Question

我有一个带麦克风的设备通过以太网连接到我的电脑，它不能被Qt视为音频设备，因此，我从它获取数据包并将它们放到QByteArray。我需要从流中播放这些数据包。在互联网的某个地方，我找到了几乎同样问题的解决方案，但是使用了内部麦克风。

#include <QApplication>

#include <iostream>
#include <cassert>

#include <QCoreApplication>
#include <QAudioInput>
#include <QAudioOutput>
#include <QBuffer>

int main(int argc, char *argv[]) {
    QCoreApplication app(argc, argv);

    QBuffer rdBuff;
    QBuffer wrBuff;
    wrBuff.open(QBuffer::WriteOnly);
    rdBuff.open(QBuffer::ReadOnly);

    QObject::connect(&wrBuff, &QIODevice::bytesWritten, [&wrBuff, &rdBuff](qint64)
    {
        rdBuff.buffer().remove(0, rdBuff.pos());

        // set pointer to the beginning of the unread data
        const auto res = rdBuff.seek(0);
        assert(res);

        // write new data
        rdBuff.buffer().append(wrBuff.buffer());

        // remove all data that was already written
        wrBuff.buffer().clear();
        wrBuff.seek(0);
    });

    const auto decideAudioFormat = [](const QAudioDeviceInfo& devInfo)
    {
        QAudioFormat format;
        format.setSampleRate(8000);
        format.setChannelCount(1);
        format.setSampleSize(16);
        format.setCodec("audio/pcm");
        format.setByteOrder(QAudioFormat::LittleEndian);
        format.setSampleType(QAudioFormat::SignedInt);

        if (devInfo.isFormatSupported(format))
        {
            return format;
        }
        else
        {
            std::cerr << "Raw audio format not supported by backend, cannot play audio.\n";
            throw 0;
        }
    };

    QAudioInput audioInput(decideAudioFormat(QAudioDeviceInfo::defaultInputDevice()));
    QAudioOutput audioOutput(decideAudioFormat(QAudioDeviceInfo::defaultOutputDevice()));

    audioInput.start(&wrBuff);
    audioOutput.start(&rdBuff);

    return app.exec();
}

它工作得很好，但我需要将QByteArray设置为QAudioInput的源。有没有可能的解决方案？

Answer 1

不确定我是否直接回答了您的问题。但是一种可能的解决方案是在新数据到来时手动输入输出音频设备（推模式）。

您还可以使用自定义（QFile继承）类来录制声音，当声音到来时，同时输入文件和输出音频设备。

这是一个例子：

AudioOutput.h：

#ifndef AUDIOOUTPUT_H
#define AUDIOOUTPUT_H

#include <QtCore>
#include <QtMultimedia>

#define MAX_BUFFERED_TIME 10*1000

static inline int timeToSize(int ms, const QAudioFormat &format)
{
    return ((format.channelCount() * (format.sampleSize() / 8) * format.sampleRate()) * ms / 1000);
}

class AudioOutput : public QObject
{
    Q_OBJECT
public:
    explicit AudioOutput(QObject *parent = nullptr);

public slots:
    bool start(const QAudioDeviceInfo &devinfo,
               const QAudioFormat &format,
               int time_to_buffer);

    void write(const QByteArray &data);

private slots:
    void verifyBuffer();
    void preplay();
    void play();

private:
    bool m_initialized;
    QAudioOutput *m_audio_output;
    QIODevice *m_device;
    QByteArray m_buffer;
    bool m_buffer_requested;
    bool m_play_called;
    int m_size_to_buffer;
    int m_time_to_buffer;
    int m_max_size_to_buffer;
    QAudioFormat m_format;
};

#endif // AUDIOOUTPUT_H

AudioRecorder.h：

#ifndef AUDIORECORDER_H
#define AUDIORECORDER_H

#include <QtCore>
#include <QtMultimedia>

class AudioRecorder : public QFile
{
    Q_OBJECT
public:
    explicit AudioRecorder(const QString &name, const QAudioFormat &format, QObject *parent = nullptr);
    ~AudioRecorder();

    using QFile::open;

public slots:
    bool open();
    qint64 write(const QByteArray &data);
    void close();

private:
    void writeHeader();
    bool hasSupportedFormat();
    QAudioFormat format;
};

#endif // AUDIORECORDER_H

AudioOutput.cpp：

#include "audiooutput.h"

AudioOutput::AudioOutput(QObject *parent) : QObject(parent)
{
    m_initialized = false;
    m_audio_output = nullptr;
    m_device = nullptr;
    m_buffer_requested = true;
    m_play_called = false;
    m_size_to_buffer = 0;
    m_time_to_buffer = 0;
    m_max_size_to_buffer = 0;
}

bool AudioOutput::start(const QAudioDeviceInfo &devinfo,
                        const QAudioFormat &format,
                        int time_to_buffer)
{
    if (!devinfo.isFormatSupported(format))
    {
        qDebug() << "Format not supported by output device";
        return m_initialized;
    }

    m_format = format;

    int internal_buffer_size;

    //Adjust internal buffer size
    if (format.sampleRate() >= 44100)
        internal_buffer_size = (1024 * 10) * format.channelCount();
    else if (format.sampleRate() >= 24000)
        internal_buffer_size = (1024 * 6) * format.channelCount();
    else
        internal_buffer_size = (1024 * 4) * format.channelCount();

    //Initialize the audio output device
    m_audio_output = new QAudioOutput(devinfo, format, this);
    //Increase the buffer size to enable higher sample rates
    m_audio_output->setBufferSize(internal_buffer_size);

    m_time_to_buffer = time_to_buffer;
    //Compute the size in bytes to be buffered based on the current format
    m_size_to_buffer = timeToSize(m_time_to_buffer, m_format);
    //Define a highest size that the buffer are allowed to have in the given time
    //This value is used to discard too old buffered data
    m_max_size_to_buffer = m_size_to_buffer + timeToSize(MAX_BUFFERED_TIME, m_format);

    m_device = m_audio_output->start();

    if (!m_device)
    {
        qDebug() << "Failed to open output audio device";
        return m_initialized;
    }

    //Timer that helps to keep playing data while it's available on the internal buffer
    QTimer *timer_play = new QTimer(this);
    timer_play->setTimerType(Qt::PreciseTimer);
    connect(timer_play, &QTimer::timeout, this, &AudioOutput::preplay);
    timer_play->start(10);

    //Timer that checks for too old data in the buffer
    QTimer *timer_verifier = new QTimer(this);
    connect(timer_verifier, &QTimer::timeout, this, &AudioOutput::verifyBuffer);
    timer_verifier->start(qMax(m_time_to_buffer, 10));

    m_initialized = true;

    return m_initialized;
}

void AudioOutput::verifyBuffer()
{
    if (m_buffer.size() >= m_max_size_to_buffer)
        m_buffer.clear();
}

void AudioOutput::write(const QByteArray &data)
{
    m_buffer.append(data);
    preplay();
}

void AudioOutput::preplay()
{
    if (!m_initialized)
        return;

    //Verify if exists a pending call to play function
    //If not, call the play function async
    if (!m_play_called)
    {
        m_play_called = true;
        QMetaObject::invokeMethod(this, "play", Qt::QueuedConnection);
    }
}

void AudioOutput::play()
{
    //Set that last async call was triggered
    m_play_called = false;

    if (m_buffer.isEmpty())
    {
        //If data is empty set that nothing should be played
        //until the buffer has at least the minimum buffered size already set
        m_buffer_requested = true;
        return;
    }
    else if (m_buffer.size() < m_size_to_buffer)
    {
        //If buffer doesn't contains enough data,
        //check if exists a already flag telling that the buffer comes
        //from a empty state and should not play anything until have the minimum data size
        if (m_buffer_requested)
            return;
    }
    else
    {
        //Buffer is ready and data can be played
        m_buffer_requested = false;
    }

    int readlen = m_audio_output->periodSize();

    int chunks = m_audio_output->bytesFree() / readlen;

    //Play data while it's available in the output device
    while (chunks)
    {
        //Get chunk from the buffer
        QByteArray samples = m_buffer.mid(0, readlen);
        int len = samples.size();
        m_buffer.remove(0, len);

        //Write data to the output device
        if (len)
            m_device->write(samples);

        //If chunk is smaller than the output chunk size, exit loop
        if (len != readlen)
            break;

        //Decrease the available number of chunks
        chunks--;
    }
}

AudioRecorder.cpp：

#include "audiorecorder.h"

AudioRecorder::AudioRecorder(const QString &name, const QAudioFormat &format, QObject *parent) : QFile(name, parent), format(format)
{

}

AudioRecorder::~AudioRecorder()
{
    if (!isOpen())
        return;

    close();
}

bool AudioRecorder::hasSupportedFormat()
{
    return (format.sampleSize() == 8
            && format.sampleType() == QAudioFormat::UnSignedInt)
            || (format.sampleSize() > 8
                && format.sampleType() == QAudioFormat::SignedInt
                && format.byteOrder() == QAudioFormat::LittleEndian);
}

bool AudioRecorder::open()
{
    if (!hasSupportedFormat())
    {
        setErrorString("Wav PCM supports only 8-bit unsigned samples "
                       "or 16-bit (or more) signed samples (in little endian)");
        return false;
    }
    else
    {
        if (!QFile::open(ReadWrite | Truncate))
            return false;
        writeHeader();
        return true;
    }
}

qint64 AudioRecorder::write(const QByteArray &data)
{
    return QFile::write(data);
}

void AudioRecorder::writeHeader()
{
    QDataStream out(this);
    out.setByteOrder(QDataStream::LittleEndian);

    // RIFF chunk
    out.writeRawData("RIFF", 4);
    out << quint32(0); // Placeholder for the RIFF chunk size (filled by close())
    out.writeRawData("WAVE", 4);

    // Format description chunk
    out.writeRawData("fmt ", 4);
    out << quint32(16); // "fmt " chunk size (always 16 for PCM)
    out << quint16(1); // data format (1 => PCM)
    out << quint16(format.channelCount());
    out << quint32(format.sampleRate());
    out << quint32(format.sampleRate() * format.channelCount()
                   * format.sampleSize() / 8 ); // bytes per second
    out << quint16(format.channelCount() * format.sampleSize() / 8); // Block align
    out << quint16(format.sampleSize()); // Significant Bits Per Sample

    // Data chunk
    out.writeRawData("data", 4);
    out << quint32(0); // Placeholder for the data chunk size (filled by close())

    Q_ASSERT(pos() == 44); // Must be 44 for WAV PCM
}

void AudioRecorder::close()
{
    // Fill the header size placeholders
    quint32 fileSize = size();

    QDataStream out(this);
    // Set the same ByteOrder like in writeHeader()
    out.setByteOrder(QDataStream::LittleEndian);
    // RIFF chunk size
    seek(4);
    out << quint32(fileSize - 8);

    // data chunk size
    seek(40);
    out << quint32(fileSize - 44);

    QFile::close();
}

main.cpp中：

#include <QtCore>
#include "audiooutput.h"
#include "audiorecorder.h"
#include <signal.h>

QByteArray tone_generator()
{
    //Tone generator from http://www.cplusplus.com/forum/general/129827/

    const unsigned int samplerate = 8000;
    const unsigned short channels = 1;

    const double pi = M_PI;
    const qint16 amplitude = std::numeric_limits<qint16>::max() * 0.5;

    const unsigned short n_frequencies = 8;
    const unsigned short n_seconds_each = 1;

    float frequencies[n_frequencies] = {55.0, 110.0, 220.0, 440.0, 880.0, 1760.0, 3520.0, 7040.0};

    const int n_samples = channels * samplerate * n_frequencies * n_seconds_each;

    QVector<qint16> data;
    data.resize(n_samples);

    int index = n_samples / n_frequencies;

    for (unsigned short i = 0; i < n_frequencies; ++i)
    {
        float freq = frequencies[i];
        double d = (samplerate / freq);
        int c = 0;

        for (int j = index * i; j < index * (i + 1); j += 2)
        {
            double deg = 360.0 / d;
            data[j] = data[j + (channels - 1)] = qSin((c++ * deg) * pi / 180.0) * amplitude;
        }
    }

    return QByteArray((char*)data.data(), data.size() * sizeof(qint16));
}

void signalHandler(int signum)
{
    qDebug().nospace() << "Interrupt signal (" << signum << ") received.";

    qApp->exit();
}

int main(int argc, char *argv[])
{
    //Handle console close to ensure destructors being called
#ifdef Q_OS_WIN
    signal(SIGBREAK, signalHandler);
#else
    signal(SIGHUP, signalHandler);
#endif
    signal(SIGINT, signalHandler);

    QCoreApplication a(argc, argv);

    QAudioFormat format;
    format.setSampleRate(8000);
    format.setChannelCount(1);
    format.setSampleSize(16);
    format.setCodec("audio/pcm");
    format.setByteOrder(QAudioFormat::LittleEndian);
    format.setSampleType(QAudioFormat::SignedInt);

    AudioOutput output;

    AudioRecorder file("tone.wav", format);

    if (!output.start(QAudioDeviceInfo::defaultOutputDevice(), format, 10 * 1000)) //10 seconds of buffer
        return a.exec();

    if (!file.open())
    {
        qDebug() << qPrintable(file.errorString());
        return a.exec();
    }

    qDebug() << "Started!";

    QByteArray audio_data = tone_generator();

    QTimer timer;

    QObject::connect(&timer, &QTimer::timeout, [&]{
        qDebug() << "Writting" << audio_data.size() << "bytes";
        output.write(audio_data);
        file.write(audio_data);
    });

    qDebug() << "Writting" << audio_data.size() << "bytes";
    output.write(audio_data);
    file.write(audio_data);

    timer.start(8000); //8 seconds because we generated 8 seconds of sound

    return a.exec();
}

Answer 2

好。我尝试将代码（现在只是回放）应用到我的情况但是有一个问题是它只播放一次数据，如果没有发出＆＃34; MorePackets（）＆＃34;或者没有播放如果＆＃34; MorePackets（）＆＃34;发生。

micserver.h

#ifndef MICSERVER_H
#define MICSERVER_H
//-----------------------------//
#include <QObject>
#include <QDebug>
#include <QTcpServer>
#include <QTcpSocket>
#include <QAudioFormat>
#include <QAudioOutput>
#include <QFile>
#include <QBuffer>
#include <QByteArray>
#include <QDataStream>

#include <iostream>

#include "audiooutput.h"
//-----------------------------//
class MicServer : public QObject {
    Q_OBJECT
public:
    explicit MicServer(QObject *parent = nullptr);

    QTcpSocket* Socket;
private:
    QTcpServer* Server;
    QAudioFormat format;
    AudioOutput output;

    QByteArray GetPackets(QTcpSocket* ClientP, int PacketsNumP);
signals:
    void MorePackets();

public slots:
    void NewConnection();
    void Play();
};
//-----------------------------//
#endif

micserver.cpp

#include "micserver.h"
//-----------------------------//
QByteArray MicServer :: GetPackets(QTcpSocket* ClientP, int PacketsNumP) {
    QBuffer BufferL;
    BufferL.open(QBuffer :: WriteOnly);
    QDataStream InputL(&BufferL);
    InputL.setVersion(QDataStream::Qt_5_10);

    QByteArray TempArray;

    for (int i = 0; i < PacketsNumP; i++) {
        ClientP -> waitForReadyRead(3000);
        InputL << ClientP -> readAll();
    }

    for (int i = 0; i < PacketsNumP; i++) {
        TempArray.push_back(BufferL.buffer().mid(76 + i * 1172, 256));
    }

    BufferL.close();

    return TempArray;
}
//-----------------------------//
MicServer :: MicServer(QObject *parent) : QObject(parent) {
    Server = new QTcpServer;

    if (!Server -> listen(QHostAddress :: Any, 49112)) {
        qDebug() << "Failed to launch server!";
    } else {
        qDebug() << "Server launched!";
    }

    connect(Server, SIGNAL(newConnection()), this, SLOT(NewConnection()));
    connect(Server, SIGNAL(newConnection()), this, SLOT(Play()));
    connect(this, SIGNAL(MorePackets()), this, SLOT(Play()));

    format.setSampleRate(16000);
    format.setChannelCount(1);
    format.setSampleSize(16);
    format.setCodec("audio/pcm");
    format.setByteOrder(QAudioFormat::LittleEndian);
    format.setSampleType(QAudioFormat::SignedInt);

    output.start(QAudioDeviceInfo::defaultOutputDevice(), format);
}

void MicServer :: NewConnection() {
    Socket = Server -> nextPendingConnection();
    qDebug() << "New connection!";

    qDebug() << Socket -> localAddress().toString();
}

void MicServer :: Play() {
    QByteArray audio_data = GetPackets(Socket, 250);

    QTimer timer;

    QObject::connect(&timer, &QTimer::timeout, [&]{
        qDebug() << "Writting" << audio_data.size() << "bytes";
        output.write(audio_data);
    });

    qDebug() << "Writting" << audio_data.size() << "bytes";
    output.write(audio_data);

    timer.start(2000);

    emit(MorePackets());
}

来自数组的QAudioInput

2 个答案: