It’s kind of like having a direct line to your sound hardware, but with more code and less magic.
So here’s what happens when you want to play some sweet tunes on your computer:
1. You open up your favorite music player (let’s say VLC) and hit that play button.
2. VLC sends a request to the ALSA API, which is like saying ” Can we please have some audio data?”
3. The ALSA API then goes ahead and grabs that audio data from wherever it needs to (like your hard drive or streaming service) and packages it up into little chunks called buffers.
4. These buffered chunks of audio are then sent over to the sound hardware, which plays them back for you in all their glory.
5. The ALSA API keeps track of how many bytes have been played so far (called the “position”) and sends that information back to VLC, who uses it to keep track of where they’re at in the song.
6. Repeat steps 2-5 until you hit stop or pause or something like that.
Now some specific functions within the ALSA API:
1. snd_pcm_open() This function opens up a new audio stream (called a “PCM”) and returns a handle to it, which we can use later on for playing/recording stuff.
2. snd_pcm_prepare() This function prepares our PCM for playback by setting up all the necessary buffers and whatnot.
3. snd_pcm_writei() This function writes some audio data to our PCM, which is then sent over to the sound hardware for playing back.
4. snd_pcm_readi() This function reads some audio data from our PCM (which could be used for recording stuff).
5. snd_pcm_close() This function closes up our PCM and frees up any resources that were being used by it.
6. snd_pcm_drain() This function drains all the remaining audio data from our PCM, which is useful if you want to make sure everything has been played back before moving on to something else (like closing your music player).
7. snd_pcm_avail_update() This function updates the available buffer size for our PCM, which can be helpful if we’re dealing with a lot of audio data at once and need to make sure everything is being handled efficiently.
8. snd_pcm_hw_params_set_format() This function sets the format (like 16-bit or 32-bit) for our PCM, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
9. snd_pcm_hw_params_set_rate() This function sets the sample rate (like 44100 Hz or 48000 Hz) for our PCM, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
10. snd_pcm_hw_params() This function sets the hardware parameters (like buffer size and sample format) for our PCM based on what was passed in as arguments, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
11. snd_pcm_hw_params_get_buffer_size() This function gets the current buffer size for our PCM based on what was passed in as arguments, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
12. snd_pcm_hw_params_get_format() This function gets the current sample format (like 16-bit or 32-bit) for our PCM based on what was passed in as arguments, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
13. snd_pcm_hw_params_get_rate() This function gets the current sample rate (like 44100 Hz or 48000 Hz) for our PCM based on what was passed in as arguments, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
14. snd_pcm_hw_params() This function sets the hardware parameters (like buffer size and sample format) for our PCM based on what was passed in as arguments, which can be useful if we’re dealing with different types of audio data and need to make sure everything is being handled correctly.
So basically, ALSA API lets us talk directly to the sound hardware without any fancy drivers or system-level stuff getting in the way. It opens up a new audio stream (called PCM), prepares it for playback by setting up all necessary buffers and whatnot, writes some audio data to it which is then sent over to the sound hardware for playing back, reads some audio data from it if we want to record stuff, closes it up when we’re done, drains any remaining audio data before moving on to something else, updates available buffer size, sets format and sample rate based on what was passed in as arguments, gets current buffer size, format, and sample rate for our PCM.