diff options
Diffstat (limited to 'hw/audio/hda-codec.c')
| -rw-r--r-- | hw/audio/hda-codec.c | 960 |
1 files changed, 960 insertions, 0 deletions
diff --git a/hw/audio/hda-codec.c b/hw/audio/hda-codec.c new file mode 100644 index 000000000..feb8f9e2b --- /dev/null +++ b/hw/audio/hda-codec.c @@ -0,0 +1,960 @@ +/* + * 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/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", +}; + +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 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" +OBJECT_DECLARE_SIMPLE_TYPE(HDAAudioState, 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; + bool use_timer; +}; + +static inline int64_t hda_bytes_per_second(HDAAudioStream *st) +{ + return 2LL * 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 (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 buft_start = st->buft_start; + int64_t wpos = st->wpos; + int64_t rpos = st->rpos; + + 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 = 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 buft_start = st->buft_start; + int64_t wpos = st->wpos; + int64_t rpos = st->rpos; + + 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 = 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_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); + + 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 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", __func__, 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->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; + } + } + return 0; +} + +static void hda_audio_exit(HDACodecDevice *hda) +{ + HDAAudioState *a = HDA_AUDIO(hda); + 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) { + continue; + } + if (a->use_timer) { + timer_del(st->buft); + } + 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); +} + +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 = (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 = (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 * []) { + &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 = (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 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; + 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) |