| /* |
| * 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) |
