Commit ca9cc28c authored by balrog's avatar balrog
Browse files

pthreads-based audio and miscellaneous audio clean-up (malc).

ESD support (malc, Frederick Reeve).


git-svn-id: svn://svn.savannah.nongnu.org/qemu/trunk@3917 c046a42c-6fe2-441c-8c8c-71466251a162
parent b34d259a
......@@ -73,6 +73,7 @@ AUDIO_OBJS += ossaudio.o
endif
ifdef CONFIG_COREAUDIO
AUDIO_OBJS += coreaudio.o
AUDIO_PT = yes
endif
ifdef CONFIG_ALSA
AUDIO_OBJS += alsaaudio.o
......@@ -84,6 +85,17 @@ ifdef CONFIG_FMOD
AUDIO_OBJS += fmodaudio.o
audio/audio.o audio/fmodaudio.o: CPPFLAGS := -I$(CONFIG_FMOD_INC) $(CPPFLAGS)
endif
ifdef CONFIG_ESD
AUDIO_PT = yes
AUDIO_PT_INT = yes
AUDIO_OBJS += esdaudio.o
endif
ifdef AUDIO_PT
LDFLAGS += -pthread
endif
ifdef AUDIO_PT_INT
AUDIO_OBJS += audio_pt_int.o
endif
AUDIO_OBJS+= wavcapture.o
OBJS+=$(addprefix audio/, $(AUDIO_OBJS))
......
......@@ -404,6 +404,9 @@ endif
ifdef CONFIG_ALSA
LIBS += -lasound
endif
ifdef CONFIG_ESD
LIBS += -lesd
endif
ifdef CONFIG_DSOUND
LIBS += -lole32 -ldxguid
endif
......@@ -412,6 +415,9 @@ LIBS += $(CONFIG_FMOD_LIB)
endif
SOUND_HW = sb16.o es1370.o
ifdef CONFIG_AC97
SOUND_HW += ac97.o
endif
ifdef CONFIG_ADLIB
SOUND_HW += fmopl.o adlib.o
endif
......@@ -641,8 +647,9 @@ endif
ifeq (1, 0)
audio.o sdlaudio.o dsoundaudio.o ossaudio.o wavaudio.o noaudio.o \
fmodaudio.o alsaaudio.o mixeng.o sb16.o es1370.o gus.o adlib.o: \
CFLAGS := $(CFLAGS) -Wall -Werror -W -Wsign-compare
fmodaudio.o alsaaudio.o mixeng.o sb16.o es1370.o ac97.o gus.o adlib.o \
esdaudio.o audio_pt_int.o: \
CFLAGS := $(CFLAGS) -O0 -g -Wall -Werror -W -Wsign-compare -Wno-unused
endif
# Include automatically generated dependency files
......
......@@ -86,9 +86,9 @@ static struct {
};
struct alsa_params_req {
unsigned int freq;
audfmt_e fmt;
unsigned int nchannels;
int freq;
snd_pcm_format_t fmt;
int nchannels;
unsigned int buffer_size;
unsigned int period_size;
};
......@@ -96,6 +96,7 @@ struct alsa_params_req {
struct alsa_params_obt {
int freq;
audfmt_e fmt;
int endianness;
int nchannels;
snd_pcm_uframes_t samples;
};
......@@ -143,7 +144,7 @@ static int alsa_write (SWVoiceOut *sw, void *buf, int len)
return audio_pcm_sw_write (sw, buf, len);
}
static int aud_to_alsafmt (audfmt_e fmt)
static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
{
switch (fmt) {
case AUD_FMT_S8:
......@@ -173,7 +174,8 @@ static int aud_to_alsafmt (audfmt_e fmt)
}
}
static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
int *endianness)
{
switch (alsafmt) {
case SND_PCM_FORMAT_S8:
......@@ -234,7 +236,6 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
return 0;
}
#if defined DEBUG_MISMATCHES || defined DEBUG
static void alsa_dump_info (struct alsa_params_req *req,
struct alsa_params_obt *obt)
{
......@@ -248,7 +249,6 @@ static void alsa_dump_info (struct alsa_params_req *req,
req->buffer_size, req->period_size);
dolog ("obtained: samples %ld\n", obt->samples);
}
#endif
static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
{
......@@ -291,6 +291,7 @@ static int alsa_open (int in, struct alsa_params_req *req,
unsigned int period_size, buffer_size;
snd_pcm_uframes_t obt_buffer_size;
const char *typ = in ? "ADC" : "DAC";
snd_pcm_format_t obtfmt;
freq = req->freq;
period_size = req->period_size;
......@@ -327,9 +328,8 @@ static int alsa_open (int in, struct alsa_params_req *req,
}
err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
if (err < 0) {
if (err < 0 && conf.verbose) {
alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
goto err;
}
err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
......@@ -494,6 +494,17 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
if (err < 0) {
alsa_logerr2 (err, typ, "Failed to get format\n");
goto err;
}
if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
dolog ("Invalid format was returned %d\n", obtfmt);
goto err;
}
err = snd_pcm_prepare (handle);
if (err < 0) {
alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
......@@ -504,28 +515,41 @@ static int alsa_open (int in, struct alsa_params_req *req,
snd_pcm_uframes_t threshold;
int bytes_per_sec;
bytes_per_sec = freq
<< (nchannels == 2)
<< (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
bytes_per_sec = freq << (nchannels == 2);
switch (obt->fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
bytes_per_sec <<= 1;
break;
case AUD_FMT_S32:
case AUD_FMT_U32:
bytes_per_sec <<= 2;
break;
}
threshold = (conf.threshold * bytes_per_sec) / 1000;
alsa_set_threshold (handle, threshold);
}
obt->fmt = req->fmt;
obt->nchannels = nchannels;
obt->freq = freq;
obt->samples = obt_buffer_size;
*handlep = handle;
#if defined DEBUG_MISMATCHES || defined DEBUG
if (obt->fmt != req->fmt ||
obt->nchannels != req->nchannels ||
obt->freq != req->freq) {
dolog ("Audio paramters mismatch for %s\n", typ);
if (conf.verbose &&
(obt->fmt != req->fmt ||
obt->nchannels != req->nchannels ||
obt->freq != req->freq)) {
dolog ("Audio paramters for %s\n", typ);
alsa_dump_info (req, obt);
}
#endif
#ifdef DEBUG
alsa_dump_info (req, obt);
......@@ -665,9 +689,6 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
audfmt_e effective_fmt;
int endianness;
int err;
snd_pcm_t *handle;
audsettings_t obt_as;
......@@ -681,16 +702,10 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
return -1;
}
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
alsa_anal_close (&handle);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianness;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
......@@ -751,9 +766,6 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
struct alsa_params_req req;
struct alsa_params_obt obt;
int endianness;
int err;
audfmt_e effective_fmt;
snd_pcm_t *handle;
audsettings_t obt_as;
......@@ -767,16 +779,10 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
return -1;
}
err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
alsa_anal_close (&handle);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianness;
obt_as.fmt = obt.fmt;
obt_as.endianness = obt.endianness;
audio_pcm_init_info (&hw->info, &obt_as);
hw->samples = obt.samples;
......
......@@ -55,6 +55,9 @@ static struct audio_driver *drvtab[] = {
#endif
#ifdef CONFIG_SDL
&sdl_audio_driver,
#endif
#ifdef CONFIG_ESD
&esd_audio_driver,
#endif
&no_audio_driver,
&wav_audio_driver
......@@ -414,7 +417,7 @@ static void audio_print_options (const char *prefix,
{
audfmt_e *fmtp = opt->valp;
printf (
"format, %s = %s, (one of: U8 S8 U16 S16)\n",
"format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n",
state,
audio_audfmt_to_string (*fmtp)
);
......
......@@ -202,6 +202,7 @@ extern struct audio_driver fmod_audio_driver;
extern struct audio_driver alsa_audio_driver;
extern struct audio_driver coreaudio_audio_driver;
extern struct audio_driver dsound_audio_driver;
extern struct audio_driver esd_audio_driver;
extern volume_t nominal_volume;
void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as);
......
#include "qemu-common.h"
#include "audio.h"
#define AUDIO_CAP "audio-pt"
#include "audio_int.h"
#include "audio_pt_int.h"
static void logerr (struct audio_pt *pt, int err, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (pt->drv, fmt, ap);
va_end (ap);
AUD_log (NULL, "\n");
AUD_log (pt->drv, "Reason: %s\n", strerror (err));
}
int audio_pt_init (struct audio_pt *p, void *(*func) (void *),
void *opaque, const char *drv, const char *cap)
{
int err, err2;
const char *efunc;
p->drv = drv;
err = pthread_mutex_init (&p->mutex, NULL);
if (err) {
efunc = "pthread_mutex_init";
goto err0;
}
err = pthread_cond_init (&p->cond, NULL);
if (err) {
efunc = "pthread_cond_init";
goto err1;
}
err = pthread_create (&p->thread, NULL, func, opaque);
if (err) {
efunc = "pthread_create";
goto err2;
}
return 0;
err2:
err2 = pthread_cond_destroy (&p->cond);
if (err2) {
logerr (p, err2, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC);
}
err1:
err2 = pthread_mutex_destroy (&p->mutex);
if (err2) {
logerr (p, err2, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC);
}
err0:
logerr (p, err, "%s(%s): %s failed", cap, AUDIO_FUNC, efunc);
return -1;
}
int audio_pt_fini (struct audio_pt *p, const char *cap)
{
int err, ret = 0;
err = pthread_cond_destroy (&p->cond);
if (err) {
logerr (p, err, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC);
ret = -1;
}
err = pthread_mutex_destroy (&p->mutex);
if (err) {
logerr (p, err, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC);
ret = -1;
}
return ret;
}
int audio_pt_lock (struct audio_pt *p, const char *cap)
{
int err;
err = pthread_mutex_lock (&p->mutex);
if (err) {
logerr (p, err, "%s(%s): pthread_mutex_lock failed", cap, AUDIO_FUNC);
return -1;
}
return 0;
}
int audio_pt_unlock (struct audio_pt *p, const char *cap)
{
int err;
err = pthread_mutex_unlock (&p->mutex);
if (err) {
logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC);
return -1;
}
return 0;
}
int audio_pt_wait (struct audio_pt *p, const char *cap)
{
int err;
err = pthread_cond_wait (&p->cond, &p->mutex);
if (err) {
logerr (p, err, "%s(%s): pthread_cond_wait failed", cap, AUDIO_FUNC);
return -1;
}
return 0;
}
int audio_pt_unlock_and_signal (struct audio_pt *p, const char *cap)
{
int err;
err = pthread_mutex_unlock (&p->mutex);
if (err) {
logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC);
return -1;
}
err = pthread_cond_signal (&p->cond);
if (err) {
logerr (p, err, "%s(%s): pthread_cond_signal failed", cap, AUDIO_FUNC);
return -1;
}
return 0;
}
int audio_pt_join (struct audio_pt *p, void **arg, const char *cap)
{
int err;
void *ret;
err = pthread_join (p->thread, &ret);
if (err) {
logerr (p, err, "%s(%s): pthread_join failed", cap, AUDIO_FUNC);
return -1;
}
*arg = ret;
return 0;
}
#ifndef QEMU_AUDIO_PT_INT_H
#define QEMU_AUDIO_PT_INT_H
#include <pthread.h>
struct audio_pt {
const char *drv;
pthread_t thread;
pthread_cond_t cond;
pthread_mutex_t mutex;
};
int audio_pt_init (struct audio_pt *, void *(*) (void *), void *,
const char *, const char *);
int audio_pt_fini (struct audio_pt *, const char *);
int audio_pt_lock (struct audio_pt *, const char *);
int audio_pt_unlock (struct audio_pt *, const char *);
int audio_pt_wait (struct audio_pt *, const char *);
int audio_pt_unlock_and_signal (struct audio_pt *, const char *);
int audio_pt_join (struct audio_pt *, void **, const char *);
#endif /* audio_pt_int.h */
......@@ -23,16 +23,20 @@
*/
#ifdef DSBTYPE_IN
#define NAME "capture buffer"
#define NAME2 "DirectSoundCapture"
#define TYPE in
#define IFACE IDirectSoundCaptureBuffer
#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
#define FIELD dsound_capture_buffer
#define FIELD2 dsound_capture
#else
#define NAME "playback buffer"
#define NAME2 "DirectSound"
#define TYPE out
#define IFACE IDirectSoundBuffer
#define BUFPTR LPDIRECTSOUNDBUFFER
#define FIELD dsound_buffer
#define FIELD2 dsound
#endif
static int glue (dsound_unlock_, TYPE) (
......@@ -192,6 +196,11 @@ static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as)
DSBCAPS bc;
#endif
if (!s->FIELD2) {
dsound_logerr ("Attempt to initialize voice without " NAME2 " object");
return -1;
}
err = waveformat_from_audio_settings (&wfx, as);
if (err) {
return -1;
......
......@@ -320,23 +320,22 @@ static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
switch (as->fmt) {
case AUD_FMT_S8:
wfx->wBitsPerSample = 8;
break;
case AUD_FMT_U8:
wfx->wBitsPerSample = 8;
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
break;
case AUD_FMT_U16:
wfx->wBitsPerSample = 16;
wfx->nAvgBytesPerSec <<= 1;
wfx->nBlockAlign <<= 1;
case AUD_FMT_S32:
case AUD_FMT_U32:
wfx->wBitsPerSample = 32;
wfx->nAvgBytesPerSec <<= 2;
wfx->nBlockAlign <<= 2;
break;
default:
......@@ -387,8 +386,13 @@ static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as)
as->fmt = AUD_FMT_S16;
break;
case 32:
as->fmt = AUD_FMT_S32;
break;
default:
dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n",
dolog ("Invalid wave format, bits per sample is not "
"8, 16 or 32, but %d\n",
wfx->wBitsPerSample);
return -1;
}
......
/*
* QEMU ESD audio driver
*
* Copyright (c) 2006 Frederick Reeve (brushed up by malc)
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <esd.h>
#include "qemu-common.h"
#include "audio.h"
#include <signal.h>
#define AUDIO_CAP "esd"
#include "audio_int.h"
#include "audio_pt_int.h"
typedef struct {
HWVoiceOut hw;
int done;
int live;
int decr;
int rpos;
void *pcm_buf;
int fd;
struct audio_pt pt;
} ESDVoiceOut;
typedef struct {
HWVoiceIn hw;
int done;
int dead;
int incr;
int wpos;
void *pcm_buf;
int fd;
struct audio_pt pt;
} ESDVoiceIn;
static struct {
int samples;
int divisor;
char *dac_host;
char *adc_host;
} conf = {
1024,
2,
NULL,
NULL
};
static void GCC_FMT_ATTR (2, 3) qesd_logerr (int err, const char *fmt, ...)
{
va_list ap;
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
}
/* playback */
static void *qesd_thread_out (void *arg)
{
ESDVoiceOut *esd = arg;
HWVoiceOut *hw = &esd->hw;
int threshold;
threshold = conf.divisor ? hw->samples / conf.divisor : 0;
for (;;) {
int decr, to_mix, rpos;
for (;;) {
if (esd->done) {
goto exit;
}
if (esd->live > threshold) {
break;
}
if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) {
goto exit;
}
}