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qemu-patch-raspberry4/hw/audio/hda-codec.c
Juan Quintela d49805aeea savevm: Remove all the unneeded version_minimum_id_old (x86) 
After previous Peter patch, they are redundant.  This way we don't
assign them except when needed.  Once there, there were lots of case
where the ".fields" indentation was wrong:

     .fields = (VMStateField []) {
and
     .fields =      (VMStateField []) {

Change all the combinations to:

     .fields = (VMStateField[]){

The biggest problem (appart from aesthetics) was that checkpatch complained
when we copy&pasted the code from one place to another.

Signed-off-by: Juan Quintela <quintela@redhat.com>
Acked-by: Alexey Kardashevskiy <aik@ozlabs.ru>
Reviewed-by: Michael S. Tsirkin <mst@redhat.com>
2014-06-16 04:55:26 +02:00

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/*
* 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 "hw/hw.h"
#include "hw/pci/pci.h"
#include "intel-hda.h"
#include "intel-hda-defs.h"
#include "audio/audio.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 = AUD_FMT_S8; break;
case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
case AC_FMT_BITS_32: as->fmt = AUD_FMT_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"
/* -------------------------------------------------------------------------- */
static const char *fmt2name[] = {
[ AUD_FMT_U8 ] = "PCM-U8",
[ AUD_FMT_S8 ] = "PCM-S8",
[ AUD_FMT_U16 ] = "PCM-U16",
[ AUD_FMT_S16 ] = "PCM-S16",
[ AUD_FMT_U32 ] = "PCM-U32",
[ AUD_FMT_S32 ] = "PCM-S32",
};
typedef struct HDAAudioState HDAAudioState;
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 buf[HDA_BUFFER_SIZE];
uint32_t bpos;
};
#define TYPE_HDA_AUDIO "hda-audio"
#define HDA_AUDIO(obj) OBJECT_CHECK(HDAAudioState, (obj), TYPE_HDA_AUDIO)
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;
};
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;
}
}
rc = hda_codec_xfer(&st->state->hda, st->stream, false,
st->buf, sizeof(st->buf));
if (!rc) {
break;
}
st->bpos = 0;
}
}
static void hda_audio_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)) {
rc = hda_codec_xfer(&st->state->hda, st->stream, true,
st->buf, sizeof(st->buf));
if (!rc) {
break;
}
st->bpos = 0;
}
len = AUD_write(st->voice.out, st->buf + st->bpos,
sizeof(st->buf) - st->bpos);
st->bpos += len;
sent += len;
if (st->bpos != sizeof(st->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;
dprint(st->state, 1, "%s: %s (stream %d)\n", st->node->name,
st->running ? "on" : "off", st->stream);
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_audio_setup(HDAAudioStream *st)
{
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);
if (st->output) {
st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
st->node->name, st,
hda_audio_output_cb, &st->as);
} else {
st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
st->node->name, st,
hda_audio_input_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",
__FUNCTION__, 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",
__FUNCTION__, 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 int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
{
HDAAudioState *a = HDA_AUDIO(hda);
HDAAudioStream *st;
const desc_node *node;
const desc_param *param;
uint32_t i, type;
a->desc = desc;
a->name = object_get_typename(OBJECT(a));
dprint(a, 1, "%s: cad %d\n", __FUNCTION__, a->hda.cad);
AUD_register_card("hda", &a->card);
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 (NULL == param)
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->bpos = sizeof(st->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;
}
}
return 0;
}
static int hda_audio_exit(HDACodecDevice *hda)
{
HDAAudioState *a = HDA_AUDIO(hda);
HDAAudioStream *st;
int i;
dprint(a, 1, "%s\n", __FUNCTION__);
for (i = 0; i < ARRAY_SIZE(a->st); i++) {
st = a->st + i;
if (st->node == NULL) {
continue;
}
if (st->output) {
AUD_close_out(&a->card, st->voice.out);
} else {
AUD_close_in(&a->card, st->voice.in);
}
}
AUD_remove_card(&a->card);
return 0;
}
static int hda_audio_post_load(void *opaque, int version)
{
HDAAudioState *a = opaque;
HDAAudioStream *st;
int i;
dprint(a, 1, "%s\n", __FUNCTION__);
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 const VMStateDescription vmstate_hda_audio_stream = {
.name = "hda-audio-stream",
.version_id = 1,
.fields = (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(bpos, HDAAudioStream),
VMSTATE_BUFFER(buf, HDAAudioStream),
VMSTATE_END_OF_LIST()
}
};
static const VMStateDescription vmstate_hda_audio = {
.name = "hda-audio",
.version_id = 2,
.post_load = hda_audio_post_load,
.fields = (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_PROP_UINT32("debug", HDAAudioState, debug, 0),
DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer, true),
DEFINE_PROP_END_OF_LIST(),
};
static int hda_audio_init_output(HDACodecDevice *hda)
{
HDAAudioState *a = HDA_AUDIO(hda);
if (!a->mixer) {
return hda_audio_init(hda, &output_nomixemu);
} else {
return hda_audio_init(hda, &output_mixemu);
}
}
static int hda_audio_init_duplex(HDACodecDevice *hda)
{
HDAAudioState *a = HDA_AUDIO(hda);
if (!a->mixer) {
return hda_audio_init(hda, &duplex_nomixemu);
} else {
return hda_audio_init(hda, &duplex_mixemu);
}
}
static int hda_audio_init_micro(HDACodecDevice *hda)
{
HDAAudioState *a = HDA_AUDIO(hda);
if (!a->mixer) {
return hda_audio_init(hda, &micro_nomixemu);
} else {
return hda_audio_init(hda, &micro_mixemu);
}
}
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);
dc->reset = hda_audio_reset;
dc->vmsd = &vmstate_hda_audio;
dc->props = hda_audio_properties;
}
static const TypeInfo hda_audio_info = {
.name = TYPE_HDA_AUDIO,
.parent = TYPE_HDA_CODEC_DEVICE,
.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,
.instance_size = sizeof(HDAAudioState),
.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,
.instance_size = sizeof(HDAAudioState),
.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,
.instance_size = sizeof(HDAAudioState),
.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)
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