--- /dev/null
+/*
+ * 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 (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->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, µ_nomixemu);
+ } else {
+ return hda_audio_init(hda, µ_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)
Code Review / kvmfornfv.git / blobdiff