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qemu/audio/ossaudio.c
Volker Rümelin 9833438ef6 audio: restore mixing-engine playback buffer size 
Commit ff095e5231 "audio: api for mixeng code free backends"
introduced another FIFO for the audio subsystem with exactly the
same size as the mixing-engine FIFO. Most audio backends use
this generic FIFO. The generic FIFO used together with the
mixing-engine FIFO doubles the audio FIFO size, because that's
just two independent FIFOs connected together in series.

For audio playback this nearly doubles the playback latency.

This patch restores the effective mixing-engine playback buffer
size to a pre v4.2.0 size by only accepting the amount of
samples for the mixing-engine queue which the downstream queue
accepts.

Signed-off-by: Volker Rümelin <vr_qemu@t-online.de>
Reviewed-by: Akihiko Odaki <akihiko.odaki@gmail.com>
Message-Id: <20220301191311.26695-10-vr_qemu@t-online.de>
Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
2022-03-04 11:05:13 +01:00

795 lines
20 KiB
C
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/*
* QEMU OSS audio driver
*
* Copyright (c) 2003-2005 Vassili Karpov (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 "qemu/osdep.h"
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "qemu/main-loop.h"
#include "qemu/module.h"
#include "qemu/host-utils.h"
#include "audio.h"
#include "trace.h"
#define AUDIO_CAP "oss"
#include "audio_int.h"
#if defined OSS_GETVERSION && defined SNDCTL_DSP_POLICY
#define USE_DSP_POLICY
#endif
typedef struct OSSVoiceOut {
HWVoiceOut hw;
int fd;
int nfrags;
int fragsize;
int mmapped;
Audiodev *dev;
} OSSVoiceOut;
typedef struct OSSVoiceIn {
HWVoiceIn hw;
int fd;
int nfrags;
int fragsize;
Audiodev *dev;
} OSSVoiceIn;
struct oss_params {
int freq;
int fmt;
int nchannels;
int nfrags;
int fragsize;
};
static void GCC_FMT_ATTR (2, 3) oss_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));
}
static void GCC_FMT_ATTR (3, 4) oss_logerr2 (
int err,
const char *typ,
const char *fmt,
...
)
{
va_list ap;
AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
va_start (ap, fmt);
AUD_vlog (AUDIO_CAP, fmt, ap);
va_end (ap);
AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
}
static void oss_anal_close (int *fdp)
{
int err;
qemu_set_fd_handler (*fdp, NULL, NULL, NULL);
err = close (*fdp);
if (err) {
oss_logerr (errno, "Failed to close file(fd=%d)\n", *fdp);
}
*fdp = -1;
}
static void oss_helper_poll_out (void *opaque)
{
AudioState *s = opaque;
audio_run(s, "oss_poll_out");
}
static void oss_helper_poll_in (void *opaque)
{
AudioState *s = opaque;
audio_run(s, "oss_poll_in");
}
static void oss_poll_out (HWVoiceOut *hw)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
qemu_set_fd_handler(oss->fd, NULL, oss_helper_poll_out, hw->s);
}
static void oss_poll_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
qemu_set_fd_handler(oss->fd, oss_helper_poll_in, NULL, hw->s);
}
static int aud_to_ossfmt (AudioFormat fmt, int endianness)
{
switch (fmt) {
case AUDIO_FORMAT_S8:
return AFMT_S8;
case AUDIO_FORMAT_U8:
return AFMT_U8;
case AUDIO_FORMAT_S16:
if (endianness) {
return AFMT_S16_BE;
} else {
return AFMT_S16_LE;
}
case AUDIO_FORMAT_U16:
if (endianness) {
return AFMT_U16_BE;
} else {
return AFMT_U16_LE;
}
default:
dolog ("Internal logic error: Bad audio format %d\n", fmt);
#ifdef DEBUG_AUDIO
abort ();
#endif
return AFMT_U8;
}
}
static int oss_to_audfmt (int ossfmt, AudioFormat *fmt, int *endianness)
{
switch (ossfmt) {
case AFMT_S8:
*endianness = 0;
*fmt = AUDIO_FORMAT_S8;
break;
case AFMT_U8:
*endianness = 0;
*fmt = AUDIO_FORMAT_U8;
break;
case AFMT_S16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_LE:
*endianness = 0;
*fmt = AUDIO_FORMAT_U16;
break;
case AFMT_S16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_S16;
break;
case AFMT_U16_BE:
*endianness = 1;
*fmt = AUDIO_FORMAT_U16;
break;
default:
dolog ("Unrecognized audio format %d\n", ossfmt);
return -1;
}
return 0;
}
#if defined DEBUG_MISMATCHES || defined DEBUG
static void oss_dump_info (struct oss_params *req, struct oss_params *obt)
{
dolog ("parameter | requested value | obtained value\n");
dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
dolog ("channels | %10d | %10d\n",
req->nchannels, obt->nchannels);
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
dolog ("nfrags | %10d | %10d\n", req->nfrags, obt->nfrags);
dolog ("fragsize | %10d | %10d\n",
req->fragsize, obt->fragsize);
}
#endif
#ifdef USE_DSP_POLICY
static int oss_get_version (int fd, int *version, const char *typ)
{
if (ioctl (fd, OSS_GETVERSION, &version)) {
#if defined(__FreeBSD__) || defined(__FreeBSD_kernel__)
/*
* Looks like atm (20100109) FreeBSD knows OSS_GETVERSION
* since 7.x, but currently only on the mixer device (or in
* the Linuxolator), and in the native version that part of
* the code is in fact never reached so the ioctl fails anyway.
* Until this is fixed, just check the errno and if its what
* FreeBSD's sound drivers return atm assume they are new enough.
*/
if (errno == EINVAL) {
*version = 0x040000;
return 0;
}
#endif
oss_logerr2 (errno, typ, "Failed to get OSS version\n");
return -1;
}
return 0;
}
#endif
static int oss_open(int in, struct oss_params *req, audsettings *as,
struct oss_params *obt, int *pfd, Audiodev *dev)
{
AudiodevOssOptions *oopts = &dev->u.oss;
AudiodevOssPerDirectionOptions *opdo = in ? oopts->in : oopts->out;
int fd;
int oflags = (oopts->has_exclusive && oopts->exclusive) ? O_EXCL : 0;
audio_buf_info abinfo;
int fmt, freq, nchannels;
int setfragment = 1;
const char *dspname = opdo->has_dev ? opdo->dev : "/dev/dsp";
const char *typ = in ? "ADC" : "DAC";
#ifdef USE_DSP_POLICY
int policy = oopts->has_dsp_policy ? oopts->dsp_policy : 5;
#endif
/* Kludge needed to have working mmap on Linux */
oflags |= (oopts->has_try_mmap && oopts->try_mmap) ?
O_RDWR : (in ? O_RDONLY : O_WRONLY);
fd = open (dspname, oflags | O_NONBLOCK);
if (-1 == fd) {
oss_logerr2 (errno, typ, "Failed to open `%s'\n", dspname);
return -1;
}
freq = req->freq;
nchannels = req->nchannels;
fmt = req->fmt;
req->nfrags = opdo->has_buffer_count ? opdo->buffer_count : 4;
req->fragsize = audio_buffer_bytes(
qapi_AudiodevOssPerDirectionOptions_base(opdo), as, 23220);
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
oss_logerr2 (errno, typ, "Failed to set number of channels %d\n",
req->nchannels);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
oss_logerr2 (errno, typ, "Failed to set frequency %d\n", req->freq);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_NONBLOCK, NULL)) {
oss_logerr2 (errno, typ, "Failed to set non-blocking mode\n");
goto err;
}
#ifdef USE_DSP_POLICY
if (policy >= 0) {
int version;
if (!oss_get_version (fd, &version, typ)) {
trace_oss_version(version);
if (version >= 0x040000) {
int policy2 = policy;
if (ioctl(fd, SNDCTL_DSP_POLICY, &policy2)) {
oss_logerr2 (errno, typ,
"Failed to set timing policy to %d\n",
policy);
goto err;
}
setfragment = 0;
}
}
}
#endif
if (setfragment) {
int mmmmssss = (req->nfrags << 16) | ctz32 (req->fragsize);
if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
oss_logerr2 (errno, typ, "Failed to set buffer length (%d, %d)\n",
req->nfrags, req->fragsize);
goto err;
}
}
if (ioctl (fd, in ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo)) {
oss_logerr2 (errno, typ, "Failed to get buffer length\n");
goto err;
}
if (!abinfo.fragstotal || !abinfo.fragsize) {
AUD_log (AUDIO_CAP, "Returned bogus buffer information(%d, %d) for %s\n",
abinfo.fragstotal, abinfo.fragsize, typ);
goto err;
}
obt->fmt = fmt;
obt->nchannels = nchannels;
obt->freq = freq;
obt->nfrags = abinfo.fragstotal;
obt->fragsize = abinfo.fragsize;
*pfd = fd;
#ifdef DEBUG_MISMATCHES
if ((req->fmt != obt->fmt) ||
(req->nchannels != obt->nchannels) ||
(req->freq != obt->freq) ||
(req->fragsize != obt->fragsize) ||
(req->nfrags != obt->nfrags)) {
dolog ("Audio parameters mismatch\n");
oss_dump_info (req, obt);
}
#endif
#ifdef DEBUG
oss_dump_info (req, obt);
#endif
return 0;
err:
oss_anal_close (&fd);
return -1;
}
static size_t oss_get_available_bytes(OSSVoiceOut *oss)
{
int err;
struct count_info cntinfo;
assert(oss->mmapped);
err = ioctl(oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
if (err < 0) {
oss_logerr(errno, "SNDCTL_DSP_GETOPTR failed\n");
return 0;
}
return audio_ring_dist(cntinfo.ptr, oss->hw.pos_emul, oss->hw.size_emul);
}
static void oss_run_buffer_out(HWVoiceOut *hw)
{
OSSVoiceOut *oss = (OSSVoiceOut *)hw;
if (!oss->mmapped) {
audio_generic_run_buffer_out(hw);
}
}
static size_t oss_buffer_get_free(HWVoiceOut *hw)
{
OSSVoiceOut *oss = (OSSVoiceOut *)hw;
if (oss->mmapped) {
return INT_MAX;
} else {
return audio_generic_buffer_get_free(hw);
}
}
static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
if (oss->mmapped) {
*size = MIN(oss_get_available_bytes(oss), hw->size_emul - hw->pos_emul);
return hw->buf_emul + hw->pos_emul;
} else {
return audio_generic_get_buffer_out(hw, size);
}
}
static size_t oss_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
if (oss->mmapped) {
assert(buf == hw->buf_emul + hw->pos_emul && size < hw->size_emul);
hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
return size;
} else {
return audio_generic_put_buffer_out(hw, buf, size);
}
}
static size_t oss_write(HWVoiceOut *hw, void *buf, size_t len)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
size_t pos;
if (oss->mmapped) {
size_t total_len;
len = MIN(len, oss_get_available_bytes(oss));
total_len = len;
while (len) {
size_t to_copy = MIN(len, hw->size_emul - hw->pos_emul);
memcpy(hw->buf_emul + hw->pos_emul, buf, to_copy);
hw->pos_emul = (hw->pos_emul + to_copy) % hw->size_emul;
buf += to_copy;
len -= to_copy;
}
return total_len;
}
pos = 0;
while (len) {
ssize_t bytes_written;
void *pcm = advance(buf, pos);
bytes_written = write(oss->fd, pcm, len);
if (bytes_written < 0) {
if (errno != EAGAIN) {
oss_logerr(errno, "failed to write %zu bytes\n",
len);
}
return pos;
}
pos += bytes_written;
if (bytes_written < len) {
break;
}
len -= bytes_written;
}
return pos;
}
static void oss_fini_out (HWVoiceOut *hw)
{
int err;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
ldebug ("oss_fini\n");
oss_anal_close (&oss->fd);
if (oss->mmapped && hw->buf_emul) {
err = munmap(hw->buf_emul, hw->size_emul);
if (err) {
oss_logerr(errno, "Failed to unmap buffer %p, size %zu\n",
hw->buf_emul, hw->size_emul);
}
hw->buf_emul = NULL;
}
}
static int oss_init_out(HWVoiceOut *hw, struct audsettings *as,
void *drv_opaque)
{
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
struct oss_params req, obt;
int endianness;
int err;
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
Audiodev *dev = drv_opaque;
AudiodevOssOptions *oopts = &dev->u.oss;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (oss_open(0, &req, as, &obt, &fd, dev)) {
return -1;
}
err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
oss_anal_close (&fd);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianness;
audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned DAC buffer, size %d, alignment %d\n",
obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->mmapped = 0;
if (oopts->has_try_mmap && oopts->try_mmap) {
hw->size_emul = hw->samples * hw->info.bytes_per_frame;
hw->buf_emul = mmap(
NULL,
hw->size_emul,
PROT_READ | PROT_WRITE,
MAP_SHARED,
fd,
0
);
if (hw->buf_emul == MAP_FAILED) {
oss_logerr(errno, "Failed to map %zu bytes of DAC\n",
hw->size_emul);
hw->buf_emul = NULL;
} else {
int err;
int trig = 0;
if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
} else {
trig = PCM_ENABLE_OUTPUT;
if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (
errno,
"SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
);
} else {
oss->mmapped = 1;
}
}
if (!oss->mmapped) {
err = munmap(hw->buf_emul, hw->size_emul);
if (err) {
oss_logerr(errno, "Failed to unmap buffer %p size %zu\n",
hw->buf_emul, hw->size_emul);
}
hw->buf_emul = NULL;
}
}
}
oss->fd = fd;
oss->dev = dev;
return 0;
}
static void oss_enable_out(HWVoiceOut *hw, bool enable)
{
int trig;
OSSVoiceOut *oss = (OSSVoiceOut *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
if (enable) {
hw->poll_mode = opdo->try_poll;
ldebug("enabling voice\n");
if (hw->poll_mode) {
oss_poll_out(hw);
}
if (!oss->mmapped) {
return;
}
audio_pcm_info_clear_buf(&hw->info, hw->buf_emul, hw->samples);
trig = PCM_ENABLE_OUTPUT;
if (ioctl(oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr(errno,
"SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n");
return;
}
} else {
if (hw->poll_mode) {
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
hw->poll_mode = 0;
}
if (!oss->mmapped) {
return;
}
ldebug ("disabling voice\n");
trig = 0;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
return;
}
}
}
static int oss_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
struct oss_params req, obt;
int endianness;
int err;
int fd;
AudioFormat effective_fmt;
struct audsettings obt_as;
Audiodev *dev = drv_opaque;
oss->fd = -1;
req.fmt = aud_to_ossfmt (as->fmt, as->endianness);
req.freq = as->freq;
req.nchannels = as->nchannels;
if (oss_open(1, &req, as, &obt, &fd, dev)) {
return -1;
}
err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
if (err) {
oss_anal_close (&fd);
return -1;
}
obt_as.freq = obt.freq;
obt_as.nchannels = obt.nchannels;
obt_as.fmt = effective_fmt;
obt_as.endianness = endianness;
audio_pcm_init_info (&hw->info, &obt_as);
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
if (obt.nfrags * obt.fragsize % hw->info.bytes_per_frame) {
dolog ("warning: Misaligned ADC buffer, size %d, alignment %d\n",
obt.nfrags * obt.fragsize, hw->info.bytes_per_frame);
}
hw->samples = (obt.nfrags * obt.fragsize) / hw->info.bytes_per_frame;
oss->fd = fd;
oss->dev = dev;
return 0;
}
static void oss_fini_in (HWVoiceIn *hw)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
oss_anal_close (&oss->fd);
}
static size_t oss_read(HWVoiceIn *hw, void *buf, size_t len)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
size_t pos = 0;
while (len) {
ssize_t nread;
void *dst = advance(buf, pos);
nread = read(oss->fd, dst, len);
if (nread == -1) {
switch (errno) {
case EINTR:
case EAGAIN:
break;
default:
oss_logerr(errno, "Failed to read %zu bytes of audio (to %p)\n",
len, dst);
break;
}
break;
}
pos += nread;
len -= nread;
}
return pos;
}
static void oss_enable_in(HWVoiceIn *hw, bool enable)
{
OSSVoiceIn *oss = (OSSVoiceIn *) hw;
AudiodevOssPerDirectionOptions *opdo = oss->dev->u.oss.out;
if (enable) {
hw->poll_mode = opdo->try_poll;
if (hw->poll_mode) {
oss_poll_in(hw);
}
} else {
if (hw->poll_mode) {
qemu_set_fd_handler (oss->fd, NULL, NULL, NULL);
hw->poll_mode = 0;
}
}
}
static void oss_init_per_direction(AudiodevOssPerDirectionOptions *opdo)
{
if (!opdo->has_try_poll) {
opdo->try_poll = true;
opdo->has_try_poll = true;
}
}
static void *oss_audio_init(Audiodev *dev)
{
AudiodevOssOptions *oopts;
assert(dev->driver == AUDIODEV_DRIVER_OSS);
oopts = &dev->u.oss;
oss_init_per_direction(oopts->in);
oss_init_per_direction(oopts->out);
if (access(oopts->in->has_dev ? oopts->in->dev : "/dev/dsp",
R_OK | W_OK) < 0 ||
access(oopts->out->has_dev ? oopts->out->dev : "/dev/dsp",
R_OK | W_OK) < 0) {
return NULL;
}
return dev;
}
static void oss_audio_fini (void *opaque)
{
}
static struct audio_pcm_ops oss_pcm_ops = {
.init_out = oss_init_out,
.fini_out = oss_fini_out,
.write = oss_write,
.buffer_get_free = oss_buffer_get_free,
.run_buffer_out = oss_run_buffer_out,
.get_buffer_out = oss_get_buffer_out,
.put_buffer_out = oss_put_buffer_out,
.enable_out = oss_enable_out,
.init_in = oss_init_in,
.fini_in = oss_fini_in,
.read = oss_read,
.run_buffer_in = audio_generic_run_buffer_in,
.enable_in = oss_enable_in
};
static struct audio_driver oss_audio_driver = {
.name = "oss",
.descr = "OSS http://www.opensound.com",
.init = oss_audio_init,
.fini = oss_audio_fini,
.pcm_ops = &oss_pcm_ops,
.can_be_default = 1,
.max_voices_out = INT_MAX,
.max_voices_in = INT_MAX,
.voice_size_out = sizeof (OSSVoiceOut),
.voice_size_in = sizeof (OSSVoiceIn)
};
static void register_audio_oss(void)
{
audio_driver_register(&oss_audio_driver);
}
type_init(register_audio_oss);
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