quakespasm/Quake/snd_alsa.c

333 lines
9.0 KiB
C

/*
snd_alsa.c
ALSA 1.0 sound driver for Linux Hexen II
Copyright (C) 1999,2004 contributors of the QuakeForge project
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
*/
#include "quakedef.h"
#include <alsa/asoundlib.h>
#define NB_PERIODS 4
//static const char alsa_default[] = "hw:0,0";
//static const char alsa_default[] = "plughw:0";
static const char alsa_default[] = "default";
static const char *pcmname = alsa_default;
static snd_pcm_t *pcm = NULL;
static snd_pcm_uframes_t buffer_size;
static const int tryrates[] = { 11025, 22050, 44100, 48000, 16000, 24000, 8000 };
static const int MAX_TRYRATES = sizeof(tryrates)/sizeof(tryrates[0]);
#if defined(__GNUC__) && \
!(defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L)
#define ALSA_CHECK_ERR(check, fmt, args...) \
do { \
if (check < 0) { \
Con_Printf ("ALSA: " fmt, ##args); \
goto error; \
} \
} while (0)
#else
#define ALSA_CHECK_ERR(check, ...) \
do { \
if (check < 0) { \
Con_Printf ("ALSA: " __VA_ARGS__); \
goto error; \
} \
} while (0)
#endif
qboolean SNDDMA_Init (dma_t *dma)
{
int i, err;
unsigned int rate;
int tmp_bits, tmp_chan;
snd_pcm_hw_params_t *hw = NULL;
snd_pcm_sw_params_t *sw = NULL;
snd_pcm_uframes_t frag_size;
i = COM_CheckParm("-alsadev");
if (i != 0 && i < com_argc - 1)
pcmname = com_argv[i + 1];
err = snd_pcm_open (&pcm, pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (err < 0)
{
Con_Printf ("ALSA: error opening device \"%s\": %s\n", pcmname, snd_strerror(err));
return false;
}
Con_Printf ("ALSA: Using device: %s\n", pcmname);
err = snd_pcm_hw_params_malloc (&hw);
ALSA_CHECK_ERR(err, "unable to allocate hardware params. %s\n", snd_strerror(err));
err = snd_pcm_hw_params_any (pcm, hw);
ALSA_CHECK_ERR(err, "unable to init hardware params. %s\n", snd_strerror(err));
err = snd_pcm_hw_params_set_access (pcm, hw, SND_PCM_ACCESS_MMAP_INTERLEAVED);
ALSA_CHECK_ERR(err, "unable to set interleaved access. %s\n", snd_strerror(err));
i = (loadas8bit.value) ? 8 : 16;
tmp_bits = (i == 8) ? SND_PCM_FORMAT_U8 : SND_PCM_FORMAT_S16;
err = snd_pcm_hw_params_set_format (pcm, hw, (snd_pcm_format_t) tmp_bits);
if (err < 0)
{
Con_Printf ("Problems setting %d bit format, trying alternatives..\n", i);
tmp_bits = (i == 8) ? SND_PCM_FORMAT_S16 : SND_PCM_FORMAT_U8;
err = snd_pcm_hw_params_set_format (pcm, hw, (snd_pcm_format_t) tmp_bits);
ALSA_CHECK_ERR(err, "Neither 8 nor 16 bit format supported. %s\n", snd_strerror(err));
}
tmp_bits = (tmp_bits == SND_PCM_FORMAT_U8) ? 8 : 16;
i = tmp_chan = (COM_CheckParm("-sndmono") == 0) ? 2 : 1;
err = snd_pcm_hw_params_set_channels (pcm, hw, tmp_chan);
if (err < 0)
{
Con_Printf ("Problems setting channels to %s, retrying for %s\n",
(i == 2) ? "stereo" : "mono",
(i == 2) ? "mono" : "stereo");
tmp_chan = (i == 2) ? 1 : 2;
err = snd_pcm_hw_params_set_channels (pcm, hw, tmp_chan);
ALSA_CHECK_ERR(err, "unable to set desired channels. %s\n", snd_strerror(err));
}
rate = (int)sndspeed.value;
err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (err < 0)
{
Con_Printf("Problems setting sample rate, trying alternatives..\n");
for (i = 0; i < MAX_TRYRATES; i++)
{
rate = tryrates[i];
err = snd_pcm_hw_params_set_rate_near (pcm, hw, &rate, 0);
if (err < 0)
{
Con_DPrintf ("Unable to set sample rate %d\n", tryrates[i]);
rate = 0;
}
else
{
if (rate != tryrates[i])
{
Con_Printf ("Warning: Rate set (%u) didn't match requested rate (%d)!\n", rate, tryrates[i]);
// goto error;
}
break;
}
}
if (rate == 0)
{
Con_Printf ("Unable to set any sample rates.\n");
goto error;
}
}
else
{
if (rate != (int)sndspeed.value)
{
Con_Printf ("Warning: Rate set (%u) didn't match requested rate (%d)!\n", rate, (int)sndspeed.value);
// goto error;
}
}
/* pick a buffer size that is a power of 2 (by masking off low bits) */
buffer_size = i = (int)(rate * 0.15f);
while (buffer_size & (buffer_size-1))
buffer_size &= (buffer_size-1);
/* then check if it is the nearest power of 2 and bump it up if not */
if (i - buffer_size >= buffer_size >> 1)
buffer_size *= 2;
err = snd_pcm_hw_params_set_buffer_size_near (pcm, hw, &buffer_size);
ALSA_CHECK_ERR(err, "unable to set buffer size near %lu (%s)\n",
(unsigned long)buffer_size, snd_strerror(err));
err = snd_pcm_hw_params_get_buffer_size (hw, &buffer_size);
ALSA_CHECK_ERR(err, "unable to get buffer size. %s\n", snd_strerror(err));
if (buffer_size & (buffer_size-1))
{
Con_Printf ("ALSA: WARNING: non-power of 2 buffer size. sound may be\n");
Con_Printf ("unsatisfactory. Recommend using either the plughw or hw\n");
Con_Printf ("devices or adjusting dmix to have a power of 2 buf size\n");
}
/* pick a period size near the buffer_size we got from ALSA */
frag_size = buffer_size / NB_PERIODS;
err = snd_pcm_hw_params_set_period_size_near (pcm, hw, &frag_size, 0);
ALSA_CHECK_ERR(err, "unable to set period size near %i. %s\n",
(int)frag_size, snd_strerror(err));
err = snd_pcm_hw_params (pcm, hw);
ALSA_CHECK_ERR(err, "unable to install hardware params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_malloc (&sw);
ALSA_CHECK_ERR(err, "unable to allocate software params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_current (pcm, sw);
ALSA_CHECK_ERR(err, "unable to determine current software params. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U);
ALSA_CHECK_ERR(err, "unable to set playback threshold. %s\n", snd_strerror(err));
err = snd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U);
ALSA_CHECK_ERR(err, "unable to set playback stop threshold. %s\n", snd_strerror(err));
err = snd_pcm_sw_params (pcm, sw);
ALSA_CHECK_ERR(err, "unable to install software params. %s\n", snd_strerror(err));
memset ((void *) dma, 0, sizeof(dma_t));
shm = dma;
shm->channels = tmp_chan;
/*
// don't mix less than this in mono samples:
err = snd_pcm_hw_params_get_period_size (hw,
(snd_pcm_uframes_t *) (char *) (&shm->submission_chunk), 0);
ALSA_CHECK_ERR(err, "unable to get period size. %s\n", snd_strerror(err));
*/
shm->submission_chunk = 1;
shm->samplepos = 0;
shm->samplebits = tmp_bits;
Con_Printf ("ALSA: %lu bytes buffer with mmap interleaved access\n", (unsigned long)buffer_size);
shm->samples = buffer_size * shm->channels; // mono samples in buffer
shm->speed = rate;
SNDDMA_GetDMAPos (); // sets shm->buffer
snd_pcm_hw_params_free(hw);
snd_pcm_sw_params_free(sw);
return true;
error:
// full clean-up
if (hw)
snd_pcm_hw_params_free(hw);
if (sw)
snd_pcm_sw_params_free(sw);
shm = NULL;
snd_pcm_close (pcm);
pcm = NULL;
return false;
}
int SNDDMA_GetDMAPos (void)
{
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes;
const snd_pcm_channel_area_t *areas;
if (!shm)
return 0;
nframes = shm->samples/shm->channels;
snd_pcm_avail_update (pcm);
snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
// The following commit was absent in QF, causing the
// very first sound to be corrupted
snd_pcm_mmap_commit (pcm, offset, nframes);
offset *= shm->channels;
nframes *= shm->channels;
shm->samplepos = offset;
shm->buffer = (unsigned char *) areas->addr; // FIXME! there's an area per channel
return shm->samplepos;
}
void SNDDMA_Shutdown (void)
{
if (shm)
{
// full clean-up
Con_Printf ("Shutting down ALSA sound\n");
snd_pcm_drop (pcm); // do I need this?
snd_pcm_close (pcm);
pcm = NULL;
shm->buffer = NULL;
shm = NULL;
}
}
/*
==============
SNDDMA_LockBuffer
Makes sure dma buffer is valid
==============
*/
void SNDDMA_LockBuffer (void)
{
/* nothing to do here */
}
/*
==============
SNDDMA_Submit
Unlock the dma buffer /
Send sound to the device
==============
*/
void SNDDMA_Submit (void)
{
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes;
const snd_pcm_channel_area_t *areas;
int state;
int count = paintedtime - soundtime;
nframes = count / shm->channels;
snd_pcm_avail_update (pcm);
snd_pcm_mmap_begin (pcm, &areas, &offset, &nframes);
state = snd_pcm_state (pcm);
switch (state)
{
case SND_PCM_STATE_PREPARED:
snd_pcm_mmap_commit (pcm, offset, nframes);
snd_pcm_start (pcm);
break;
case SND_PCM_STATE_RUNNING:
snd_pcm_mmap_commit (pcm, offset, nframes);
break;
default:
break;
}
}
void SNDDMA_BlockSound (void)
{
snd_pcm_pause (pcm, 1);
}
void SNDDMA_UnblockSound (void)
{
snd_pcm_pause (pcm, 0);
}