我已经使用PulseAudio和Qt5.4的QAudioOutput在Raspberry Pi的3.5mm耳机插孔模拟输出上成功播放了音频。音频已通过XBee链接以8位样本从XBee链接以8KHz成功传输。
PulseAudio的等待时间很长，所以我决定与libasound(ALSA)链接并直接播放音频。我的代码在下面，可以成功打开并播放声音，但是几乎无法识别，有很多crack啪声和吱吱声。如果我对着远程麦克风讲话，我会很快听到Pi发出的耳机刮擦声和吱吱声(但这不是很好的音频)。我想我的参数搞砸了。
1.)数据以BigEndian传输-QAudioOutput允许您通知其样本为BigEndian。但是这些是U8样本，所以我需要担心字节顺序吗？
2.)您在下面的配置中看到任何问题吗？
3.)如何确定Pi上输出的ALSA的分片大小？
4.)有人可以解释如何将缓冲区写入音频设备吗？
谢谢!
这是我的代码:
UdpReceiver::UdpReceiver(QObject *parent) :
QObject(parent)
{
// Debug
qDebug() << "Setting up a UDP Socket...";
// Create a socket
m_Socket = new QUdpSocket(this);
// Bind to the 2616 port
bool didBind = m_Socket->bind(QHostAddress::Any, 0x2616);
if ( !didBind ) {
qDebug() << "Error - could not bind to UDP Port!";
}
else {
qDebug() << "Success binding to port 0x2616!";
}
// Get notified that data is incoming to the socket
connect(m_Socket, SIGNAL(readyRead()), this, SLOT(readyRead()));
// Init to Zero
m_NumberUDPPacketsReceived = 0;
}
void UdpReceiver::readyRead() {
// When data comes in
QByteArray buffer;
buffer.resize(m_Socket->pendingDatagramSize());
QHostAddress sender;
quint16 senderPort;
// Cap buffer size
int lenToRead = buffer.size();
if ( buffer.size() > NOMINAL_AUDIO_BUFFER_SIZE ) {
lenToRead = NOMINAL_AUDIO_BUFFER_SIZE;
}
// Read the data from the UDP Port
m_Socket->readDatagram(buffer.data(), lenToRead,
&sender, &senderPort);
// Kick off audio playback
if ( m_NumberUDPPacketsReceived == 0 ) {
qDebug() << "Received Data - Setting up ALSA Now....";
// Error handling
int err;
// Device to Write to
char *snd_device_out = "hw:0,0";
if ((err = snd_pcm_open (&playback_handle, snd_device_out, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
fprintf (stderr, "cannot open audio device %s (%s)\n",
snd_device_out,
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_U8)) < 0) { // Unsigned 8 bit
fprintf (stderr, "cannot set sample format (%s)\n",
snd_strerror (err));
exit (1);
}
uint sample_rate = 8000;
if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, &sample_rate, 0)) < 0) { // 8 KHz
fprintf (stderr, "cannot set sample rate (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 1)) < 0) { // 1 Channel Mono
fprintf (stderr, "cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n",
snd_strerror (err));
exit (1);
}
snd_pcm_hw_params_free (hw_params);
// Flush handle prepare for playback
snd_pcm_drop(playback_handle);
if ((err = snd_pcm_prepare (playback_handle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
qDebug() << "Done Setting up ALSA....";
}
// Grab the buffer
m_Buffer = buffer.data();
// Write the data to the ALSA device
int error;
for (int i = 0; i < 10; ++i) {
if ((error = snd_pcm_writei (playback_handle, m_Buffer, NOMINAL_AUDIO_BUFFER_SIZE)) != NOMINAL_AUDIO_BUFFER_SIZE) {
fprintf (stderr, "write to audio interface failed (%s)\n",
snd_strerror (error));
exit (1);
}
}
// Count up
m_NumberUDPPacketsReceived++;
}
最佳答案
snd_pcm_hw_params_set_rate_near()将费率更改为最接近的支持费率。您可能不想要那样。 它具有缓冲区和周期大小；您需要根据您的时间要求设置它们(请参阅ALSA: Relation between period size of speaker and mic)。
关于c++ - 在树莓派模拟输出上用于ALSA接收和播放原始PCM的配置，我们在Stack Overflow上找到一个类似的问题：https://stackoverflow.com/questions/28384928/