How to monitor playback on ALSA via asoundlib? - linux

Question

I'm building an application that allows for ALSA configuration and in the GUI there is a peek meter that allows the client to see playback levels in realtime. I'm having a hard time determining what device to connect to because I don't know if ALSA has a default "loopback" or not and what it's called. I am also having trouble converting the read data into a sample, then finding said sample's amplitude. Here is what I have built so far:
Grab device and set hardware params
if (0 == snd_pcm_open(&pPcm, "default", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK))
{
if (0 == snd_pcm_set_params(pPcm, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 1, 96000, 1, 1)) // This last argument confuses me because I'm not given a unit of measurement (second, millisecond, mircosecond, etc.)
{
return snd_pcm_start(pPcm);
}
}
pPcm = nullptr;
return -1;
Read from device and return the peek of the audio signal
int iRtn = -1;
if (nullptr == pPcm)
{
if (-1 == SetupListener())
{
return iRtn;
}
}
// Check to make the state is sane for reading.
if (SND_PCM_STATE_PREPARED == snd_pcm_state(pPcm) ||
SND_PCM_STATE_RUNNING == snd_pcm_state(pPcm))
{
snd_pcm_resume(pPcm); // This might be superfluous.
// The state is sane, read from the stream.
signed short iBuffer = 0;
int iNumRead = snd_pcm_readi(pPcm, &iBuffer, 1);
if (0 < iNumRead)
{
// This calculates an approximation.
// We have some audio data, acquire it's peek in dB. (decibels)
float nSample = static_cast<float>(iBuffer);
float nAmplitude = nSample / MAX_AMPLITUDE_S16; // MAX_AMPLITUDE_S16 is defined as "32767"
float nDecibels = (0 < nAmplitude) ? 20 * log10(nAmplitude) : 0;
iRtn = static_cast<int>(nDecibels); // Cast to integer for GUI element.
}
}
return iRtn;
The ALSA documentation seems very barren and so I apologize if I'm misusing the API.