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fteqw/engine/client/snd_linux.c

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#include <unistd.h>
#include <fcntl.h>
#include <stdlib.h>
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/mman.h>
#include <sys/shm.h>
#include <sys/wait.h>
#include <sys/soundcard.h>
#include <stdio.h>
#include "quakedef.h"
#ifdef __linux__
#include <sys/stat.h>
#endif
static int tryrates[] = { 11025, 22051, 44100, 8000, 48000 };
static void OSS_SetUnderWater(soundcardinfo_t *sc, qboolean underwater) //simply a stub. Any ideas how to actually implement this properly?
{
}
static unsigned int OSS_MMap_GetDMAPos(soundcardinfo_t *sc)
{
struct count_info count;
if (sc->audio_fd != -1)
{
if (ioctl(sc->audio_fd, SNDCTL_DSP_GETOPTR, &count)==-1)
{
perror("/dev/dsp");
Con_Printf("Uh, sound dead.\n");
close(sc->audio_fd);
sc->audio_fd = -1;
return 0;
}
// shm->samplepos = (count.bytes / (shm->samplebits / 8)) & (shm->samples-1);
// fprintf(stderr, "%d \r", count.ptr);
sc->sn.samplepos = count.ptr / (sc->sn.samplebits / 8);
}
return sc->sn.samplepos;
}
static void OSS_MMap_Submit(soundcardinfo_t *sc, int start, int end)
{
}
static unsigned int OSS_Alsa_GetDMAPos(soundcardinfo_t *sc)
{
struct audio_buf_info info;
unsigned int bytes;
if (ioctl (sc->audio_fd, SNDCTL_DSP_GETOSPACE, &info) != -1)
{
bytes = sc->snd_sent + info.bytes;
sc->sn.samplepos = bytes / (sc->sn.samplebits / 8);
}
return sc->sn.samplepos;
}
static void OSS_Alsa_Submit(soundcardinfo_t *sc, int start, int end)
{
unsigned int bytes, offset, ringsize;
unsigned chunk;
int result;
/*we can't change the data that was already written*/
bytes = end * sc->sn.numchannels * (sc->sn.samplebits/8);
bytes -= sc->snd_sent;
if (!bytes)
return;
ringsize = sc->sn.samples * (sc->sn.samplebits/8);
chunk = bytes;
offset = sc->snd_sent % ringsize;
if (offset + chunk >= ringsize)
chunk = ringsize - offset;
result = write(sc->audio_fd, sc->sn.buffer + offset, chunk);
if (result < chunk)
{
if (result >= 0)
sc->snd_sent += result;
printf("full?\n");
return;
}
sc->snd_sent += chunk;
chunk = bytes - chunk;
if (chunk)
{
result = write(sc->audio_fd, sc->sn.buffer, chunk);
if (result > 0)
sc->snd_sent += result;
}
}
static void OSS_Shutdown(soundcardinfo_t *sc)
{
if (sc->sn.buffer) //close it properly, so we can go and restart it later.
{
if (sc->Submit == OSS_Alsa_Submit)
free(sc->sn.buffer); /*if using alsa-compat, just free the buffer*/
else
munmap(sc->sn.buffer, sc->sn.samples * (sc->sn.samplebits/8));
}
if (sc->audio_fd != -1)
close(sc->audio_fd);
*sc->name = '\0';
}
static void *OSS_Lock(soundcardinfo_t *sc, unsigned int *sampidx)
{
return sc->sn.buffer;
}
static void OSS_Unlock(soundcardinfo_t *sc, void *buffer)
{
}
static int OSS_InitCard(soundcardinfo_t *sc, int cardnum)
{ //FIXME: implement snd_multipledevices somehow.
int rc;
int fmt;
int tmp;
int i;
struct audio_buf_info info;
int caps;
char *snddev = NULL;
cvar_t *devname;
qboolean alsadetected = false;
#ifdef __linux__
struct stat sb;
if (stat("/proc/asound", &sb) != -1)
alsadetected = true;
#endif
devname = Cvar_Get(va("snd_devicename%i", cardnum+1), cardnum?"":"/dev/dsp", 0, "Sound controls");
snddev = devname->string;
if (!*snddev)
return 2;
sc->inactive_sound = true; //linux sound devices always play sound, even when we're not the active app...
// open the sound device, confirm capability to mmap, and get size of dma buffer
Con_Printf("Initing OSS sound device %s\n", snddev);
sc->audio_fd = open(snddev, O_WRONLY | O_NONBLOCK); //try the primary device
if (sc->audio_fd < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not open %s\n", snddev);
OSS_Shutdown(sc);
return 0;
}
Q_strncpyz(sc->name, snddev, sizeof(sc->name));
//reset it
rc = ioctl(sc->audio_fd, SNDCTL_DSP_RESET, 0);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not reset %s\n", snddev);
OSS_Shutdown(sc);
return 0;
}
//check its general capabilities, we need trigger+mmap
if (ioctl(sc->audio_fd, SNDCTL_DSP_GETCAPS, &caps)==-1)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Sound driver too old\n");
OSS_Shutdown(sc);
return 0;
}
//choose channels
#ifdef SNDCTL_DSP_CHANNELS /*I'm paranoid, okay?*/
tmp = sc->sn.numchannels;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_CHANNELS, &tmp);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not set %s to channels=%d\n", snddev, sc->sn.numchannels);
OSS_Shutdown(sc);
return 0;
}
sc->sn.numchannels = tmp;
#else
tmp = 0;
if (sc->sn.numchannels == 2)
tmp = 1;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_STEREO, &tmp);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not set %s to stereo=%d\n", snddev, sc->sn.numchannels);
OSS_Shutdown(sc);
return 0;
}
if (tmp)
sc->sn.numchannels = 2;
else
sc->sn.numchannels = 1;
#endif
//choose bits
// ask the device what it supports
ioctl(sc->audio_fd, SNDCTL_DSP_GETFMTS, &fmt);
if (!(fmt & AFMT_S16_LE) && sc->sn.samplebits > 8)
sc->sn.samplebits = 8; // they asked for 16bit (the default) but their card does not support it
if (!(fmt & AFMT_U8) && sc->sn.samplebits == 8)
{ //their card doesn't support 8bit which we're trying to use.
Con_Printf(CON_ERROR "OSS: No needed sample formats supported\n");
OSS_Shutdown(sc);
return 0;
}
if (sc->sn.samplebits == 16)
{
rc = AFMT_S16_LE;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_SETFMT, &rc);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not support 16-bit data. Try 8-bit.\n");
OSS_Shutdown(sc);
return 0;
}
}
else if (sc->sn.samplebits == 8)
{
rc = AFMT_U8;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_SETFMT, &rc);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not support 8-bit data.\n");
OSS_Shutdown(sc);
return 0;
}
}
else
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: %d-bit sound not supported.\n", sc->sn.samplebits);
OSS_Shutdown(sc);
return 0;
}
//choose speed
//use the default - menu set value.
tmp = sc->sn.speed;
if (ioctl(sc->audio_fd, SNDCTL_DSP_SPEED, &tmp) != 0)
{ //humph, default didn't work. Go for random preset ones that should work.
for (i=0 ; i<sizeof(tryrates)/4 ; i++)
{
tmp = tryrates[i];
if (!ioctl(sc->audio_fd, SNDCTL_DSP_SPEED, &tmp)) break;
}
if (i == (sizeof(tryrates)/4))
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Failed to obtain a suitable rate\n");
OSS_Shutdown(sc);
return 0;
}
}
sc->sn.speed = tmp;
//figure out buffer size
if (ioctl(sc->audio_fd, SNDCTL_DSP_GETOSPACE, &info)==-1)
{
perror("GETOSPACE");
Con_Printf(CON_ERROR "OSS: Um, can't do GETOSPACE?\n");
OSS_Shutdown(sc);
return 0;
}
sc->sn.samples = info.fragstotal * info.fragsize;
sc->sn.samples /= (sc->sn.samplebits/8);
/*samples is the number of samples*channels */
// memory map the dma buffer
sc->sn.buffer = MAP_FAILED;
if (alsadetected)
{
Con_Printf("Alsa detected. Refusing to mmap.\n");
}
else if ((caps & DSP_CAP_TRIGGER) && (caps & DSP_CAP_MMAP))
{
sc->sn.buffer = (unsigned char *) mmap(NULL, sc->sn.samples*(sc->sn.samplebits/8), PROT_WRITE, MAP_FILE|MAP_SHARED, sc->audio_fd, 0);
if (sc->sn.buffer == MAP_FAILED)
{
Con_Printf("%s: device reported mmap capability, but mmap failed.\n", snddev);
if (alsadetected)
{
char *f, *n;
f = (char *)com_argv[0];
while((n = strchr(f, '/')))
f = n + 1;
Con_Printf("Your system is running alsa.\nTry: sudo echo \"%s 0 0 direct\" > /proc/asound/card0/pcm0p/oss\n", f);
}
}
}
if (sc->sn.buffer == MAP_FAILED)
{
sc->sn.buffer = NULL;
sc->samplequeue = info.bytes / (sc->sn.samplebits/8);
sc->sn.samples*=2;
sc->sn.buffer = malloc(sc->sn.samples*(sc->sn.samplebits/8));
sc->Submit = OSS_Alsa_Submit;
sc->GetDMAPos = OSS_Alsa_GetDMAPos;
}
else
{
// toggle the trigger & start her up
tmp = 0;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_SETTRIGGER, &tmp);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not toggle.\n");
OSS_Shutdown(sc);
return 0;
}
tmp = PCM_ENABLE_OUTPUT;
rc = ioctl(sc->audio_fd, SNDCTL_DSP_SETTRIGGER, &tmp);
if (rc < 0)
{
perror(snddev);
Con_Printf(CON_ERROR "OSS: Could not toggle.\n");
OSS_Shutdown(sc);
return 0;
}
sc->Submit = OSS_MMap_Submit;
sc->GetDMAPos = OSS_MMap_GetDMAPos;
}
sc->sn.samplepos = 0;
sc->Lock = OSS_Lock;
sc->Unlock = OSS_Unlock;
sc->SetWaterDistortion = OSS_SetUnderWater;
sc->Shutdown = OSS_Shutdown;
return 1;
}
int (*pOSS_InitCard) (soundcardinfo_t *sc, int cardnum) = &OSS_InitCard;
#ifdef VOICECHAT //this does apparently work after all.
#include <stdint.h>
static qboolean QDECL OSS_Capture_Enumerate (void (QDECL *callback) (const char *drivername, const char *devicecode, const char *readablename))
{
//open /dev/dsp or /dev/mixer or env("OSS_MIXERDEV") or something
//SNDCTL_SYSINFO to get sysinfo.numcards
//for i=0; i<sysinfo.numcards
//SNDCTL_CARDINFO
return false;
}
void *OSS_Capture_Init(int rate, const char *snddev)
{
int tmp;
intptr_t fd;
if (!snddev || !*snddev)
snddev = "/dev/dsp";
fd = open(snddev, O_RDONLY | O_NONBLOCK); //try the primary device
if (fd == -1)
return NULL;
#ifdef SNDCTL_DSP_CHANNELS
tmp = 1;
if (ioctl(fd, SNDCTL_DSP_CHANNELS, &tmp) != 0)
#else
tmp = 0;
if (ioctl(fd, SNDCTL_DSP_STEREO, &tmp) != 0)
#endif
{
Con_Printf("Couldn't set mono\n");
perror(snddev);
}
tmp = AFMT_S16_LE;
if (ioctl(fd, SNDCTL_DSP_SETFMT, &tmp) != 0)
{
Con_Printf("Couldn't set sample bits\n");
perror(snddev);
}
tmp = rate;
if (ioctl(fd, SNDCTL_DSP_SPEED, &tmp) != 0)
{
Con_Printf("Couldn't set capture rate\n");
perror(snddev);
}
fd++;
return (void*)fd;
}
void OSS_Capture_Start(void *ctx)
{
/*oss will automagically restart it when we next read*/
}
void OSS_Capture_Stop(void *ctx)
{
intptr_t fd = ((intptr_t)ctx)-1;
ioctl(fd, SNDCTL_DSP_RESET, NULL);
}
void OSS_Capture_Shutdown(void *ctx)
{
intptr_t fd = ((intptr_t)ctx)-1;
close(fd);
}
unsigned int OSS_Capture_Update(void *ctx, unsigned char *buffer, unsigned int minbytes, unsigned int maxbytes)
{
intptr_t fd = ((intptr_t)ctx)-1;
ssize_t res;
res = read(fd, buffer, maxbytes);
if (res < 0)
return 0;
return res;
}
snd_capture_driver_t OSS_Capture =
{
1,
"OSS",
OSS_Capture_Enumerate,
OSS_Capture_Init,
OSS_Capture_Start,
OSS_Capture_Update,
OSS_Capture_Stop,
OSS_Capture_Shutdown
};
#endif