How do you play or record audio (to .WAV) on Linux in C++?

Question

Hello, I've been looking for a way to and audio on a Linux (preferably Ubuntu) system. 和音频的方法。 I'm currently working on a front-end to a voice recognition toolkit that'll automate a few steps required to adapt a voice model for PocketSphinx and Julius .

Suggestions of alternative means of audio input/output are , as well as a fix to the . 提供音频输入/输出的替代方法的建议，以及对的的修复。

Here is the current code I've used so far to play a .WAV file:

void Engine::sayText ( const string OutputText ) {
    string audioUri = "temp.wav";
    string requestUri = this->getRequestUri( OPENMARY_PROCESS , OutputText.c_str( ) );
    int error , audioStream;
    pa_simple *pulseConnection;
    pa_sample_spec simpleSpecs;
    simpleSpecs.format = PA_SAMPLE_S16LE;
    simpleSpecs.rate = 44100;
    simpleSpecs.channels = 2;

    eprintf( E_MESSAGE , "Generating audio for '%s' from '%s'..." , OutputText.c_str( ) , requestUri.c_str( ) );
    FILE* audio = this->getHttpFile( requestUri , audioUri );
    fclose(audio);
    eprintf( E_MESSAGE , "Generated audio.");

    if ( ( audioStream = open( audioUri.c_str( ) , O_RDONLY ) ) < 0 ) {
        fprintf( stderr , __FILE__": open() failed: %s\n" , strerror( errno ) );
        goto finish;
    }

    if ( dup2( audioStream , STDIN_FILENO ) < 0 ) {
        fprintf( stderr , __FILE__": dup2() failed: %s\n" , strerror( errno ) );
        goto finish;
    }

    close( audioStream );

    pulseConnection = pa_simple_new( NULL , "AudioPush" , PA_STREAM_PLAYBACK , NULL , "openMary C++" , &simpleSpecs , NULL , NULL , &error );

    for (int i = 0;;i++ ) {
        const int bufferSize = 1024;
        uint8_t audioBuffer[bufferSize];
        ssize_t r;
        eprintf( E_MESSAGE , "Buffering %d..",i);
        /* Read some data ... */
        if ( ( r = read( STDIN_FILENO , audioBuffer , sizeof (audioBuffer ) ) ) <= 0 ) {
            if ( r == 0 ) /* EOF */
                break;

            eprintf( E_ERROR , __FILE__": read() failed: %s\n" , strerror( errno ) );
    if ( pulseConnection )
        pa_simple_free( pulseConnection );

        }

        /* ... and play it */
        if ( pa_simple_write( pulseConnection , audioBuffer , ( size_t ) r , &error ) < 0 ) {
            fprintf( stderr , __FILE__": pa_simple_write() failed: %s\n" , pa_strerror( error ) );
    if ( pulseConnection )
        pa_simple_free( pulseConnection );

        }

        usleep(2);

    }
    /* Make sure that every single sample was played */
    if ( pa_simple_drain( pulseConnection , &error ) < 0 ) {
        fprintf( stderr , __FILE__": pa_simple_drain() failed: %s\n" , pa_strerror( error ) );
    if ( pulseConnection )
        pa_simple_free( pulseConnection );
    }    
}

NOTE: If you want the rest of the code to this file, you can download it here directly from Launchpad.

Update: I tried using GStreamermm , and this won't work:

    Glib::RefPtr<Pipeline> pipeline;
    Glib::RefPtr<Element> sink, filter, source;
    Glib::RefPtr<Gio::File> audioSrc = Gio::File::create_for_path(uri);

    pipeline = Pipeline::create("audio-playback");
    source = ElementFactory::create_element("alsasrc","source");
    filter = ElementFactory::create_element("identity","filter");
    sink = ElementFactory::create_element("alsasink","sink");
    //sink->get_property("file",audioSrc);
    if (!source || !filter || !sink){
        showErrorDialog("Houston!","We got a problem.");
        return;
    }
    pipeline->add(source)->add(filter)->add(sink);
    source->link(sink);

    pipeline->set_state(Gst::STATE_PLAYING);
    showInformation("Close this to stop recording");
    pipeline->set_state(Gst::STATE_PAUSED);

Answer 1

The "Hello World" application in the GStreamer documentation shows how to play an Ogg/Vorbis file. To make this work with WAV files, you can simply replace "oggdemux" with "wavparse" and replace "vorbisdec" with "identity" (the identity plugin does nothing -- it's just a placeholder).

To install development support for GStreamer (on Ubuntu)...

sudo apt-get install libgstreamer0.10-dev

You need the following on the gcc command-line to enable the use of GStreamer libraries...

$(pkg-config --cflags --libs gstreamer-0.10)

By the way, you may find it useful to use "gst-launch" for prototyping GStreamer pipelines before writing the code.

## recording
gst-launch-0.10 autoaudiosrc ! wavenc ! filesink location=temp.wav

## playback
gst-launch-0.10 filesrc location=temp.wav ! wavparse ! autoaudiosink

A feature of GStreamer that may be useful for voice recognition is that it is easy to insert audio quality filters into a pipeline -- so you could, for example, reduce noise that might otherwise be in the recording. A pointer to a list of the GStreamer "good" plugins is here .

Also of interest, "PocketSphinx" (which seems to be related to your project) already has some GStreamer integration. See Using PocketSphinx with GStreamer and Python

Answer 2

GStreamer/Pulse/JACK are great. For simple and fast things you might use SoX http://sox.sourceforge.net/