fteqw/engine/client/snd_alsa.c

/*
	snd_alsa.c

	Support for the ALSA 1.0.1 sound driver

	Copyright (C) 1999,2000  contributors of the QuakeForge project
	Please see the file "AUTHORS" for a list of contributors

	This program is free software; you can redistribute it and/or
	modify it under the terms of the GNU General Public License
	as published by the Free Software Foundation; either version 2
	of the License, or (at your option) any later version.

	This program is distributed in the hope that it will be useful,
	but WITHOUT ANY WARRANTY; without even the implied warranty of
	MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.

	See the GNU General Public License for more details.

	You should have received a copy of the GNU General Public License
	along with this program; if not, write to:

		Free Software Foundation, Inc.
		59 Temple Place - Suite 330
		Boston, MA  02111-1307, USA

*/
//actually stolen from darkplaces.
//I guess noone can be arsed to write it themselves. :/
//
//This file is otherwise known as 'will the linux jokers please stop fucking over the open sound system please'

#include <alsa/asoundlib.h>

#include "quakedef.h"
#include <dlfcn.h>

static void *alsasharedobject;

int (*psnd_pcm_open)				(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);
int (*psnd_pcm_close)				(snd_pcm_t *pcm);
const char *(*psnd_strerror)			(int errnum);
int (*psnd_pcm_hw_params_any)			(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
int (*psnd_pcm_hw_params_set_access)		(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t _access);
int (*psnd_pcm_hw_params_set_format)		(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);
int (*psnd_pcm_hw_params_set_channels)		(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);
int (*psnd_pcm_hw_params_set_rate_near)		(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
int (*psnd_pcm_hw_params_set_period_size_near)	(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val, int *dir);
int (*psnd_pcm_hw_params)			(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);
int (*psnd_pcm_sw_params_current)		(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
int (*psnd_pcm_sw_params_set_start_threshold)	(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
int (*psnd_pcm_sw_params_set_stop_threshold)	(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);
int (*psnd_pcm_sw_params)			(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);
int (*psnd_pcm_hw_params_get_buffer_size)	(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
int (*psnd_pcm_hw_params_set_buffer_size_near)	(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
snd_pcm_sframes_t (*psnd_pcm_avail_update)	(snd_pcm_t *pcm);
snd_pcm_state_t (*psnd_pcm_state)		(snd_pcm_t *pcm);
int (*psnd_pcm_start)				(snd_pcm_t *pcm);

size_t (*psnd_pcm_hw_params_sizeof)		(void);
size_t (*psnd_pcm_sw_params_sizeof)		(void);

int (*psnd_pcm_mmap_begin)			(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames);
snd_pcm_sframes_t (*psnd_pcm_mmap_commit)	(snd_pcm_t *pcm, snd_pcm_uframes_t offset, snd_pcm_uframes_t frames);

snd_pcm_sframes_t (*psnd_pcm_writei)		(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);
int (*psnd_pcm_prepare)				(snd_pcm_t *pcm);


static unsigned int ALSA_MMap_GetDMAPos (soundcardinfo_t *sc)
{
	const snd_pcm_channel_area_t *areas;
	snd_pcm_uframes_t offset;
	snd_pcm_uframes_t nframes = sc->sn.samples / sc->sn.numchannels;

	psnd_pcm_avail_update (sc->handle);
	psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);
	offset *= sc->sn.numchannels;
	nframes *= sc->sn.numchannels;
	sc->sn.samplepos = offset;
	sc->sn.buffer = areas->addr;
	return sc->sn.samplepos;
}

static void ALSA_MMap_Submit (soundcardinfo_t *sc, int start, int end)
{
	int			state;
	int			count = end - start;
	const snd_pcm_channel_area_t *areas;
	snd_pcm_uframes_t nframes;
	snd_pcm_uframes_t offset;

	nframes = count / sc->sn.numchannels;

	psnd_pcm_avail_update (sc->handle);
	psnd_pcm_mmap_begin (sc->handle, &areas, &offset, &nframes);

	state = psnd_pcm_state (sc->handle);

	switch (state) {
		case SND_PCM_STATE_PREPARED:
			psnd_pcm_mmap_commit (sc->handle, offset, nframes);
			psnd_pcm_start (sc->handle);
			break;
		case SND_PCM_STATE_RUNNING:
			psnd_pcm_mmap_commit (sc->handle, offset, nframes);
			break;
		default:
			break;
	}
}
static unsigned int ALSA_RW_GetDMAPos (soundcardinfo_t *sc)
{
	int frames;
	frames = psnd_pcm_avail_update(sc->handle);
	if (frames >= 0)
	{
		sc->sn.samplepos = (sc->snd_sent + frames) * sc->sn.numchannels;
	}
	return sc->sn.samplepos;
}
static void ALSA_RW_Submit (soundcardinfo_t *sc, int start, int end)
{
	int			state;
        unsigned int frames, offset, ringsize;
        unsigned chunk;
        int result;
	int stride = sc->sn.numchannels * (sc->sn.samplebits/8);

        /*we can't change the data that was already written*/
        frames = end - sc->snd_sent;
        if (!frames)
                return;

	state = psnd_pcm_state (sc->handle);

        ringsize = sc->sn.samples / sc->sn.numchannels;

        chunk = frames;
        offset = sc->snd_sent % ringsize;

        if (offset + chunk >= ringsize)
                chunk = ringsize - offset;
	result = psnd_pcm_writei(sc->handle, sc->sn.buffer + offset*stride, chunk);
        if (result < chunk)
        {
                if (result >= 0)
                        sc->snd_sent += result;
                return;
        }
        sc->snd_sent += chunk;

        chunk = frames - chunk;
        if (chunk)
        {
		result = psnd_pcm_writei(sc->handle, sc->sn.buffer, chunk);
                if (result > 0)
                        sc->snd_sent += result;
        }

	if (state == SND_PCM_STATE_PREPARED)
		psnd_pcm_start (sc->handle);
}

static void ALSA_Shutdown (soundcardinfo_t *sc)
{
	psnd_pcm_close (sc->handle);

	if (sc->Submit == ALSA_RW_Submit)
		free(sc->sn.buffer);
}

static void *ALSA_LockBuffer(soundcardinfo_t *sc, unsigned int *sampidx)
{
	return sc->sn.buffer;
}

static void ALSA_UnlockBuffer(soundcardinfo_t *sc, void *buffer)
{
}

static void ALSA_SetUnderWater(soundcardinfo_t *sc, qboolean underwater)
{
}

static qboolean Alsa_InitAlsa(void)
{
	static qboolean tried;
	static qboolean alsaworks;
	if (tried)
		return alsaworks;
	tried = true;

	// Try alternative names of libasound, sometimes it is not linked correctly.
	alsasharedobject = dlopen("libasound.so.2", RTLD_LAZY|RTLD_LOCAL);
	if (!alsasharedobject)
	{
		alsasharedobject = dlopen("libasound.so", RTLD_LAZY|RTLD_LOCAL);
		if (!alsasharedobject)
		{
			return false;
		}
	}


	psnd_pcm_open				= dlsym(alsasharedobject, "snd_pcm_open");
	psnd_pcm_close				= dlsym(alsasharedobject, "snd_pcm_close");
	psnd_strerror				= dlsym(alsasharedobject, "snd_strerror");
	psnd_pcm_hw_params_any			= dlsym(alsasharedobject, "snd_pcm_hw_params_any");
	psnd_pcm_hw_params_set_access		= dlsym(alsasharedobject, "snd_pcm_hw_params_set_access");
	psnd_pcm_hw_params_set_format		= dlsym(alsasharedobject, "snd_pcm_hw_params_set_format");
	psnd_pcm_hw_params_set_channels		= dlsym(alsasharedobject, "snd_pcm_hw_params_set_channels");
	psnd_pcm_hw_params_set_rate_near	= dlsym(alsasharedobject, "snd_pcm_hw_params_set_rate_near");
	psnd_pcm_hw_params_set_period_size_near	= dlsym(alsasharedobject, "snd_pcm_hw_params_set_period_size_near");
	psnd_pcm_hw_params			= dlsym(alsasharedobject, "snd_pcm_hw_params");
	psnd_pcm_sw_params_current		= dlsym(alsasharedobject, "snd_pcm_sw_params_current");
	psnd_pcm_sw_params_set_start_threshold	= dlsym(alsasharedobject, "snd_pcm_sw_params_set_start_threshold");
	psnd_pcm_sw_params_set_stop_threshold	= dlsym(alsasharedobject, "snd_pcm_sw_params_set_stop_threshold");
	psnd_pcm_sw_params			= dlsym(alsasharedobject, "snd_pcm_sw_params");
	psnd_pcm_hw_params_get_buffer_size	= dlsym(alsasharedobject, "snd_pcm_hw_params_get_buffer_size");
	psnd_pcm_avail_update			= dlsym(alsasharedobject, "snd_pcm_avail_update");
	psnd_pcm_state				= dlsym(alsasharedobject, "snd_pcm_state");
	psnd_pcm_start				= dlsym(alsasharedobject, "snd_pcm_start");
	psnd_pcm_hw_params_sizeof		= dlsym(alsasharedobject, "snd_pcm_hw_params_sizeof");
	psnd_pcm_sw_params_sizeof		= dlsym(alsasharedobject, "snd_pcm_sw_params_sizeof");
	psnd_pcm_hw_params_set_buffer_size_near = dlsym(alsasharedobject, "snd_pcm_hw_params_set_buffer_size_near");

	psnd_pcm_mmap_begin			= dlsym(alsasharedobject, "snd_pcm_mmap_begin");
	psnd_pcm_mmap_commit			= dlsym(alsasharedobject, "snd_pcm_mmap_commit");

	psnd_pcm_writei				= dlsym(alsasharedobject, "snd_pcm_writei");
	psnd_pcm_prepare			= dlsym(alsasharedobject, "snd_pcm_prepare");

	alsaworks = psnd_pcm_open
		&& psnd_pcm_close
		&& psnd_strerror
		&& psnd_pcm_hw_params_any
		&& psnd_pcm_hw_params_set_access
		&& psnd_pcm_hw_params_set_format
		&& psnd_pcm_hw_params_set_channels
		&& psnd_pcm_hw_params_set_rate_near
		&& psnd_pcm_hw_params_set_period_size_near
		&& psnd_pcm_hw_params
		&& psnd_pcm_sw_params_current
		&& psnd_pcm_sw_params_set_start_threshold
		&& psnd_pcm_sw_params_set_stop_threshold
		&& psnd_pcm_sw_params
		&& psnd_pcm_hw_params_get_buffer_size
		&& psnd_pcm_avail_update
		&& psnd_pcm_state
		&& psnd_pcm_start
		&& psnd_pcm_hw_params_sizeof
		&& psnd_pcm_sw_params_sizeof
		&& psnd_pcm_hw_params_set_buffer_size_near
		&& psnd_pcm_mmap_begin
		&& psnd_pcm_mmap_commit
		&& psnd_pcm_writei && psnd_pcm_prepare
		;

	return alsaworks;
}

static int ALSA_InitCard (soundcardinfo_t *sc, int cardnum)
{
	snd_pcm_t   *pcm;
	snd_pcm_uframes_t buffer_size;

	soundcardinfo_t *ec;	//existing card
	char *pcmname;
	cvar_t *devname;

	int					 err;
	int					 bps, stereo;
	unsigned int		 rate;
	snd_pcm_hw_params_t	*hw;
	snd_pcm_sw_params_t	*sw;
	snd_pcm_uframes_t	 frag_size;
	qboolean mmap = false;

	if (!Alsa_InitAlsa())
	{
		Con_Printf(CON_ERROR "Alsa does not appear to be installed or compatible\n");
		return 2;
	}

	hw = alloca(psnd_pcm_hw_params_sizeof());
	sw = alloca(psnd_pcm_sw_params_sizeof());
	memset(sw, 0, psnd_pcm_sw_params_sizeof());
	memset(hw, 0, psnd_pcm_hw_params_sizeof());

//WARNING: 'default' as the default sucks arse. it adds about a second's worth of lag.
	devname = Cvar_Get(va("snd_alsadevice%i", cardnum+1), (cardnum==0?"hw":(cardnum==1?"default":"")), 0, "Sound controls");
	pcmname = devname->string;

	if (!*pcmname)
		return 2;

	for (ec = sndcardinfo; ec; ec = ec->next)
		if (!strcmp(ec->name, pcmname))
			break;
	if (ec)
		return 2;	//no more

	sc->inactive_sound = true;	//linux sound devices always play sound, even when we're not the active app...

	Con_Printf("Initing ALSA sound device \"%s\"\n", pcmname);

	err = psnd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK,
						  SND_PCM_NONBLOCK);
	if (0 > err)
	{
		Con_Printf (CON_ERROR "Error: open error: %s\n", psnd_strerror (err));
		return 0;
	}
	Con_Printf ("ALSA: Using PCM %s.\n", pcmname);

	err = psnd_pcm_hw_params_any (pcm, hw);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: error setting hw_params_any. %s\n",
					psnd_strerror (err));
		goto error;
	}

	err = psnd_pcm_hw_params_set_access (pcm, hw,  mmap?SND_PCM_ACCESS_MMAP_INTERLEAVED:SND_PCM_ACCESS_RW_INTERLEAVED);
	if (0 > err)
	{
		Con_Printf (CON_ERROR "ALSA: Failure to set interleaved PCM access. %s\n",
					psnd_strerror (err));
		goto error;
	}

	// get sample bit size
	bps = sc->sn.samplebits;
	{
		snd_pcm_format_t spft;
		if (bps == 16)
			spft = SND_PCM_FORMAT_S16;
		else
			spft = SND_PCM_FORMAT_U8;

		err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
		while (err < 0)
		{
			if (spft == SND_PCM_FORMAT_S16)
			{
				bps = 8;
				spft = SND_PCM_FORMAT_U8;
			}
			else
			{
				Con_Printf (CON_ERROR "ALSA: no usable formats. %s\n", psnd_strerror (err));
				goto error;
			}
			err = psnd_pcm_hw_params_set_format (pcm, hw, spft);
		}
	}

	// get speaker channels
	stereo = sc->sn.numchannels;
	err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
	while (err < 0)
	{
		if (stereo > 2)
			stereo = 2;
		else if (stereo > 1)
			stereo = 1;
		else
		{
			Con_Printf (CON_ERROR "ALSA: no usable number of channels. %s\n", psnd_strerror (err));
			goto error;
		}
		err = psnd_pcm_hw_params_set_channels (pcm, hw, stereo);
	}

	// get rate
	rate = sc->sn.speed;
	err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
	while (err < 0)
	{
		if (rate > 48000)
			rate = 48000;
		else if (rate > 44100)
			rate = 44100;
		else if (rate > 22150)
			rate = 22150;
		else if (rate > 11025)
			rate = 11025;
		else if (rate > 800)
			rate = 800;
		else
		{
			Con_Printf (CON_ERROR "ALSA: no usable rates. %s\n", psnd_strerror (err));
			goto error;
		}
		err = psnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
	}

	if (rate > 11025)
		frag_size = 8 * bps * rate / 11025;
	else
		frag_size = 8 * bps;

	err = psnd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to set period size near %i. %s\n",
					(int) frag_size, psnd_strerror (err));
		goto error;
	}
	err = psnd_pcm_hw_params (pcm, hw);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to install hw params: %s\n",
					psnd_strerror (err));
		goto error;
	}
	err = psnd_pcm_sw_params_current (pcm, sw);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to determine current sw params. %s\n",
					psnd_strerror (err));
		goto error;
	}
	err = psnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to set playback threshold. %s\n",
					psnd_strerror (err));
		goto error;
	}
	err = psnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to set playback stop threshold. %s\n",
					psnd_strerror (err));
		goto error;
	}
	err = psnd_pcm_sw_params (pcm, sw);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to install sw params. %s\n",
					psnd_strerror (err));
		goto error;
	}

	sc->sn.numchannels = stereo;
	sc->sn.samplepos = 0;
	sc->sn.samplebits = bps;

	buffer_size = sc->sn.samples / stereo;
	if (buffer_size)
	{
		err = psnd_pcm_hw_params_set_buffer_size_near(pcm, hw, &buffer_size);
		if (err < 0)
		{
			Con_Printf (CON_ERROR "ALSA: unable to set buffer size. %s\n", psnd_strerror (err));
			goto error;
		}
	}

	err = psnd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
	if (0 > err) {
		Con_Printf (CON_ERROR "ALSA: unable to get buffer size. %s\n",
					psnd_strerror (err));
		goto error;
	}

	sc->sn.samples = buffer_size * sc->sn.numchannels;		// mono samples in buffer
	sc->sn.speed = rate;
	sc->handle = pcm;

	sc->Lock		= ALSA_LockBuffer;
	sc->Unlock		= ALSA_UnlockBuffer;
	sc->SetWaterDistortion	= ALSA_SetUnderWater;
	sc->Shutdown		= ALSA_Shutdown;
	if (mmap)
	{
		sc->GetDMAPos	= ALSA_MMap_GetDMAPos;
		sc->Submit	= ALSA_MMap_Submit;
		sc->GetDMAPos(sc);		// sets shm->buffer

		//alsa doesn't seem to like high mixahead values
		//(maybe it tells us above somehow...)
		//so force it lower
		//quake's default of 0.2 was for 10fps rendering anyway
		//so force it down to 0.1 which is the default for halflife at least, and should give better latency
		{
			extern cvar_t _snd_mixahead;
			if (_snd_mixahead.value >= 0.2)
			{
				Con_Printf("Alsa Hack: _snd_mixahead forced lower\n");
				_snd_mixahead.value = 0.1;
			}
		}
	}
	else
	{
		sc->GetDMAPos	= ALSA_RW_GetDMAPos;
		sc->Submit	= ALSA_RW_Submit;

		sc->samplequeue = sc->sn.samples;
		sc->sn.buffer = malloc(sc->sn.samples * (sc->sn.samplebits/8));

		err = psnd_pcm_prepare(pcm);
		if (0 > err)
		{
			Con_Printf (CON_ERROR "ALSA: unable to prepare for use. %s\n",
					psnd_strerror (err));
			goto error;
		}
	}

	return true;

error:
	psnd_pcm_close (pcm);
	return false;
}

int (*pALSA_InitCard) (soundcardinfo_t *sc, int cardnum) = &ALSA_InitCard;