audio: new timer code for hda codec, fix audio_get_conf_int

-----BEGIN PGP SIGNATURE-----
 Version: GnuPG v2.0.22 (GNU/Linux)
 
 iQIcBAABAgAGBQJbMOpJAAoJEEy22O7T6HE46J4QAJFzQjX1trVObx7lqhVMiQiP
 C6IS6/FkqY22fylWQuvl2Sq/TfRpftXWvAIEYNybbYo2uT0oeZQDARP9Atq+WDjk
 yzlVHEsARmiXPnvyae3XsDAdnoEOfQne31kOE1qDbQxPHUHMigWSFw/6/zDqfV3P
 dObk1SJFrUfxVY6puECbtcli5SXsZG9DA7Vmy/LT0xsOAReOKxNNLN4/MipAwR3y
 iss21IGTSx157zF3nK5W7++BQVwI3+ATiTeOT7taCujzoKno0cKKQ/z9ItsFILGM
 eAyODn7X+lMSb2DTGmYi8mtzwBMxu5DsIvF07wevc/jD1zeeFayAj6+WT564MxDD
 t2sZ11q0h3rqm8MpPjje97BFjHpI296ZOLGBR1s6Z2z0V3YM76SsfmomEORfCaQ7
 3dJfFOGYoYpJQ/qvGF/uZSg5MwuqfpqpdMxwEFcl+zBLrIMcWAanTNNUtHs42rJV
 83klCGeZKe29X0U+CaDFMzXAAlUmRdjq9BFj4f0chzd8tx6hEQa8L2GJDKy32gHn
 xF6fwSInasP9p0rRI9bESd6J+1BOqhyvYS7G1zjx8feTIh3jVDFqxk9gjU7fFSmg
 RjPJgZxmYW6p6I762KundcsFzTxE7ZRTIeJ53XnDlQuQo8O1cj7ynTVV24QSkmA+
 1j+fLHHNNfQx4XhIR7cV
 =6DsA
 -----END PGP SIGNATURE-----

Merge remote-tracking branch 'remotes/kraxel/tags/audio-20180625-pull-request' into staging

audio: new timer code for hda codec, fix audio_get_conf_int

# gpg: Signature made Mon 25 Jun 2018 14:12:41 BST
# gpg:                using RSA key 4CB6D8EED3E87138
# gpg: Good signature from "Gerd Hoffmann (work) <kraxel@redhat.com>"
# gpg:                 aka "Gerd Hoffmann <gerd@kraxel.org>"
# gpg:                 aka "Gerd Hoffmann (private) <kraxel@gmail.com>"
# Primary key fingerprint: A032 8CFF B93A 17A7 9901  FE7D 4CB6 D8EE D3E8 7138

* remotes/kraxel/tags/audio-20180625-pull-request:
  audio: Convert use of atoi to qemu_strtoi
  audio/hda: enable new timer code by default.
  audio/hda: detect output buffer overruns
  audio/hda: tweak timer adjust logic
  audio/hda: turn some dprintfs into trace points
  audio/hda: create millisecond timers that handle IO

Signed-off-by: Peter Maydell <peter.maydell@linaro.org>
master
Peter Maydell 2018-06-25 15:25:26 +01:00
commit 35e238c933
5 changed files with 273 additions and 46 deletions

View File

@ -335,9 +335,8 @@ static int audio_get_conf_int (const char *key, int defval, int *defaultp)
char *strval;
strval = getenv (key);
if (strval) {
if (strval && !qemu_strtoi(strval, NULL, 10, &val)) {
*defaultp = 0;
val = atoi (strval);
return val;
}
else {

View File

@ -18,11 +18,13 @@
*/
#include "qemu/osdep.h"
#include "qemu/atomic.h"
#include "hw/hw.h"
#include "hw/pci/pci.h"
#include "intel-hda.h"
#include "intel-hda-defs.h"
#include "audio/audio.h"
#include "trace.h"
/* -------------------------------------------------------------------------- */
@ -126,6 +128,10 @@ static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
#define PARAM nomixemu
#include "hda-codec-common.h"
#define HDA_TIMER_TICKS (SCALE_MS)
#define B_SIZE sizeof(st->buf)
#define B_MASK (sizeof(st->buf) - 1)
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
@ -154,8 +160,13 @@ struct HDAAudioStream {
SWVoiceIn *in;
SWVoiceOut *out;
} voice;
uint8_t buf[HDA_BUFFER_SIZE];
uint32_t bpos;
uint8_t compat_buf[HDA_BUFFER_SIZE];
uint32_t compat_bpos;
uint8_t buf[8192]; /* size must be power of two */
int64_t rpos;
int64_t wpos;
QEMUTimer *buft;
int64_t buft_start;
};
#define TYPE_HDA_AUDIO "hda-audio"
@ -174,55 +185,217 @@ struct HDAAudioState {
/* properties */
uint32_t debug;
bool mixer;
bool use_timer;
};
static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
{
return 2 * st->as.nchannels * st->as.freq;
}
static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
{
int64_t limit = B_SIZE / 8;
int64_t corr = 0;
if (target_pos > limit) {
corr = HDA_TIMER_TICKS;
}
if (target_pos < -limit) {
corr = -HDA_TIMER_TICKS;
}
if (corr == 0) {
return;
}
trace_hda_audio_adjust(st->node->name, target_pos);
atomic_fetch_add(&st->buft_start, corr);
}
static void hda_audio_input_timer(void *opaque)
{
HDAAudioStream *st = opaque;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
int64_t buft_start = atomic_fetch_add(&st->buft_start, 0);
int64_t wpos = atomic_fetch_add(&st->wpos, 0);
int64_t rpos = atomic_fetch_add(&st->rpos, 0);
int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
/ NANOSECONDS_PER_SECOND;
wanted_rpos &= -4; /* IMPORTANT! clip to frames */
if (wanted_rpos <= rpos) {
/* we already transmitted the data */
goto out_timer;
}
int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
while (to_transfer) {
uint32_t start = (rpos & B_MASK);
uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
int rc = hda_codec_xfer(
&st->state->hda, st->stream, false, st->buf + start, chunk);
if (!rc) {
break;
}
rpos += chunk;
to_transfer -= chunk;
atomic_fetch_add(&st->rpos, chunk);
}
out_timer:
if (st->running) {
timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
}
}
static void hda_audio_input_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
int recv = 0;
int len;
bool rc;
while (avail - recv >= sizeof(st->buf)) {
if (st->bpos != sizeof(st->buf)) {
len = AUD_read(st->voice.in, st->buf + st->bpos,
sizeof(st->buf) - st->bpos);
st->bpos += len;
recv += len;
if (st->bpos != sizeof(st->buf)) {
break;
}
int64_t wpos = atomic_fetch_add(&st->wpos, 0);
int64_t rpos = atomic_fetch_add(&st->rpos, 0);
int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
while (to_transfer) {
uint32_t start = (uint32_t) (wpos & B_MASK);
uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
wpos += read;
to_transfer -= read;
atomic_fetch_add(&st->wpos, read);
if (chunk != read) {
break;
}
rc = hda_codec_xfer(&st->state->hda, st->stream, false,
st->buf, sizeof(st->buf));
}
}
static void hda_audio_output_timer(void *opaque)
{
HDAAudioStream *st = opaque;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
int64_t buft_start = atomic_fetch_add(&st->buft_start, 0);
int64_t wpos = atomic_fetch_add(&st->wpos, 0);
int64_t rpos = atomic_fetch_add(&st->rpos, 0);
int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
/ NANOSECONDS_PER_SECOND;
wanted_wpos &= -4; /* IMPORTANT! clip to frames */
if (wanted_wpos <= wpos) {
/* we already received the data */
goto out_timer;
}
int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
while (to_transfer) {
uint32_t start = (wpos & B_MASK);
uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
int rc = hda_codec_xfer(
&st->state->hda, st->stream, true, st->buf + start, chunk);
if (!rc) {
break;
}
st->bpos = 0;
wpos += chunk;
to_transfer -= chunk;
atomic_fetch_add(&st->wpos, chunk);
}
out_timer:
if (st->running) {
timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
}
}
static void hda_audio_output_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
int64_t wpos = atomic_fetch_add(&st->wpos, 0);
int64_t rpos = atomic_fetch_add(&st->rpos, 0);
int64_t to_transfer = audio_MIN(wpos - rpos, avail);
if (wpos - rpos == B_SIZE) {
/* drop buffer, reset timer adjust */
st->rpos = 0;
st->wpos = 0;
st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
trace_hda_audio_overrun(st->node->name);
return;
}
hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1));
while (to_transfer) {
uint32_t start = (uint32_t) (rpos & B_MASK);
uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
rpos += written;
to_transfer -= written;
atomic_fetch_add(&st->rpos, written);
if (chunk != written) {
break;
}
}
}
static void hda_audio_compat_input_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
int recv = 0;
int len;
bool rc;
while (avail - recv >= sizeof(st->compat_buf)) {
if (st->compat_bpos != sizeof(st->compat_buf)) {
len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
sizeof(st->compat_buf) - st->compat_bpos);
st->compat_bpos += len;
recv += len;
if (st->compat_bpos != sizeof(st->compat_buf)) {
break;
}
}
rc = hda_codec_xfer(&st->state->hda, st->stream, false,
st->compat_buf, sizeof(st->compat_buf));
if (!rc) {
break;
}
st->compat_bpos = 0;
}
}
static void hda_audio_compat_output_cb(void *opaque, int avail)
{
HDAAudioStream *st = opaque;
int sent = 0;
int len;
bool rc;
while (avail - sent >= sizeof(st->buf)) {
if (st->bpos == sizeof(st->buf)) {
while (avail - sent >= sizeof(st->compat_buf)) {
if (st->compat_bpos == sizeof(st->compat_buf)) {
rc = hda_codec_xfer(&st->state->hda, st->stream, true,
st->buf, sizeof(st->buf));
st->compat_buf, sizeof(st->compat_buf));
if (!rc) {
break;
}
st->bpos = 0;
st->compat_bpos = 0;
}
len = AUD_write(st->voice.out, st->buf + st->bpos,
sizeof(st->buf) - st->bpos);
st->bpos += len;
len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
sizeof(st->compat_buf) - st->compat_bpos);
st->compat_bpos += len;
sent += len;
if (st->bpos != sizeof(st->buf)) {
if (st->compat_bpos != sizeof(st->compat_buf)) {
break;
}
}
@ -237,8 +410,18 @@ static void hda_audio_set_running(HDAAudioStream *st, bool running)
return;
}
st->running = running;
dprint(st->state, 1, "%s: %s (stream %d)\n", st->node->name,
st->running ? "on" : "off", st->stream);
trace_hda_audio_running(st->node->name, st->stream, st->running);
if (st->state->use_timer) {
if (running) {
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
st->rpos = 0;
st->wpos = 0;
st->buft_start = now;
timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
} else {
timer_del(st->buft);
}
}
if (st->output) {
AUD_set_active_out(st->voice.out, st->running);
} else {
@ -274,22 +457,36 @@ static void hda_audio_set_amp(HDAAudioStream *st)
static void hda_audio_setup(HDAAudioStream *st)
{
bool use_timer = st->state->use_timer;
audio_callback_fn cb;
if (st->node == NULL) {
return;
}
dprint(st->state, 1, "%s: format: %d x %s @ %d Hz\n",
st->node->name, st->as.nchannels,
fmt2name[st->as.fmt], st->as.freq);
trace_hda_audio_format(st->node->name, st->as.nchannels,
fmt2name[st->as.fmt], st->as.freq);
if (st->output) {
if (use_timer) {
cb = hda_audio_output_cb;
st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
hda_audio_output_timer, st);
} else {
cb = hda_audio_compat_output_cb;
}
st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
st->node->name, st,
hda_audio_output_cb, &st->as);
st->node->name, st, cb, &st->as);
} else {
if (use_timer) {
cb = hda_audio_input_cb;
st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
hda_audio_input_timer, st);
} else {
cb = hda_audio_compat_input_cb;
}
st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
st->node->name, st,
hda_audio_input_cb, &st->as);
st->node->name, st, cb, &st->as);
}
}
@ -505,7 +702,7 @@ static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
/* unmute output by default */
st->gain_left = QEMU_HDA_AMP_STEPS;
st->gain_right = QEMU_HDA_AMP_STEPS;
st->bpos = sizeof(st->buf);
st->compat_bpos = sizeof(st->compat_buf);
st->output = true;
} else {
st->output = false;
@ -532,6 +729,9 @@ static void hda_audio_exit(HDACodecDevice *hda)
if (st->node == NULL) {
continue;
}
if (a->use_timer) {
timer_del(st->buft);
}
if (st->output) {
AUD_close_out(&a->card, st->voice.out);
} else {
@ -581,6 +781,26 @@ static void hda_audio_reset(DeviceState *dev)
}
}
static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
{
HDAAudioStream *st = opaque;
return st->state->use_timer;
}
static const VMStateDescription vmstate_hda_audio_stream_buf = {
.name = "hda-audio-stream/buffer",
.version_id = 1,
.needed = vmstate_hda_audio_stream_buf_needed,
.fields = (VMStateField[]) {
VMSTATE_BUFFER(buf, HDAAudioStream),
VMSTATE_INT64(rpos, HDAAudioStream),
VMSTATE_INT64(wpos, HDAAudioStream),
VMSTATE_TIMER_PTR(buft, HDAAudioStream),
VMSTATE_INT64(buft_start, HDAAudioStream),
VMSTATE_END_OF_LIST()
}
};
static const VMStateDescription vmstate_hda_audio_stream = {
.name = "hda-audio-stream",
.version_id = 1,
@ -592,9 +812,13 @@ static const VMStateDescription vmstate_hda_audio_stream = {
VMSTATE_UINT32(gain_right, HDAAudioStream),
VMSTATE_BOOL(mute_left, HDAAudioStream),
VMSTATE_BOOL(mute_right, HDAAudioStream),
VMSTATE_UINT32(bpos, HDAAudioStream),
VMSTATE_BUFFER(buf, HDAAudioStream),
VMSTATE_UINT32(compat_bpos, HDAAudioStream),
VMSTATE_BUFFER(compat_buf, HDAAudioStream),
VMSTATE_END_OF_LIST()
},
.subsections = (const VMStateDescription * []) {
&vmstate_hda_audio_stream_buf,
NULL
}
};
@ -615,6 +839,7 @@ static const VMStateDescription vmstate_hda_audio = {
static Property hda_audio_properties[] = {
DEFINE_PROP_UINT32("debug", HDAAudioState, debug, 0),
DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true),
DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer, true),
DEFINE_PROP_END_OF_LIST(),
};

View File

@ -407,13 +407,6 @@ static bool intel_hda_xfer(HDACodecDevice *dev, uint32_t stnr, bool output,
if (st->bpl == NULL) {
return false;
}
if (st->ctl & (1 << 26)) {
/*
* Wait with the next DMA xfer until the guest
* has acked the buffer completion interrupt
*/
return false;
}
left = len;
s = st->bentries;

View File

@ -17,3 +17,9 @@ milkymist_ac97_in_cb(int avail, uint32_t remaining) "avail %d remaining %u"
milkymist_ac97_in_cb_transferred(int transferred) "transferred %d"
milkymist_ac97_out_cb(int free, uint32_t remaining) "free %d remaining %u"
milkymist_ac97_out_cb_transferred(int transferred) "transferred %d"
# hw/audio/hda-codec.c
hda_audio_running(const char *stream, int nr, bool running) "st %s, nr %d, run %d"
hda_audio_format(const char *stream, int chan, const char *fmt, int freq) "st %s, %d x %s @ %d Hz"
hda_audio_adjust(const char *stream, int pos) "st %s, pos %d"
hda_audio_overrun(const char *stream) "st %s"

View File

@ -6,6 +6,10 @@
.driver = "migration",\
.property = "decompress-error-check",\
.value = "off",\
},{\
.driver = "hda-audio",\
.property = "use-timer",\
.value = "false",\
},
#define HW_COMPAT_2_11 \