qf
/
quakeforge
mirror of https://git.code.sf.net/p/quake/quakeforge
0
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
quakeforge/libs/audio/targets/snd_alsa.c

735 lines
17 KiB
C
Raw Blame History

/*
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
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdio.h>
#include <dlfcn.h>
#include <alsa/asoundlib.h>
#include "QF/cvar.h"
#include "QF/qargs.h"
#include "QF/sys.h"
#include "snd_internal.h"
typedef struct {
const snd_pcm_channel_area_t *areas;
snd_pcm_uframes_t offset;
snd_pcm_uframes_t nframes;
} alsa_pkt_t;
static int snd_inited;
static int snd_blocked = 0;
static snd_pcm_uframes_t buffer_size;
static void *alsa_handle;
static snd_pcm_t *pcm;
static snd_async_handler_t *async_handler;
static cvar_t *snd_bits;
static cvar_t *snd_device;
static cvar_t *snd_rate;
static cvar_t *snd_stereo;
//FIXME xfer probably should not be touching this (such data should probably
//come through snd_t)
static snd_output_data_t plugin_info_snd_output_data;
#define QF_ALSA_NEED(ret, func, params) \
static ret (*qf##func) params;
#include "alsa_funcs_list.h"
#undef QF_ALSA_NEED
static qboolean
load_libasound (void)
{
if (!(alsa_handle = dlopen ("libasound.so.2", RTLD_GLOBAL | RTLD_NOW))) {
Sys_Printf ("Couldn't load libasound.so.2: %s\n", dlerror ());
return false;
}
#define QF_ALSA_NEED(ret, func, params) \
if (!(qf##func = dlsym (alsa_handle, #func))) { \
Sys_Printf ("Couldn't load ALSA function %s\n", #func); \
dlclose (alsa_handle); \
alsa_handle = 0; \
return false; \
}
#include "alsa_funcs_list.h"
#undef QF_ALSA_NEED
return true;
}
#define snd_pcm_hw_params_sizeof qfsnd_pcm_hw_params_sizeof
#define snd_pcm_sw_params_sizeof qfsnd_pcm_sw_params_sizeof
static void
SNDDMA_Init_Cvars (void)
{
snd_stereo = Cvar_Get ("snd_stereo", "1", CVAR_ROM, NULL,
"sound stereo output");
snd_rate = Cvar_Get ("snd_rate", "0", CVAR_ROM, NULL,
"sound playback rate. 0 is system default");
snd_device = Cvar_Get ("snd_device", "", CVAR_ROM, NULL,
"sound device. \"\" is system default");
snd_bits = Cvar_Get ("snd_bits", "0", CVAR_ROM, NULL,
"sound sample depth. 0 is system default");
}
static __attribute__((const)) snd_pcm_uframes_t
round_buffer_size (snd_pcm_uframes_t sz)
{
snd_pcm_uframes_t mask = ~0;
while (sz & mask) {
sz &= mask;
mask <<= 1;
}
return sz;
}
static inline int
clamp_16 (int val)
{
if (val > 0x7fff)
val = 0x7fff;
else if (val < -0x8000)
val = -0x8000;
return val;
}
static inline int
clamp_8 (int val)
{
if (val > 0x7f)
val = 0x7f;
else if (val < -0x80)
val = -0x80;
return val;
}
static void
alsa_ni_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count,
float volume)
{
const snd_pcm_channel_area_t *areas;
int out_idx, out_max;
float *p;
areas = snd->xfer_data;
p = (float *) paintbuffer;
out_max = snd->frames - 1;
out_idx = *plugin_info_snd_output_data.paintedtime;
while (out_idx > out_max)
out_idx -= out_max + 1;
if (snd->samplebits == 16) {
short *out_0 = (short *) areas[0].addr;
short *out_1 = (short *) areas[1].addr;
if (snd->channels == 2) {
while (count--) {
out_0[out_idx] = clamp_16 ((*p++ * volume) * 0x8000);
out_1[out_idx] = clamp_16 ((*p++ * volume) * 0x8000);
if (out_idx++ > out_max)
out_idx = 0;
}
} else {
while (count--) {
out_0[out_idx] = clamp_16 ((*p++ * volume) * 0x8000);
p++; // skip right channel
if (out_idx++ > out_max)
out_idx = 0;
}
}
} else if (snd->samplebits == 8) {
byte *out_0 = (byte *) areas[0].addr;
byte *out_1 = (byte *) areas[1].addr;
if (snd->channels == 2) {
while (count--) {
out_0[out_idx] = clamp_8 ((*p++ * volume) * 0x80);
out_1[out_idx] = clamp_8 ((*p++ * volume) * 0x80);
if (out_idx++ > out_max)
out_idx = 0;
}
} else {
while (count--) {
out_0[out_idx] = clamp_8 ((*p++ * volume) * 0x8000);
p++; // skip right channel
if (out_idx++ > out_max)
out_idx = 0;
}
}
}
}
static void
alsa_xfer (snd_t *snd, portable_samplepair_t *paintbuffer, int count,
float volume)
{
int out_idx, out_max, step, val;
float *p;
alsa_pkt_t *packet = snd->xfer_data;;
p = (float *) paintbuffer;
count *= snd->channels;
out_max = (snd->frames * snd->channels) - 1;
out_idx = snd->paintedtime * snd->channels;
while (out_idx > out_max)
out_idx -= out_max + 1;
step = 3 - snd->channels;
if (snd->samplebits == 16) {
short *out = (short *) packet->areas[0].addr;
while (count--) {
val = (*p * volume) * 0x8000;
p += step;
if (val > 0x7fff)
val = 0x7fff;
else if (val < -0x8000)
val = -0x8000;
out[out_idx++] = val;
if (out_idx > out_max)
out_idx = 0;
}
} else if (snd->samplebits == 8) {
unsigned char *out = (unsigned char *) packet->areas[0].addr;
while (count--) {
val = (*p * volume) * 128;
p += step;
if (val > 0x7f)
val = 0x7f;
else if (val < -0x80)
val = -0x80;
out[out_idx++] = val + 0x80;
if (out_idx > out_max)
out_idx = 0;
}
}
}
static int
alsa_recover (snd_pcm_t *pcm, int err)
{
if (err == -EPIPE) {
Sys_Printf ("snd_alsa: xrun\n");
// xrun
if ((err = qfsnd_pcm_prepare (pcm)) < 0) {
Sys_MaskPrintf (SYS_snd, "snd_alsa: recover from xrun failed: %s\n",
qfsnd_strerror (err));
return err;
}
return 0;
} else if (err == -ESTRPIPE) {
Sys_Printf ("snd_alsa: suspend\n");
// suspend
while ((err = qfsnd_pcm_resume(pcm)) == -EAGAIN) {
usleep (20 * 1000);
}
if (err < 0 && (err = qfsnd_pcm_prepare (pcm)) < 0) {
Sys_MaskPrintf (SYS_snd,
"snd_alsa: recover from suspend failed: %s\n",
qfsnd_strerror (err));
return err;
}
return 0;
}
return err;
}
static int
alsa_process (snd_pcm_t *pcm, snd_t *snd)
{
alsa_pkt_t packet;
int res;
int ret = 1;
snd_pcm_uframes_t size = snd->submission_chunk;
snd->xfer_data = &packet;
while (size > 0) {
packet.nframes = size;
if ((res = qfsnd_pcm_mmap_begin (pcm, &packet.areas, &packet.offset,
&packet.nframes)) < 0) {
if ((res = alsa_recover (pcm, -EPIPE)) < 0) {
Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n",
qfsnd_strerror (res));
return res;
}
ret = 0;
}
snd->buffer = packet.areas[0].addr;
snd->paint_channels (snd, snd->paintedtime + packet.nframes);
if ((res = qfsnd_pcm_mmap_commit (pcm, packet.offset,
packet.nframes)) < 0
|| (snd_pcm_uframes_t) res != packet.nframes) {
if ((res = alsa_recover (pcm, res >= 0 ? -EPIPE : res)) < 0) {
Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n",
qfsnd_strerror (res));
return res;
}
ret = 0;
}
size -= packet.nframes;
}
return ret;
}
static void
alsa_callback (snd_async_handler_t *handler)
{
snd_pcm_t *pcm = qfsnd_async_handler_get_pcm (handler);
snd_t *snd = qfsnd_async_handler_get_callback_private (handler);
int res;
int avail;
int first = 0;
while (1) {
snd_pcm_state_t state = qfsnd_pcm_state (pcm);
if (state == SND_PCM_STATE_XRUN) {
if ((res = alsa_recover (pcm, -EPIPE)) < 0) {
Sys_Printf ("snd_alsa: XRUN recovery failed: %s\n",
qfsnd_strerror (res));
//FIXME disable/restart sound
return;
}
} else if (state == SND_PCM_STATE_SUSPENDED) {
if ((res = alsa_recover (pcm, -EPIPE)) < 0) {
Sys_Printf ("snd_alsa: suspend recovery failed: %s\n",
qfsnd_strerror (res));
//FIXME disable/restart sound
return;
}
}
if ((avail = qfsnd_pcm_avail_update (pcm)) < 0) {
if ((res = alsa_recover (pcm, -EPIPE)) < 0) {
Sys_Printf ("snd_alsa: avail update failed: %s\n",
qfsnd_strerror (res));
//FIXME disable/restart sound
return;
}
first = 1;
continue;
}
if (avail < snd->submission_chunk) {
if (first) {
first = 0;
if ((res = qfsnd_pcm_start (pcm)) < 0) {
Sys_Printf ("snd_alsa: start failed: %s\n",
qfsnd_strerror (res));
return;
}
continue;
}
break;
}
if ((res = alsa_process (pcm, snd))) {
if (res < 0) {
//FIXME disable/restart sound
return;
}
break;
}
first = 1;
}
}
static int
alsa_open_playback (snd_t *snd, const char *device)
{
if (!*device) {
device = "default";
}
Sys_Printf ("Using PCM %s.\n", device);
int res = qfsnd_pcm_open (&pcm, device, SND_PCM_STREAM_PLAYBACK,
SND_PCM_NONBLOCK);
if (res < 0) {
Sys_Printf ("snd_alsa: audio open error: %s\n", qfsnd_strerror (res));
return 0;
}
return 1;
}
static int
alsa_playback_set_mmap (snd_t *snd, snd_pcm_hw_params_t *hw)
{
int res;
res = qfsnd_pcm_hw_params_set_access (pcm, hw,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (res == 0) {
snd->xfer = alsa_xfer;
return 1;
}
Sys_MaskPrintf (SYS_snd, "snd_alsa: Failure to set interleaved PCM "
"access. (%d) %s\n", res, qfsnd_strerror (res));
res = qfsnd_pcm_hw_params_set_access (pcm, hw,
SND_PCM_ACCESS_MMAP_NONINTERLEAVED);
if (res == 0) {
snd->xfer = alsa_ni_xfer;
return 1;
}
Sys_MaskPrintf (SYS_snd, "snd_alsa: Failure to set noninterleaved PCM "
"access. (%d) %s\n", res, qfsnd_strerror (res));
Sys_Printf ("snd_alsa: could not set mmap access\n");
return 0;
}
static int
alsa_playback_set_bps (snd_t *snd, snd_pcm_hw_params_t *hw)
{
int res;
int bps = 0;
if (snd_bits->int_val == 16) {
bps = SND_PCM_FORMAT_S16_LE;
snd->samplebits = 16;
} else if (snd_bits->int_val == 8) {
bps = SND_PCM_FORMAT_U8;
snd->samplebits = 8;
} else if (snd_bits->int_val) {
Sys_Printf ("snd_alsa: invalid sample bits: %d\n", bps);
return 0;
}
if (bps) {
if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) {
return 1;
}
} else {
bps = SND_PCM_FORMAT_S16_LE;
if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) {
snd->samplebits = 16;
return 1;
}
bps = SND_PCM_FORMAT_U8;
if ((res = qfsnd_pcm_hw_params_set_format (pcm, hw, bps)) == 0) {
snd->samplebits = 8;
return 1;
}
Sys_Printf ("snd_alsa: no usable formats. %s\n", qfsnd_strerror (res));
}
snd->samplebits = -1;
Sys_Printf ("snd_alsa: desired format not supported\n");
return 0;
}
static int
alsa_playback_set_channels (snd_t *snd, snd_pcm_hw_params_t *hw)
{
int res;
int channels = 1;
if (snd_stereo->int_val) {
channels = 2;
}
if ((res = qfsnd_pcm_hw_params_set_channels (pcm, hw, channels)) == 0) {
snd->channels = channels;
return 1;
}
Sys_Printf ("snd_alsa: desired channels not supported\n");
return 0;
}
static int
alsa_playback_set_rate (snd_t *snd, snd_pcm_hw_params_t *hw)
{
int res;
unsigned rate = 0;
static int default_rates[] = { 48000, 44100, 22050, 11025, 0 };
if (snd_rate->int_val) {
rate = snd_rate->int_val;
}
if (rate) {
if ((res = qfsnd_pcm_hw_params_set_rate (pcm, hw, rate, 0)) == 0) {
snd->speed = rate;
return 1;
}
Sys_Printf ("snd_alsa: desired rate %i not supported. %s\n", rate,
qfsnd_strerror (res));
} else {
// use default rate
int dir = 0;
for (int *def_rate = default_rates; *def_rate; def_rate++) {
rate = *def_rate;
res = qfsnd_pcm_hw_params_set_rate_near (pcm, hw, &rate, &dir);
if (res == 0) {
snd->speed = rate;
return 1;
}
}
Sys_Printf ("snd_alsa: no usable rate\n");
}
return 0;
}
static int
alsa_playback_set_period_size (snd_t *snd, snd_pcm_hw_params_t *hw)
{
int res;
snd_pcm_uframes_t period_size;
// works out to about 5.5ms (5.3 for 48k, 5.8 for 44k1) but consistent for
// different sample rates give or take rounding
period_size = 64 * (snd->speed / 11025);
res = qfsnd_pcm_hw_params_set_period_size_near (pcm, hw, &period_size, 0);
if (res == 0) {
// don't mix less than this in frames:
res = qfsnd_pcm_hw_params_get_period_size (hw, &period_size, 0);
if (res == 0) {
snd->submission_chunk = period_size;
return 1;
}
Sys_Printf ("snd_alsa: unable to get period size. %s\n",
qfsnd_strerror (res));
} else {
Sys_Printf ("snd_alsa: unable to set period size near %i. %s\n",
(int) period_size, qfsnd_strerror (res));
}
return 0;
}
static int
SNDDMA_Init (snd_t *snd)
{
int res;
const char *device = snd_device->string;
snd_pcm_hw_params_t *hw;
snd_pcm_sw_params_t *sw;
if (!load_libasound ())
return false;
snd_pcm_hw_params_alloca (&hw);
snd_pcm_sw_params_alloca (&sw);
while (1) {
if (!alsa_open_playback (snd, device)) {
return 0;
}
if ((res = qfsnd_pcm_hw_params_any (pcm, hw)) < 0) {
Sys_Printf ("snd_alsa: error setting hw_params_any. %s\n",
qfsnd_strerror (res));
goto error;
}
if (alsa_playback_set_mmap (snd, hw)) {
break;
}
if (*device) {
goto error;
}
qfsnd_pcm_close (pcm);
device = "plughw";
}
if (!alsa_playback_set_bps (snd, hw)) {
goto error;
}
if (!alsa_playback_set_channels (snd, hw)) {
goto error;
}
if (!alsa_playback_set_rate (snd, hw)) {
goto error;
}
if (!alsa_playback_set_period_size (snd, hw)) {
goto error;
}
if ((res = qfsnd_pcm_hw_params (pcm, hw)) < 0) {
Sys_Printf ("snd_alsa: unable to install hw params: %s\n",
qfsnd_strerror (res));
goto error;
}
if ((res = qfsnd_pcm_sw_params_current (pcm, sw)) < 0) {
Sys_Printf ("snd_alsa: unable to determine current sw params. %s\n",
qfsnd_strerror (res));
goto error;
}
if ((res = qfsnd_pcm_sw_params_set_start_threshold (pcm, sw, ~0U)) < 0) {
Sys_Printf ("snd_alsa: unable to set playback threshold. %s\n",
qfsnd_strerror (res));
goto error;
}
if ((res = qfsnd_pcm_sw_params_set_stop_threshold (pcm, sw, ~0U)) < 0) {
Sys_Printf ("snd_alsa: unable to set playback stop threshold. %s\n",
qfsnd_strerror (res));
goto error;
}
if ((res = qfsnd_pcm_sw_params (pcm, sw)) < 0) {
Sys_Printf ("snd_alsa: unable to install sw params. %s\n",
qfsnd_strerror (res));
goto error;
}
snd->framepos = 0;
if ((res = qfsnd_pcm_hw_params_get_buffer_size (hw, &buffer_size)) < 0) {
Sys_Printf ("snd_alsa: unable to get buffer size. %s\n",
qfsnd_strerror (res));
goto error;
}
if (buffer_size != round_buffer_size (buffer_size)) {
Sys_Printf ("snd_alsa: WARNING: non-power of 2 buffer size. sound may be unsatisfactory\n");
Sys_Printf ("recommend using either the plughw, or hw devices or adjusting dmix\n");
Sys_Printf ("to have a power of 2 buffer size\n");
}
if ((res = qfsnd_async_add_pcm_handler (&async_handler, pcm,
alsa_callback, snd)) < 0) {
Sys_Printf ("snd_alsa: unable to register async handler: %s",
qfsnd_strerror (res));
goto error;
}
snd->frames = buffer_size;
// send the first period to fill the buffer
// also sets snd->buffer
if (alsa_process (pcm, snd) < 0) {
goto error;
}
qfsnd_pcm_start (pcm);
Sys_Printf ("%5d channels %sinterleaved\n", snd->channels,
snd->xfer ? "non-" : "");
Sys_Printf ("%5d samples (%.1fms)\n", snd->frames,
1000.0 * snd->frames / snd->speed);
Sys_Printf ("%5d samplepos\n", snd->framepos);
Sys_Printf ("%5d samplebits\n", snd->samplebits);
Sys_Printf ("%5d submission_chunk (%.1fms)\n", snd->submission_chunk,
1000.0 * snd->submission_chunk / snd->speed);
Sys_Printf ("%5d speed\n", snd->speed);
Sys_Printf ("0x%lx dma buffer\n", (long) snd->buffer);
snd_inited = 1;
return 1;
error:
qfsnd_pcm_close (pcm);
snd->channels = 0;
snd->frames = 0;
snd->samplebits = 0;
snd->submission_chunk = 0;
snd->speed = 0;
return 0;
}
static void
SNDDMA_shutdown (snd_t *snd)
{
if (snd_inited) {
qfsnd_async_del_handler (async_handler);
async_handler = 0;
qfsnd_pcm_close (pcm);
snd_inited = 0;
}
}
static void
SNDDMA_BlockSound (snd_t *snd)
{
if (snd_inited && ++snd_blocked == 1)
qfsnd_pcm_pause (pcm, 1);
}
static void
SNDDMA_UnblockSound (snd_t *snd)
{
if (!snd_inited || !snd_blocked)
return;
if (!--snd_blocked)
qfsnd_pcm_pause (pcm, 0);
}
static general_data_t plugin_info_general_data = {
};
static general_funcs_t plugin_info_general_funcs = {
.init = SNDDMA_Init_Cvars,
.shutdown = NULL,
};
static snd_output_data_t plugin_info_snd_output_data = {
.model = som_pull,
};
static snd_output_funcs_t plugin_info_snd_output_funcs = {
.init = SNDDMA_Init,
.shutdown = SNDDMA_shutdown,
.block_sound = SNDDMA_BlockSound,
.unblock_sound = SNDDMA_UnblockSound,
};
static plugin_data_t plugin_info_data = {
.general = &plugin_info_general_data,
.snd_output = &plugin_info_snd_output_data,
};
static plugin_funcs_t plugin_info_funcs = {
.general = &plugin_info_general_funcs,
.snd_output = &plugin_info_snd_output_funcs,
};
static plugin_t plugin_info = {
.type = qfp_snd_output,
.api_version = QFPLUGIN_VERSION,
.plugin_version = "0.1",
.description = "ALSA digital output",
.copyright = "Copyright (C) 1996-1997 id Software, Inc.\n"
"Copyright (C) 1999,2000,2001 contributors of the QuakeForge "
"project\n"
"Please see the file \"AUTHORS\" for a list of contributors",
.functions = &plugin_info_funcs,
.data = &plugin_info_data,
};
PLUGIN_INFO(snd_output, alsa)
{
return &plugin_info;
}
Reference in New Issue View Git Blame Copy Permalink