HOW AUDIO EMULATION WORKS IN QEMU:
==================================

Things are a bit tricky, but here's a rough description:

  QEMUSoundCard: models a given emulated sound card
  SWVoiceOut:    models an audio output from a QEMUSoundCard
  SWVoiceIn:     models an audio input from a QEMUSoundCard

  HWVoiceOut:    models an audio output (backend) on the host.
  HWVoiceIn:     models an audio input (backend) on the host.

Each voice can have its own settings in terms of sample size, endianess, rate, etc...


Emulation for a given soundcard typically does:

  1/ Create a QEMUSoundCard object and register it with AUD_register_card()
  2/ For each emulated output, call AUD_open_out() to create a SWVoiceOut object.
  3/ For each emulated input, call AUD_open_in() to create a SWVoiceIn object.

  Note that you must pass a callback function to AUD_open_out() and AUD_open_in();
  more on this later.

  Each SWVoiceOut is associated to a single HWVoiceOut, each SWVoiceIn is
  associated to a single HWVoiceIn.

  However you can have several SWVoiceOut associated to the same HWVoiceOut
  (same thing for SWVoiceIn/HWVoiceIn).

SOUND PLAYBACK DETAILS:
=======================

Each HWVoiceOut has the following too:

  - A fixed-size circular buffer of stereo samples (for stereo).
    whose format is either floats or int64_t per sample (depending on build configuration).

  - A 'samples' field giving the (constant) number of sample pairs in the stereo buffer.

  - A target conversion function, called 'clip()' that is used to read from the stereo
    buffer and write into a platform-specific sound buffers (e.g. WinWave-managed buffers
    on Windows).

  - A 'rpos' offset into the circular buffer which tells where to read the next samples
    from the stereo buffer for the next conversion through 'clip'.

  - A 'run_out' method that is called each time to tell the output backend to
    send samples from the stereo buffer to the host sound card/server. This method
    shall also modify 'rpos' and returns the number of samples 'played'. A more detailed
    description of this process appears below.

  - A 'write' method callback used to write a buffer of emulated sound samples from
    a SWVoiceOut into the stereo buffer. *All* backends simply call the generic
    function audio_pcm_sw_write() to implement this. It's difficult to see why
    it's needed at all ?

    (Similarly, all backends have a 'read' methods which simply calls 'audio_pcm_sw_read')

Each SWVoiceOut has the following:

  - a 'conv()' function used to read sound samples from the emulated sound card and
    copy/mix them to the corresponding HWVoiceOut's stereo buffer.

  - a 'total_hw_samples_mixed' which correspond to the number of samples that have
    already been mixed into the target HWVoiceOut stereo buffer (starting from the
    HWVoiceOut's 'rpos' offset). NOTE: this is a count of samples in the HWVoiceOut
    stereo buffer, not emulated hardware sound samples, which can have different
    properties (frequency, size, endianess).

  - a 'ratio' value, which is the ratio of the target HWVoiceOut's frequency by
    the SWVoiceOut's frequency, multiplied by (1 << 32), as a 64-bit integer.

    So, if the HWVoiceOut has a frequency of 44kHz, and the SWVoiceOut has a frequency
    of 11kHz, then ratio will be (44/11*(1 << 32)) = 0x4_0000_0000

  - a callback provided by the emulated hardware when the SWVoiceOut is created.
    This function is used to mix the SWVoiceOut's samples into the target
    HWVoiceOut stereo buffer (it must also perform frequency interpolation,
    volume adjustment, etc..).

    This callback normally calls another helper functions in the audio subsystem
    (AUD_write()) to to the mixing/volume-adjustment from emulated hardware sample
    buffers.

Here's a small graphics that explains it better:

   SWVoiceOut:  emulated hardware sound buffers:

          |
          |   (mixed through AUD_write() from user-provided callback)
          |
          v

   HWVoiceOut: stereo sample circular buffer

          |
          |   (through HWVoiceOut's 'clip' function, invoked from the
          |    'run_out' method)
          v

   backend-specific sound buffers

THERE IS NO COMMON TIMEBASE BETWEEN ALL LAYERS. DON'T EXPECT ANY HIGH-ACCURACY /
LOW-LATENCY IN THIS IMPLEMENTATION.


The function audio_timer() in audio/audio.c is called periodically and it is used as
a pulse to perform sound buffer transfers and mixing. More specifically for audio
output voices:

- For each HWVoiceOut, find the number of active SWVoiceOut, and the minimum number
  of 'total_hw_samples_mixed' that have already been written to the buffer. We will
  call this value the number of 'live' samples in the stereo buffer.

- if 'live' is 0, call the callback of each active SWVoiceOut to fill the stereo
  buffer, if needed, then exit.

- otherwise, call the 'run_out' method of the HWVoiceOut object. This will change
  the value of 'rpos' and return the number of samples played. Then the
  'total_hw_samples_mixed' field of all active SWVoiceOuts is decremented by
  'played', and the callback is called to re-fill the stereo buffer.

It's important to note that the SWVoiceOut callback:

- takes a 'free' parameter which is the number of emulated sound samples that can
  be sent to the hardware stereo buffer (before rate adjustment, i.e. not the number
  of sound samples in the SWVoiceOut emulated hardware sound buffer).

- must call AUD_write(sw, buff, count), where 'buff' points to emulated sound
  samples, and their 'count', which must be <= the 'free' parameter.

- the implementation of AUD_write() will call the 'write' method of the target
  HWVoiceOut, which in turns calls the function audio_pcm_sw_write() which does
  standard rate/volume adjustment before mixing the conversion into the target
  stereo buffer. It also increases the 'total_hw_samples_mixed' value of the
  SWVoiceOut.

- audio_pcm_sw_write() returns the number of sound sample *bytes* that have
  been mixed into the stereo buffer, and so does AUD_write().

So, in the end, we have the pseudo-code:

    every sound timer ticks:
      for hw in list_HWVoiceOut:
         live = MIN([sw.total_hw_samples_mixed for sw in hw.list_SWVoiceOut ])
         if live > 0:
            played = hw.run_out(live)
            for sw in hw.list_SWVoiceOut:
                sw.total_hw_samples_mixed -= played

        for sw in hw.list_SWVoiceOut:
            free = hw.samples - sw.total_hw_samples_mixed
            if free > 0:
                sw.callback(sw, free)

SOUND RECORDING DETAILS:
========================

Things are similar but in reverse order.