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SDL-mirror/src/audio/alsa/SDL_alsa_audio.c
Ryan C. Gordon da3564dedd ALSA backend can handle int32 and float32 data directly. 
--HG--
extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%402042
2006-08-31 21:28:21 +00:00

689 lines
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C
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2004 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Library General Public
License as published by the Free Software Foundation; either
version 2 of the License, or (at your option) any later version.
This library 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
Library General Public License for more details.
You should have received a copy of the GNU Library General Public
License along with this library; if not, write to the Free
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
Sam Lantinga
slouken@libsdl.org
*/
#include "SDL_config.h"
/* Allow access to a raw mixing buffer */
#include <sys/types.h>
#include <signal.h> /* For kill() */
#include "SDL_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.h"
#include "SDL_alsa_audio.h"
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
#include <dlfcn.h>
#include "SDL_name.h"
#include "SDL_loadso.h"
#else
#define SDL_NAME(X) X
#endif
/* The tag name used by ALSA audio */
#define DRIVER_NAME "alsa"
/* The default ALSA audio driver */
#define DEFAULT_DEVICE "default"
/* Audio driver functions */
static int ALSA_OpenAudio(_THIS, SDL_AudioSpec * spec);
static void ALSA_WaitAudio(_THIS);
static void ALSA_PlayAudio(_THIS);
static Uint8 *ALSA_GetAudioBuf(_THIS);
static void ALSA_CloseAudio(_THIS);
#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC
static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;
static void *alsa_handle = NULL;
static int alsa_loaded = 0;
static int (*SDL_snd_pcm_open) (snd_pcm_t ** pcm, const char *name,
snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_open)) (snd_pcm_t ** pcm, const char *name,
snd_pcm_stream_t stream, int mode);
static int (*SDL_NAME(snd_pcm_close)) (snd_pcm_t * pcm);
static snd_pcm_sframes_t(*SDL_NAME(snd_pcm_writei)) (snd_pcm_t * pcm,
const void *buffer,
snd_pcm_uframes_t size);
static int (*SDL_NAME(snd_pcm_resume)) (snd_pcm_t * pcm);
static int (*SDL_NAME(snd_pcm_prepare)) (snd_pcm_t * pcm);
static int (*SDL_NAME(snd_pcm_drain)) (snd_pcm_t * pcm);
static const char *(*SDL_NAME(snd_strerror)) (int errnum);
static size_t(*SDL_NAME(snd_pcm_hw_params_sizeof)) (void);
static size_t(*SDL_NAME(snd_pcm_sw_params_sizeof)) (void);
static int (*SDL_NAME(snd_pcm_hw_params_any)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t * params);
static int (*SDL_NAME(snd_pcm_hw_params_set_access)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t *
params,
snd_pcm_access_t
access);
static int (*SDL_NAME(snd_pcm_hw_params_set_format)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t *
params,
snd_pcm_format_t val);
static int (*SDL_NAME(snd_pcm_hw_params_set_channels)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t *
params,
unsigned int val);
static int (*SDL_NAME(snd_pcm_hw_params_get_channels)) (const
snd_pcm_hw_params_t *
params);
static unsigned int
(*SDL_NAME(snd_pcm_hw_params_set_rate_near)) (snd_pcm_t *
pcm,
snd_pcm_hw_params_t
* params,
unsigned int val, int *dir);
static snd_pcm_uframes_t
(*SDL_NAME(snd_pcm_hw_params_set_period_size_near)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t
* params,
snd_pcm_uframes_t
val, int *dir);
static snd_pcm_sframes_t
(*SDL_NAME(snd_pcm_hw_params_get_period_size)) (const
snd_pcm_hw_params_t
* params);
static unsigned int
(*SDL_NAME(snd_pcm_hw_params_set_periods_near)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t
* params,
unsigned int val,
int *dir);
static int (*SDL_NAME(snd_pcm_hw_params_get_periods)) (snd_pcm_hw_params_t *
params);
static int (*SDL_NAME(snd_pcm_hw_params)) (snd_pcm_t * pcm,
snd_pcm_hw_params_t * params);
/*
*/
static int (*SDL_NAME(snd_pcm_sw_params_current)) (snd_pcm_t * pcm,
snd_pcm_sw_params_t *
swparams);
static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold)) (snd_pcm_t *
pcm,
snd_pcm_sw_params_t
* params,
snd_pcm_uframes_t
val);
static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min)) (snd_pcm_t * pcm,
snd_pcm_sw_params_t
* params,
snd_pcm_uframes_t
val);
static int (*SDL_NAME(snd_pcm_sw_params)) (snd_pcm_t * pcm,
snd_pcm_sw_params_t * params);
static int (*SDL_NAME(snd_pcm_nonblock)) (snd_pcm_t * pcm, int nonblock);
#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)
#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof)
/* cast funcs to char* first, to please GCC's strict aliasing rules. */
static struct
{
const char *name;
void **func;
} alsa_functions[] = {
{
"snd_pcm_open", (void **) (char *) &SDL_NAME(snd_pcm_open)}, {
"snd_pcm_close", (void **) (char *) &SDL_NAME(snd_pcm_close)}, {
"snd_pcm_writei", (void **) (char *) &SDL_NAME(snd_pcm_writei)}, {
"snd_pcm_resume", (void **) (char *) &SDL_NAME(snd_pcm_resume)}, {
"snd_pcm_prepare", (void **) (char *) &SDL_NAME(snd_pcm_prepare)}, {
"snd_pcm_drain", (void **) (char *) &SDL_NAME(snd_pcm_drain)}, {
"snd_strerror", (void **) (char *) &SDL_NAME(snd_strerror)}, {
"snd_pcm_hw_params_sizeof",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_sizeof)}, {
"snd_pcm_sw_params_sizeof",
(void **) (char *) &SDL_NAME(snd_pcm_sw_params_sizeof)}, {
"snd_pcm_hw_params_any",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_any)}, {
"snd_pcm_hw_params_set_access",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_set_access)}, {
"snd_pcm_hw_params_set_format",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_set_format)}, {
"snd_pcm_hw_params_set_channels",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_set_channels)}, {
"snd_pcm_hw_params_get_channels",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_get_channels)}, {
"snd_pcm_hw_params_set_rate_near",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_set_rate_near)}, {
"snd_pcm_hw_params_set_period_size_near", (void **) (char *)
&SDL_NAME(snd_pcm_hw_params_set_period_size_near)}, {
"snd_pcm_hw_params_get_period_size",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_get_period_size)},
{
"snd_pcm_hw_params_set_periods_near", (void **) (char *)
&SDL_NAME(snd_pcm_hw_params_set_periods_near)}, {
"snd_pcm_hw_params_get_periods",
(void **) (char *) &SDL_NAME(snd_pcm_hw_params_get_periods)}, {
"snd_pcm_hw_params", (void **) (char *) &SDL_NAME(snd_pcm_hw_params)}, {
"snd_pcm_sw_params_current",
(void **) (char *) &SDL_NAME(snd_pcm_sw_params_current)}, {
"snd_pcm_sw_params_set_start_threshold", (void **) (char *)
&SDL_NAME(snd_pcm_sw_params_set_start_threshold)}, {
"snd_pcm_sw_params_set_avail_min",
(void **) (char *) &SDL_NAME(snd_pcm_sw_params_set_avail_min)}, {
"snd_pcm_sw_params", (void **) (char *) &SDL_NAME(snd_pcm_sw_params)}, {
"snd_pcm_nonblock", (void **) (char *) &SDL_NAME(snd_pcm_nonblock)},};
static void
UnloadALSALibrary(void)
{
if (alsa_loaded) {
/* SDL_UnloadObject(alsa_handle);*/
dlclose(alsa_handle);
alsa_handle = NULL;
alsa_loaded = 0;
}
}
static int
LoadALSALibrary(void)
{
int i, retval = -1;
/* alsa_handle = SDL_LoadObject(alsa_library);*/
alsa_handle = dlopen(alsa_library, RTLD_NOW);
if (alsa_handle) {
alsa_loaded = 1;
retval = 0;
for (i = 0; i < SDL_arraysize(alsa_functions); i++) {
/* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/
#if HAVE_DLVSYM
*alsa_functions[i].func =
dlvsym(alsa_handle, alsa_functions[i].name, "ALSA_0.9");
if (!*alsa_functions[i].func)
#endif
*alsa_functions[i].func =
dlsym(alsa_handle, alsa_functions[i].name);
if (!*alsa_functions[i].func) {
retval = -1;
UnloadALSALibrary();
break;
}
}
}
return retval;
}
#else
static void
UnloadALSALibrary(void)
{
return;
}
static int
LoadALSALibrary(void)
{
return 0;
}
#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */
static const char *
get_audio_device(int channels)
{
const char *device;
device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */
if (device == NULL) {
if (channels == 6)
device = "surround51";
else if (channels == 4)
device = "surround40";
else
device = DEFAULT_DEVICE;
}
return device;
}
/* Audio driver bootstrap functions */
static int
Audio_Available(void)
{
int available;
int status;
snd_pcm_t *handle;
available = 0;
if (LoadALSALibrary() < 0) {
return available;
}
status =
SDL_NAME(snd_pcm_open) (&handle, get_audio_device(2),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (status >= 0) {
available = 1;
SDL_NAME(snd_pcm_close) (handle);
}
UnloadALSALibrary();
return (available);
}
static void
Audio_DeleteDevice(SDL_AudioDevice * device)
{
SDL_free(device->hidden);
SDL_free(device);
UnloadALSALibrary();
}
static SDL_AudioDevice *
Audio_CreateDevice(int devindex)
{
SDL_AudioDevice *this;
/* Initialize all variables that we clean on shutdown */
LoadALSALibrary();
this = (SDL_AudioDevice *) SDL_malloc(sizeof(SDL_AudioDevice));
if (this) {
SDL_memset(this, 0, (sizeof *this));
this->hidden = (struct SDL_PrivateAudioData *)
SDL_malloc((sizeof *this->hidden));
}
if ((this == NULL) || (this->hidden == NULL)) {
SDL_OutOfMemory();
if (this) {
SDL_free(this);
}
return (0);
}
SDL_memset(this->hidden, 0, (sizeof *this->hidden));
/* Set the function pointers */
this->OpenAudio = ALSA_OpenAudio;
this->WaitAudio = ALSA_WaitAudio;
this->PlayAudio = ALSA_PlayAudio;
this->GetAudioBuf = ALSA_GetAudioBuf;
this->CloseAudio = ALSA_CloseAudio;
this->free = Audio_DeleteDevice;
return this;
}
AudioBootStrap ALSA_bootstrap = {
DRIVER_NAME, "ALSA 0.9 PCM audio",
Audio_Available, Audio_CreateDevice
};
/* This function waits until it is possible to write a full sound buffer */
static void
ALSA_WaitAudio(_THIS)
{
/* Check to see if the thread-parent process is still alive */
{
static int cnt = 0;
/* Note that this only works with thread implementations
that use a different process id for each thread.
*/
if (parent && (((++cnt) % 10) == 0)) { /* Check every 10 loops */
if (kill(parent, 0) < 0) {
this->enabled = 0;
}
}
}
}
/*
* http://bugzilla.libsdl.org/show_bug.cgi?id=110
* "For Linux ALSA, this is FL-FR-RL-RR-C-LFE
* and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"
*/
#define SWIZ6(T) \
T *ptr = (T *) mixbuf; \
const Uint32 count = (this->spec.samples / 6); \
Uint32 i; \
for (i = 0; i < count; i++, ptr += 6) { \
T tmp; \
tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \
tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \
}
static __inline__ void
swizzle_alsa_channels_6_64bit(_THIS)
{
SWIZ6(Uint64);
}
static __inline__ void
swizzle_alsa_channels_6_32bit(_THIS)
{
SWIZ6(Uint32);
}
static __inline__ void
swizzle_alsa_channels_6_16bit(_THIS)
{
SWIZ6(Uint16);
}
static __inline__ void
swizzle_alsa_channels_6_8bit(_THIS)
{
SWIZ6(Uint8);
}
#undef SWIZ6
/*
* Called right before feeding this->mixbuf to the hardware. Swizzle channels
* from Windows/Mac order to the format alsalib will want.
*/
static __inline__ void
swizzle_alsa_channels(_THIS)
{
if (this->spec.channels == 6) {
const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */
if (fmtsize == 16)
swizzle_alsa_channels_6_16bit(this);
else if (fmtsize == 8)
swizzle_alsa_channels_6_8bit(this);
else if (fmtsize == 32)
swizzle_alsa_channels_6_32bit(this);
else if (fmtsize == 64)
swizzle_alsa_channels_6_64bit(this);
}
/* !!! FIXME: update this for 7.1 if needed, later. */
}
static void
ALSA_PlayAudio(_THIS)
{
int status;
int sample_len;
signed short *sample_buf;
swizzle_alsa_channels(this);
sample_len = this->spec.samples;
sample_buf = (signed short *) mixbuf;
while (sample_len > 0) {
status =
SDL_NAME(snd_pcm_writei) (pcm_handle, sample_buf, sample_len);
if (status < 0) {
if (status == -EAGAIN) {
SDL_Delay(1);
continue;
}
if (status == -ESTRPIPE) {
do {
SDL_Delay(1);
status = SDL_NAME(snd_pcm_resume) (pcm_handle);
} while (status == -EAGAIN);
}
if (status < 0) {
status = SDL_NAME(snd_pcm_prepare) (pcm_handle);
}
if (status < 0) {
/* Hmm, not much we can do - abort */
this->enabled = 0;
return;
}
continue;
}
sample_buf += status * this->spec.channels;
sample_len -= status;
}
}
static Uint8 *
ALSA_GetAudioBuf(_THIS)
{
return (mixbuf);
}
static void
ALSA_CloseAudio(_THIS)
{
if (mixbuf != NULL) {
SDL_FreeAudioMem(mixbuf);
mixbuf = NULL;
}
if (pcm_handle) {
SDL_NAME(snd_pcm_drain) (pcm_handle);
SDL_NAME(snd_pcm_close) (pcm_handle);
pcm_handle = NULL;
}
}
static int
ALSA_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
int status;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
snd_pcm_format_t format;
snd_pcm_uframes_t frames;
SDL_AudioFormat test_format;
/* Open the audio device */
/* Name of device should depend on # channels in spec */
status =
SDL_NAME(snd_pcm_open) (&pcm_handle,
get_audio_device(spec->channels),
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (status < 0) {
SDL_SetError("Couldn't open audio device: %s",
SDL_NAME(snd_strerror) (status));
return (-1);
}
/* Figure out what the hardware is capable of */
snd_pcm_hw_params_alloca(&hwparams);
status = SDL_NAME(snd_pcm_hw_params_any) (pcm_handle, hwparams);
if (status < 0) {
SDL_SetError("Couldn't get hardware config: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
/* SDL only uses interleaved sample output */
status =
SDL_NAME(snd_pcm_hw_params_set_access) (pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (status < 0) {
SDL_SetError("Couldn't set interleaved access: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
/* Try for a closest match on audio format */
status = -1;
for (test_format = SDL_FirstAudioFormat(spec->format);
test_format && (status < 0);) {
switch (test_format) {
case AUDIO_U8:
format = SND_PCM_FORMAT_U8;
break;
case AUDIO_S8:
format = SND_PCM_FORMAT_S8;
break;
case AUDIO_S16LSB:
format = SND_PCM_FORMAT_S16_LE;
break;
case AUDIO_S16MSB:
format = SND_PCM_FORMAT_S16_BE;
break;
case AUDIO_U16LSB:
format = SND_PCM_FORMAT_U16_LE;
break;
case AUDIO_U16MSB:
format = SND_PCM_FORMAT_U16_BE;
break;
case AUDIO_S32LSB:
format = SND_PCM_FORMAT_U32_LE;
break;
case AUDIO_S32MSB:
format = SND_PCM_FORMAT_U32_BE;
break;
case AUDIO_F32LSB:
format = SND_PCM_FORMAT_FLOAT_LE;
break;
case AUDIO_F32MSB:
format = SND_PCM_FORMAT_FLOAT_BE;
break;
default:
format = 0;
break;
}
if (format != 0) {
status =
SDL_NAME(snd_pcm_hw_params_set_format) (pcm_handle,
hwparams, format);
}
if (status < 0) {
test_format = SDL_NextAudioFormat();
}
}
if (status < 0) {
SDL_SetError("Couldn't find any hardware audio formats");
ALSA_CloseAudio(this);
return (-1);
}
spec->format = test_format;
/* Set the number of channels */
status =
SDL_NAME(snd_pcm_hw_params_set_channels) (pcm_handle, hwparams,
spec->channels);
if (status < 0) {
status = SDL_NAME(snd_pcm_hw_params_get_channels) (hwparams);
if ((status <= 0) || (status > 2)) {
SDL_SetError("Couldn't set audio channels");
ALSA_CloseAudio(this);
return (-1);
}
spec->channels = status;
}
/* Set the audio rate */
status =
SDL_NAME(snd_pcm_hw_params_set_rate_near) (pcm_handle, hwparams,
spec->freq, NULL);
if (status < 0) {
SDL_SetError("Couldn't set audio frequency: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
spec->freq = status;
/* Set the buffer size, in samples */
frames = spec->samples;
frames =
SDL_NAME(snd_pcm_hw_params_set_period_size_near) (pcm_handle,
hwparams, frames,
NULL);
spec->samples = frames;
SDL_NAME(snd_pcm_hw_params_set_periods_near) (pcm_handle, hwparams, 2,
NULL);
/* "set" the hardware with the desired parameters */
status = SDL_NAME(snd_pcm_hw_params) (pcm_handle, hwparams);
if (status < 0) {
SDL_SetError("Couldn't set hardware audio parameters: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
/* This is useful for debugging... */
/*
{ snd_pcm_sframes_t bufsize; int fragments;
bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams);
fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams);
fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);
}
*/
/* Set the software parameters */
snd_pcm_sw_params_alloca(&swparams);
status = SDL_NAME(snd_pcm_sw_params_current) (pcm_handle, swparams);
if (status < 0) {
SDL_SetError("Couldn't get software config: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
status =
SDL_NAME(snd_pcm_sw_params_set_start_threshold) (pcm_handle,
swparams, 0);
if (status < 0) {
SDL_SetError("Couldn't set start threshold: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
status =
SDL_NAME(snd_pcm_sw_params_set_avail_min) (pcm_handle, swparams,
frames);
if (status < 0) {
SDL_SetError("Couldn't set avail min: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
status = SDL_NAME(snd_pcm_sw_params) (pcm_handle, swparams);
if (status < 0) {
SDL_SetError("Couldn't set software audio parameters: %s",
SDL_NAME(snd_strerror) (status));
ALSA_CloseAudio(this);
return (-1);
}
/* Calculate the final parameters for this audio specification */
SDL_CalculateAudioSpec(spec);
/* Allocate mixing buffer */
mixlen = spec->size;
mixbuf = (Uint8 *) SDL_AllocAudioMem(mixlen);
if (mixbuf == NULL) {
ALSA_CloseAudio(this);
return (-1);
}
SDL_memset(mixbuf, spec->silence, spec->size);
/* Get the parent process id (we're the parent of the audio thread) */
parent = getpid();
/* Switch to blocking mode for playback */
SDL_NAME(snd_pcm_nonblock) (pcm_handle, 0);
/* We're ready to rock and roll. :-) */
return (0);
}
/* vi: set ts=4 sw=4 expandtab: */
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