【发布时间】：2015-04-16 11:59:03
【问题描述】：
我正在尝试将 16 位 ALSA PCM 样本转换为无符号 8 位 PCM 样本，以便在 Linux 上进行无线传输。接收机器正在成功播放传输的数据，并且录制的语音在那里并且可以识别，但是质量很糟糕并且很嘈杂。我已经在两端尝试了 ALSA 混音器来调整流，但它似乎并没有变得更好。我相信我将样本转换为 8 位 PCM 有问题，但这只是一个简单的转换，所以我不确定可能是什么错误。有人对我的转换代码有任何建议或发现有什么问题吗？谢谢。
转换代码：
// This byte array needs to be the packet size we wish to send
QByteArray prepareToSend;
prepareToSend.clear();
// Keep reading from ALSA until we fill one full frame
int frames = 1;
while ( prepareToSend.size() < TARGET_TX_BUFFER_SIZE ) {
// Create a ByteArray
QByteArray readBytes;
readBytes.resize(size);
// Read with ALSA
short sample[1]; // Data is signed 16-bit
int rc = snd_pcm_readi(m_PlaybackHandle, sample, frames);
if (rc == -EPIPE) {
/* EPIPE means overrun */
fprintf(stderr, "Overrun occurred\n");
snd_pcm_prepare(m_PlaybackHandle);
} else if (rc < 0) {
fprintf(stderr,
"Error from read: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "Short read, read %d frames\n", rc);
}
else {
// Copy bytes to the prepare to send buffer
//qDebug() << "Bytes for sample buffer: " << sizeof(sample);
prepareToSend.append((qint16)(sample[0]) >> 8); // signed 16-bit becomes u8
}
}
ALSA 配置：
// Setup parameters
int size;
snd_pcm_t *m_PlaybackHandle;
snd_pcm_hw_params_t *m_HwParams;
char *buffer;
qDebug() << "Desire to Transmit Data - Setting up ALSA Now....";
// Error handling
int err;
// Device to Write to
const char *snd_device_in = "hw:1,0";
if ((err = snd_pcm_open (&m_PlaybackHandle, snd_device_in, SND_PCM_STREAM_CAPTURE, 0)) < 0) {
fprintf (stderr, "Cannot open audio device %s (%s)\n",
snd_device_in,
snd_strerror (err));
exit (1);
}
/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(&m_HwParams);
if ((err = snd_pcm_hw_params_malloc (&m_HwParams)) < 0) {
fprintf (stderr, "Cannot allocate hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_any (m_PlaybackHandle, m_HwParams)) < 0) {
fprintf (stderr, "Cannot initialize hardware parameter structure (%s)\n",
snd_strerror (err));
exit (1);
}
if ((err = snd_pcm_hw_params_set_access (m_PlaybackHandle, m_HwParams, 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(m_PlaybackHandle, m_HwParams, SND_PCM_FORMAT_S16)) < 0) { // Has to be 16 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 (m_PlaybackHandle, m_HwParams, 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 (m_PlaybackHandle, m_HwParams, 1)) < 0) { // 1 Channel Mono
fprintf (stderr, "Cannot set channel count (%s)\n",
snd_strerror (err));
exit (1);
}
/*
Frames: samples x channels (i.e: stereo frames are composed of two samples, mono frames are composed of 1 sample,...)
Period: Number of samples tranferred after which the device acknowledges the transfer to the apllication (usually via an interrupt).
*/
/* Submit params to device */
if ((err = snd_pcm_hw_params(m_PlaybackHandle, m_HwParams)) < 0) {
fprintf (stderr, "Cannot set parameters (%s)\n",
snd_strerror (err));
exit (1);
}
/* Free the Struct */
snd_pcm_hw_params_free(m_HwParams);
// Flush handle prepare for record
snd_pcm_drop(m_PlaybackHandle);
if ((err = snd_pcm_prepare (m_PlaybackHandle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
qDebug() << "Done Setting up ALSA....";
// Prepare the device
if ((err = snd_pcm_prepare (m_PlaybackHandle)) < 0) {
fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
snd_strerror (err));
exit (1);
}
【问题讨论】：
-
8000 采样率听起来非常低...您确定您没有听到该采样率的确切含义吗？
-
是的，我在发送端用相同的帧速率使用 PulseAudio 进行了同样的尝试，听起来非常清晰。一点也不吵。我还在 PulseAudio 配置中关闭了重采样。这是在小型嵌入式 Linux 设备上运行的，因此 PulseAudio 在获取数据时导致了很长（10 秒）的延迟。我确定它正在过滤，但没有重新混合。
-
数据向右移动 8 位将消除几乎所有的声音细节。建议使用某种压缩算法将 16 位解析为 8 位。或许通过从 16 位中提取成对的位，使用多数规则，然后使用高位规则将数据压缩为 8 位。