mirrors/qemu
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
master
qemu/hw/audio/hda-codec.c
Marc-André Lureau 6d6e23361f hw/audio/hda: fix memory leak on audio setup 
When SET_STREAM_FORMAT is called, we should clear the existing setup.

Factor out common function to close a stream.

Direct leak of 144 byte(s) in 3 object(s) allocated from:
    #0 0x7f91d38f7350 in calloc (/lib64/libasan.so.8+0xf7350) (BuildId: a4ad7eb954b390cf00f07fa10952988a41d9fc7a)
    #1 0x7f91d2ab7871 in g_malloc0 (/lib64/libglib-2.0.so.0+0x64871) (BuildId: 36b60dbd02e796145a982d0151ce37202ec05649)
    #2 0x562fa2f447ee in timer_new_full /home/elmarco/src/qemu/include/qemu/timer.h:538
    #3 0x562fa2f4486f in timer_new /home/elmarco/src/qemu/include/qemu/timer.h:559
    #4 0x562fa2f448a9 in timer_new_ns /home/elmarco/src/qemu/include/qemu/timer.h:577
    #5 0x562fa2f47955 in hda_audio_setup ../hw/audio/hda-codec.c:490
    #6 0x562fa2f4897e in hda_audio_command ../hw/audio/hda-codec.c:605

Signed-off-by: Marc-André Lureau <marcandre.lureau@redhat.com>
Reviewed-by: Akihiko Odaki <akihiko.odaki@daynix.com>
Message-ID: <20241008125028.1177932-3-marcandre.lureau@redhat.com>
2024-10-14 17:34:09 +04:00

987 lines
27 KiB
C
Raw Permalink Blame History

/*
* Copyright (C) 2010 Red Hat, Inc.
*
* written by Gerd Hoffmann <kraxel@redhat.com>
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License as
* published by the Free Software Foundation; either version 2 or
* (at your option) version 3 of the License.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, see <http://www.gnu.org/licenses/>.
*/
#include "qemu/osdep.h"
#include "hw/pci/pci.h"
#include "hw/qdev-properties.h"
#include "intel-hda.h"
#include "migration/vmstate.h"
#include "qemu/host-utils.h"
#include "qemu/module.h"
#include "intel-hda-defs.h"
#include "audio/audio.h"
#include "trace.h"
#include "qom/object.h"
/* -------------------------------------------------------------------------- */
typedef struct desc_param {
uint32_t id;
uint32_t val;
} desc_param;
typedef struct desc_node {
uint32_t nid;
const char *name;
const desc_param *params;
uint32_t nparams;
uint32_t config;
uint32_t pinctl;
uint32_t *conn;
uint32_t stindex;
} desc_node;
typedef struct desc_codec {
const char *name;
uint32_t iid;
const desc_node *nodes;
uint32_t nnodes;
} desc_codec;
static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
{
int i;
for (i = 0; i < node->nparams; i++) {
if (node->params[i].id == id) {
return &node->params[i];
}
}
return NULL;
}
static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
{
int i;
for (i = 0; i < codec->nnodes; i++) {
if (codec->nodes[i].nid == nid) {
return &codec->nodes[i];
}
}
return NULL;
}
static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
{
if (format & AC_FMT_TYPE_NON_PCM) {
return;
}
as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;
switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
case 1: as->freq *= 2; break;
case 2: as->freq *= 3; break;
case 3: as->freq *= 4; break;
}
switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
case 1: as->freq /= 2; break;
case 2: as->freq /= 3; break;
case 3: as->freq /= 4; break;
case 4: as->freq /= 5; break;
case 5: as->freq /= 6; break;
case 6: as->freq /= 7; break;
case 7: as->freq /= 8; break;
}
switch (format & AC_FMT_BITS_MASK) {
case AC_FMT_BITS_8: as->fmt = AUDIO_FORMAT_S8; break;
case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
}
as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
}
/* -------------------------------------------------------------------------- */
/*
* HDA codec descriptions
*/
/* some defines */
#define QEMU_HDA_ID_VENDOR 0x1af4
#define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 | \
0x1fc /* 16 -> 96 kHz */)
#define QEMU_HDA_AMP_NONE (0)
#define QEMU_HDA_AMP_STEPS 0x4a
#define PARAM mixemu
#define HDA_MIXER
#include "hda-codec-common.h"
#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[] = {
[ AUDIO_FORMAT_U8 ] = "PCM-U8",
[ AUDIO_FORMAT_S8 ] = "PCM-S8",
[ AUDIO_FORMAT_U16 ] = "PCM-U16",
[ AUDIO_FORMAT_S16 ] = "PCM-S16",
[ AUDIO_FORMAT_U32 ] = "PCM-U32",
[ AUDIO_FORMAT_S32 ] = "PCM-S32",
};
#define TYPE_HDA_AUDIO "hda-audio"
OBJECT_DECLARE_SIMPLE_TYPE(HDAAudioState, HDA_AUDIO)
typedef struct HDAAudioStream HDAAudioStream;
struct HDAAudioStream {
HDAAudioState *state;
const desc_node *node;
bool output, running;
uint32_t stream;
uint32_t channel;
uint32_t format;
uint32_t gain_left, gain_right;
bool mute_left, mute_right;
struct audsettings as;
union {
SWVoiceIn *in;
SWVoiceOut *out;
} voice;
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;
};
struct HDAAudioState {
HDACodecDevice hda;
const char *name;
QEMUSoundCard card;
const desc_codec *desc;
HDAAudioStream st[4];
bool running_compat[16];
bool running_real[2 * 16];
/* properties */
uint32_t debug;
bool mixer;
bool use_timer;
};
static inline uint32_t hda_bytes_per_second(HDAAudioStream *st)
{
return 2 * (uint32_t)st->as.nchannels * (uint32_t)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 (target_pos < -(2 * limit)) {
corr = -(4 * HDA_TIMER_TICKS);
}
if (corr == 0) {
return;
}
trace_hda_audio_adjust(st->node->name, target_pos);
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 uptime = now - st->buft_start;
int64_t wpos = st->wpos;
int64_t rpos = st->rpos;
int64_t wanted_rpos;
if (uptime <= 0) {
/* wanted_rpos <= 0 */
goto out_timer;
}
wanted_rpos = muldiv64(uptime, hda_bytes_per_second(st),
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 = MIN(wpos - rpos, wanted_rpos - rpos);
while (to_transfer) {
uint32_t start = (rpos & B_MASK);
uint32_t chunk = 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;
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;
int64_t wpos = st->wpos;
int64_t rpos = st->rpos;
int64_t to_transfer = MIN(B_SIZE - (wpos - rpos), avail);
while (to_transfer) {
uint32_t start = (uint32_t) (wpos & B_MASK);
uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
wpos += read;
to_transfer -= read;
st->wpos += read;
if (chunk != read) {
break;
}
}
hda_timer_sync_adjust(st, -((wpos - rpos) - (B_SIZE >> 1)));
}
static void hda_audio_output_timer(void *opaque)
{
HDAAudioStream *st = opaque;
int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
int64_t uptime = now - st->buft_start;
int64_t wpos = st->wpos;
int64_t rpos = st->rpos;
int64_t wanted_wpos;
if (uptime <= 0) {
/* wanted_wpos <= 0 */
goto out_timer;
}
wanted_wpos = muldiv64(uptime, hda_bytes_per_second(st),
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 = MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
while (to_transfer) {
uint32_t start = (wpos & B_MASK);
uint32_t chunk = MIN(B_SIZE - start, to_transfer);
int rc = hda_codec_xfer(
&st->state->hda, st->stream, true, st->buf + start, chunk);
if (!rc) {
break;
}
wpos += chunk;
to_transfer -= chunk;
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 = st->wpos;
int64_t rpos = st->rpos;
int64_t to_transfer = 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;
}
while (to_transfer) {
uint32_t start = (uint32_t) (rpos & B_MASK);
uint32_t chunk = (uint32_t) MIN(B_SIZE - start, to_transfer);
uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
rpos += written;
to_transfer -= written;
st->rpos += written;
if (chunk != written) {
break;
}
}
hda_timer_sync_adjust(st, (wpos - rpos) - (B_SIZE >> 1));
}
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->compat_buf)) {
if (st->compat_bpos == sizeof(st->compat_buf)) {
rc = hda_codec_xfer(&st->state->hda, st->stream, true,
st->compat_buf, sizeof(st->compat_buf));
if (!rc) {
break;
}
st->compat_bpos = 0;
}
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->compat_bpos != sizeof(st->compat_buf)) {
break;
}
}
}
static void hda_audio_set_running(HDAAudioStream *st, bool running)
{
if (st->node == NULL) {
return;
}
if (st->running == running) {
return;
}
st->running = running;
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 {
AUD_set_active_in(st->voice.in, st->running);
}
}
static void hda_audio_set_amp(HDAAudioStream *st)
{
bool muted;
uint32_t left, right;
if (st->node == NULL) {
return;
}
muted = st->mute_left && st->mute_right;
left = st->mute_left ? 0 : st->gain_left;
right = st->mute_right ? 0 : st->gain_right;
left = left * 255 / QEMU_HDA_AMP_STEPS;
right = right * 255 / QEMU_HDA_AMP_STEPS;
if (!st->state->mixer) {
return;
}
if (st->output) {
AUD_set_volume_out(st->voice.out, muted, left, right);
} else {
AUD_set_volume_in(st->voice.in, muted, left, right);
}
}
static void hda_close_stream(HDAAudioState *a, HDAAudioStream *st)
{
if (st->node == NULL) {
return;
}
if (a->use_timer) {
timer_free(st->buft);
st->buft = NULL;
}
if (st->output) {
AUD_close_out(&a->card, st->voice.out);
st->voice.out = NULL;
} else {
AUD_close_in(&a->card, st->voice.in);
st->voice.in = NULL;
}
}
static void hda_audio_setup(HDAAudioStream *st)
{
bool use_timer = st->state->use_timer;
audio_callback_fn cb;
if (st->node == NULL) {
return;
}
trace_hda_audio_format(st->node->name, st->as.nchannels,
fmt2name[st->as.fmt], st->as.freq);
hda_close_stream(st->state, st);
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, 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, cb, &st->as);
}
}
static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
{
HDAAudioState *a = HDA_AUDIO(hda);
HDAAudioStream *st;
const desc_node *node = NULL;
const desc_param *param;
uint32_t verb, payload, response, count, shift;
if ((data & 0x70000) == 0x70000) {
/* 12/8 id/payload */
verb = (data >> 8) & 0xfff;
payload = data & 0x00ff;
} else {
/* 4/16 id/payload */
verb = (data >> 8) & 0xf00;
payload = data & 0xffff;
}
node = hda_codec_find_node(a->desc, nid);
if (node == NULL) {
goto fail;
}
dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
__func__, nid, node->name, verb, payload);
switch (verb) {
/* all nodes */
case AC_VERB_PARAMETERS:
param = hda_codec_find_param(node, payload);
if (param == NULL) {
goto fail;
}
hda_codec_response(hda, true, param->val);
break;
case AC_VERB_GET_SUBSYSTEM_ID:
hda_codec_response(hda, true, a->desc->iid);
break;
/* all functions */
case AC_VERB_GET_CONNECT_LIST:
param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
count = param ? param->val : 0;
response = 0;
shift = 0;
while (payload < count && shift < 32) {
response |= node->conn[payload] << shift;
payload++;
shift += 8;
}
hda_codec_response(hda, true, response);
break;
/* pin widget */
case AC_VERB_GET_CONFIG_DEFAULT:
hda_codec_response(hda, true, node->config);
break;
case AC_VERB_GET_PIN_WIDGET_CONTROL:
hda_codec_response(hda, true, node->pinctl);
break;
case AC_VERB_SET_PIN_WIDGET_CONTROL:
if (node->pinctl != payload) {
dprint(a, 1, "unhandled pin control bit\n");
}
hda_codec_response(hda, true, 0);
break;
/* audio in/out widget */
case AC_VERB_SET_CHANNEL_STREAMID:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
hda_audio_set_running(st, false);
st->stream = (payload >> 4) & 0x0f;
st->channel = payload & 0x0f;
dprint(a, 2, "%s: stream %d, channel %d\n",
st->node->name, st->stream, st->channel);
hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
hda_codec_response(hda, true, 0);
break;
case AC_VERB_GET_CONV:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
response = st->stream << 4 | st->channel;
hda_codec_response(hda, true, response);
break;
case AC_VERB_SET_STREAM_FORMAT:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
st->format = payload;
hda_codec_parse_fmt(st->format, &st->as);
hda_audio_setup(st);
hda_codec_response(hda, true, 0);
break;
case AC_VERB_GET_STREAM_FORMAT:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
hda_codec_response(hda, true, st->format);
break;
case AC_VERB_GET_AMP_GAIN_MUTE:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
if (payload & AC_AMP_GET_LEFT) {
response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
} else {
response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
}
hda_codec_response(hda, true, response);
break;
case AC_VERB_SET_AMP_GAIN_MUTE:
st = a->st + node->stindex;
if (st->node == NULL) {
goto fail;
}
dprint(a, 1, "amp (%s): %s%s%s%s index %d gain %3d %s\n",
st->node->name,
(payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
(payload & AC_AMP_SET_INPUT) ? "i" : "-",
(payload & AC_AMP_SET_LEFT) ? "l" : "-",
(payload & AC_AMP_SET_RIGHT) ? "r" : "-",
(payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
(payload & AC_AMP_GAIN),
(payload & AC_AMP_MUTE) ? "muted" : "");
if (payload & AC_AMP_SET_LEFT) {
st->gain_left = payload & AC_AMP_GAIN;
st->mute_left = payload & AC_AMP_MUTE;
}
if (payload & AC_AMP_SET_RIGHT) {
st->gain_right = payload & AC_AMP_GAIN;
st->mute_right = payload & AC_AMP_MUTE;
}
hda_audio_set_amp(st);
hda_codec_response(hda, true, 0);
break;
/* not supported */
case AC_VERB_SET_POWER_STATE:
case AC_VERB_GET_POWER_STATE:
case AC_VERB_GET_SDI_SELECT:
hda_codec_response(hda, true, 0);
break;
default:
goto fail;
}
return;
fail:
dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
__func__, nid, node ? node->name : "?", verb, payload);
hda_codec_response(hda, true, 0);
}
static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
{
HDAAudioState *a = HDA_AUDIO(hda);
int s;
a->running_compat[stnr] = running;
a->running_real[output * 16 + stnr] = running;
for (s = 0; s < ARRAY_SIZE(a->st); s++) {
if (a->st[s].node == NULL) {
continue;
}
if (a->st[s].output != output) {
continue;
}
if (a->st[s].stream != stnr) {
continue;
}
hda_audio_set_running(&a->st[s], running);
}
}
static void hda_audio_init(HDACodecDevice *hda,
const struct desc_codec *desc,
Error **errp)
{
HDAAudioState *a = HDA_AUDIO(hda);
HDAAudioStream *st;
const desc_node *node;
const desc_param *param;
uint32_t i, type;
if (!AUD_register_card("hda", &a->card, errp)) {
return;
}
a->desc = desc;
a->name = object_get_typename(OBJECT(a));
dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);
for (i = 0; i < a->desc->nnodes; i++) {
node = a->desc->nodes + i;
param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
if (param == NULL) {
continue;
}
type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
switch (type) {
case AC_WID_AUD_OUT:
case AC_WID_AUD_IN:
assert(node->stindex < ARRAY_SIZE(a->st));
st = a->st + node->stindex;
st->state = a;
st->node = node;
if (type == AC_WID_AUD_OUT) {
/* unmute output by default */
st->gain_left = QEMU_HDA_AMP_STEPS;
st->gain_right = QEMU_HDA_AMP_STEPS;
st->compat_bpos = sizeof(st->compat_buf);
st->output = true;
} else {
st->output = false;
}
st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
(1 << AC_FMT_CHAN_SHIFT);
hda_codec_parse_fmt(st->format, &st->as);
hda_audio_setup(st);
break;
}
}
}
static void hda_audio_exit(HDACodecDevice *hda)
{
HDAAudioState *a = HDA_AUDIO(hda);
int i;
dprint(a, 1, "%s\n", __func__);
for (i = 0; i < ARRAY_SIZE(a->st); i++) {
hda_close_stream(a, a->st + i);
}
AUD_remove_card(&a->card);
}
static int hda_audio_post_load(void *opaque, int version)
{
HDAAudioState *a = opaque;
HDAAudioStream *st;
int i;
dprint(a, 1, "%s\n", __func__);
if (version == 1) {
/* assume running_compat[] is for output streams */
for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
a->running_real[16 + i] = a->running_compat[i];
}
for (i = 0; i < ARRAY_SIZE(a->st); i++) {
st = a->st + i;
if (st->node == NULL)
continue;
hda_codec_parse_fmt(st->format, &st->as);
hda_audio_setup(st);
hda_audio_set_amp(st);
hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
}
return 0;
}
static void hda_audio_reset(DeviceState *dev)
{
HDAAudioState *a = HDA_AUDIO(dev);
HDAAudioStream *st;
int i;
dprint(a, 1, "%s\n", __func__);
for (i = 0; i < ARRAY_SIZE(a->st); i++) {
st = a->st + i;
if (st->node != NULL) {
hda_audio_set_running(st, false);
}
}
}
static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
{
HDAAudioStream *st = opaque;
return st->state && 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 = (const 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,
.fields = (const VMStateField[]) {
VMSTATE_UINT32(stream, HDAAudioStream),
VMSTATE_UINT32(channel, HDAAudioStream),
VMSTATE_UINT32(format, HDAAudioStream),
VMSTATE_UINT32(gain_left, HDAAudioStream),
VMSTATE_UINT32(gain_right, HDAAudioStream),
VMSTATE_BOOL(mute_left, HDAAudioStream),
VMSTATE_BOOL(mute_right, HDAAudioStream),
VMSTATE_UINT32(compat_bpos, HDAAudioStream),
VMSTATE_BUFFER(compat_buf, HDAAudioStream),
VMSTATE_END_OF_LIST()
},
.subsections = (const VMStateDescription * const []) {
&vmstate_hda_audio_stream_buf,
NULL
}
};
static const VMStateDescription vmstate_hda_audio = {
.name = "hda-audio",
.version_id = 2,
.post_load = hda_audio_post_load,
.fields = (const VMStateField[]) {
VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
vmstate_hda_audio_stream,
HDAAudioStream),
VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
VMSTATE_END_OF_LIST()
}
};
static Property hda_audio_properties[] = {
DEFINE_AUDIO_PROPERTIES(HDAAudioState, card),
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(),
};
static void hda_audio_init_output(HDACodecDevice *hda, Error **errp)
{
HDAAudioState *a = HDA_AUDIO(hda);
const struct desc_codec *desc = &output_mixemu;
if (!a->mixer) {
desc = &output_nomixemu;
}
hda_audio_init(hda, desc, errp);
}
static void hda_audio_init_duplex(HDACodecDevice *hda, Error **errp)
{
HDAAudioState *a = HDA_AUDIO(hda);
const struct desc_codec *desc = &duplex_mixemu;
if (!a->mixer) {
desc = &duplex_nomixemu;
}
hda_audio_init(hda, desc, errp);
}
static void hda_audio_init_micro(HDACodecDevice *hda, Error **errp)
{
HDAAudioState *a = HDA_AUDIO(hda);
const struct desc_codec *desc = &micro_mixemu;
if (!a->mixer) {
desc = &micro_nomixemu;
}
hda_audio_init(hda, desc, errp);
}
static void hda_audio_base_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
k->exit = hda_audio_exit;
k->command = hda_audio_command;
k->stream = hda_audio_stream;
set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
device_class_set_legacy_reset(dc, hda_audio_reset);
dc->vmsd = &vmstate_hda_audio;
device_class_set_props(dc, hda_audio_properties);
}
static const TypeInfo hda_audio_info = {
.name = TYPE_HDA_AUDIO,
.parent = TYPE_HDA_CODEC_DEVICE,
.instance_size = sizeof(HDAAudioState),
.class_init = hda_audio_base_class_init,
.abstract = true,
};
static void hda_audio_output_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
k->init = hda_audio_init_output;
dc->desc = "HDA Audio Codec, output-only (line-out)";
}
static const TypeInfo hda_audio_output_info = {
.name = "hda-output",
.parent = TYPE_HDA_AUDIO,
.class_init = hda_audio_output_class_init,
};
static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
k->init = hda_audio_init_duplex;
dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
}
static const TypeInfo hda_audio_duplex_info = {
.name = "hda-duplex",
.parent = TYPE_HDA_AUDIO,
.class_init = hda_audio_duplex_class_init,
};
static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
{
DeviceClass *dc = DEVICE_CLASS(klass);
HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
k->init = hda_audio_init_micro;
dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
}
static const TypeInfo hda_audio_micro_info = {
.name = "hda-micro",
.parent = TYPE_HDA_AUDIO,
.class_init = hda_audio_micro_class_init,
};
static void hda_audio_register_types(void)
{
type_register_static(&hda_audio_info);
type_register_static(&hda_audio_output_info);
type_register_static(&hda_audio_duplex_info);
type_register_static(&hda_audio_micro_info);
}
type_init(hda_audio_register_types)
Reference in New Issue View Git Blame Copy Permalink