been used to develop VST audio plug-ins for music software. There is a lot of stuff you don't need in there, but I think you can pick and choose only the audio parts to include in your build. The AudioFormatReader and its associated classes can do the file reading, and there are also classes for outputting to the sound card. There's a lot of other audio processing tools as well. It's GPL licensed, cross platform, and they claim experimental Android support. I haven't used it for a project yet, but I am waiting for the inspiration!
I can recommend RtAudio or PortAudio for cross-platform audio I/O. For audio decoding you might want to have a look at libsndfile or libaudiodecoder.
I'd check out libSDL, it has an audio subsystem that is built for doing things like that and handles ogg,mp3,flac,wav, etc..
LibVLC can do this. libvlc supports the most various audio (and video) formats. It's a C/C++ crossplatform library. It should also support Arm code generation.
You can use irrKlang library. I have used it for my games. It is very simple library to use, for example to play some file "somefile.mp3" you just need to write
engine->play2D("somefile.mp3", true);
And this library is cross-platform, too. And works with C++, C# and all .NET languages.
More features of this library (from its own site)
It has all the features known from low level audio libraries as well
as lots of useful features like a sophisticated streaming engine,
extendable audio reading, single and multithreading modes, 3d audio
emulation for low end hardware, a plugin system, multiple rolloff
models and more. All this can be accessed via an extremely simple API.
GAudio Library maybe is one you persuit.
It is simple, powerfull, cross-platform and extendable
The hello world of GAudio like this:
gaudio_init("addons");
const char* filename = "..\\media\\trek12.wav";
gsource* source = gaudio_source_create_from_file(filename,FALSE);
if(source == NULL)
{
printf("load file:%s error!\n",filename);
printf("bad source.\nerror code:%d.\n",gaudio_error_get());
gaudio_deinit();
return -1;
}
printf("play filename:%s\n",filename);
gaudio_source_play(source,FALSE);
printf("\nplaying, press any key to quit.\n");
getch();
gaudio_source_stop(source);
gaudio_source_destroy(source);
gaudio_deinit();
I have an SDL app, that works under Linux, Mac and Windows. It's something like a media player, and can play audio just fine. I'd like to add audio recording feature to it, but I'd like to encode it in real time to MP3. Can anyone point me to an example how can I use LibLame, LibSoX, or possibly some other library to achieve this?
-- OR --
I'm also willing to rewrite the whole thing into something easier to manage than C++. I've looked at Kivy and Love2d which uses Lua, but audio recording it's still an issue there. If you know ANY toolkit that:
is cross platform
helps you build GUI using your own graphics
can play AND record mp3 files
ideally can operate under framebuffer (no X Window server under Linux)
Please let me know. I'm looking at Python + Pygame + Pyaudio, it can do graphics and output sound, but still can't record MP3's, only WAV's. Any way to integrate LAME into this to make it work?
FMOD can play practically anything, and handle audio input as well, although I don't know if integrating an entire audio engine is a bit overkill for your project.
It's free for non-commercial usage.
As for encoding, LAME is definitely the de-facto choice for MP3.
There's a very simple library called lame_enc.dll which wraps LAME's capabilities in a simple API. It's Windows only, but you could look at it's source for a good reference on how to use LAME.
I'm writing a cross-platform Qt-based program that from time to time needs to play back audio supplied externally (outside my control) as raw PCM. The exact format is 16 bit little-endian PCM at various common sample rates.
My first obvious idea was to use Qt's own Phonon for audio playback, but there are two problems with this approach:
As far as I can see, Phonon does not support headerless PCM data. I would have to hack around this and fake a WAV header every time playback starts. Not a showstopper, though.
More importantly: There doesn't seem to be any way to control how Phonon (and its backends such as xine, PulseAudio, DirectX, whatever) prebuffers. Its default behaviour seems to be something like 5 seconds of prebuffering, which is way too much for me. I'd prefer about 1 second, and I'd definitely like to be able to control this!
I'm currently looking at Gstreamer, FFMPEG and libvlc. Any thoughts? Since my audio is in a very simple format and I don't need to do fancy mixing stuff (just volume control), I'd like a simple, free (as in freedom), cross-platform and widely available library.
Qt 4.6 has the new QtMultimedia module.
https://doc.qt.io/archives/4.6/qtmultimedia.html
The QAudioOutput class would seem to do what you want - it just plays raw PCM data.
ffmpeg, libvlc and gstreamer have abilities beyond raw pcm, such as codec support.
For your purposes, SDL (example 1, example 2), OpenAL, QAudioOutput are sufficient. SDL is probably the most popular option.
Also, why do you want to control buffering? Buffering a lot means less interrupts and lower power consumption.
Have you looked at OpenAL?