ALSA using pcm_s24le

Question

Using Linux Mint 17.1 and ALSA. I have two wav files producing identical sound: one using pcm_s16le and the other using pcm_s24le. Each is played correctly by Totem/videos. My code to set hardware parameters and to playback using pcm_s16le works fine. However, when I attempt to adjust these parameters to accommodate pcm_s24le as follows:

snd_pcm_hw_params_set_format(audioHandle, audioParams, SND_PCM_FORMAT_S24_LE);

[I have simply substituted 'SND_PCM_FORMAT_S24LE' for 'SND_PCM_FORMAT_S16_LE']. The call to snd_pcm_writei is

snd_pcm_writei(m_audioHandle, *m_pAudioFrameData, *m_pAudioFrameSize / (m_nChannels * m_bitsPerSample / 8);

I get mostly garbage sound (hissing, choppiness) with a hint of the correct sound.

Essentially my question is, how do I convert code that works for SND_PCM_FORMAT_S16_LE to work for SND_PCM_FORMAT_S24_LE?

Answer 1

There are three possible ways of storing 24-bit samples in memory:

          LSB                           MSB
          1st byte  2nd byte  3rd byte  4th byte   alignment
S32_LE:   00000000  xxxxxxxx  xxxxxxxx  xxxxxxxx   32 bits
S24_LE:   xxxxxxxx  xxxxxxxx  xxxxxxxx  00000000   32 bits
S24_3LE:  xxxxxxxx  xxxxxxxx  xxxxxxxx             24 bits

Most hardware uses S32_LE, except for USB, which uses S24_3LE. There is no hardware that uses S24_LE.

ALSA can automatically convert the sample format, but you have to describe your own sample format correctly.