6086a565b0
Clean up includes so that osdep.h is included first and headers which it implies are not included manually. This commit was created with scripts/clean-includes. Signed-off-by: Peter Maydell <peter.maydell@linaro.org> Message-id: 1453138432-8324-1-git-send-email-peter.maydell@linaro.org Signed-off-by: Gerd Hoffmann <kraxel@redhat.com>
733 lines
20 KiB
C
733 lines
20 KiB
C
/*
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* Copyright (C) 2010 Red Hat, Inc.
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*
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* written by Gerd Hoffmann <kraxel@redhat.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License as
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* published by the Free Software Foundation; either version 2 or
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* (at your option) version 3 of the License.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, see <http://www.gnu.org/licenses/>.
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*/
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#include "qemu/osdep.h"
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#include "hw/hw.h"
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#include "hw/pci/pci.h"
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#include "intel-hda.h"
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#include "intel-hda-defs.h"
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#include "audio/audio.h"
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/* -------------------------------------------------------------------------- */
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typedef struct desc_param {
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uint32_t id;
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uint32_t val;
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} desc_param;
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typedef struct desc_node {
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uint32_t nid;
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const char *name;
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const desc_param *params;
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uint32_t nparams;
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uint32_t config;
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uint32_t pinctl;
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uint32_t *conn;
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uint32_t stindex;
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} desc_node;
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typedef struct desc_codec {
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const char *name;
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uint32_t iid;
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const desc_node *nodes;
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uint32_t nnodes;
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} desc_codec;
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static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
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{
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int i;
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for (i = 0; i < node->nparams; i++) {
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if (node->params[i].id == id) {
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return &node->params[i];
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}
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}
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return NULL;
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}
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static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
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{
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int i;
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for (i = 0; i < codec->nnodes; i++) {
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if (codec->nodes[i].nid == nid) {
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return &codec->nodes[i];
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}
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}
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return NULL;
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}
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static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
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{
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if (format & AC_FMT_TYPE_NON_PCM) {
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return;
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}
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as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;
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switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
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case 1: as->freq *= 2; break;
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case 2: as->freq *= 3; break;
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case 3: as->freq *= 4; break;
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}
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switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
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case 1: as->freq /= 2; break;
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case 2: as->freq /= 3; break;
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case 3: as->freq /= 4; break;
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case 4: as->freq /= 5; break;
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case 5: as->freq /= 6; break;
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case 6: as->freq /= 7; break;
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case 7: as->freq /= 8; break;
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}
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switch (format & AC_FMT_BITS_MASK) {
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case AC_FMT_BITS_8: as->fmt = AUD_FMT_S8; break;
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case AC_FMT_BITS_16: as->fmt = AUD_FMT_S16; break;
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case AC_FMT_BITS_32: as->fmt = AUD_FMT_S32; break;
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}
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as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
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}
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/* -------------------------------------------------------------------------- */
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/*
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* HDA codec descriptions
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*/
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/* some defines */
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#define QEMU_HDA_ID_VENDOR 0x1af4
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#define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 | \
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0x1fc /* 16 -> 96 kHz */)
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#define QEMU_HDA_AMP_NONE (0)
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#define QEMU_HDA_AMP_STEPS 0x4a
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#define PARAM mixemu
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#define HDA_MIXER
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#include "hda-codec-common.h"
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#define PARAM nomixemu
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#include "hda-codec-common.h"
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/* -------------------------------------------------------------------------- */
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static const char *fmt2name[] = {
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[ AUD_FMT_U8 ] = "PCM-U8",
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[ AUD_FMT_S8 ] = "PCM-S8",
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[ AUD_FMT_U16 ] = "PCM-U16",
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[ AUD_FMT_S16 ] = "PCM-S16",
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[ AUD_FMT_U32 ] = "PCM-U32",
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[ AUD_FMT_S32 ] = "PCM-S32",
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};
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typedef struct HDAAudioState HDAAudioState;
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typedef struct HDAAudioStream HDAAudioStream;
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struct HDAAudioStream {
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HDAAudioState *state;
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const desc_node *node;
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bool output, running;
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uint32_t stream;
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uint32_t channel;
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uint32_t format;
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uint32_t gain_left, gain_right;
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bool mute_left, mute_right;
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struct audsettings as;
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union {
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SWVoiceIn *in;
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SWVoiceOut *out;
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} voice;
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uint8_t buf[HDA_BUFFER_SIZE];
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uint32_t bpos;
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};
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#define TYPE_HDA_AUDIO "hda-audio"
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#define HDA_AUDIO(obj) OBJECT_CHECK(HDAAudioState, (obj), TYPE_HDA_AUDIO)
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struct HDAAudioState {
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HDACodecDevice hda;
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const char *name;
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QEMUSoundCard card;
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const desc_codec *desc;
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HDAAudioStream st[4];
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bool running_compat[16];
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bool running_real[2 * 16];
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/* properties */
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uint32_t debug;
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bool mixer;
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};
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static void hda_audio_input_cb(void *opaque, int avail)
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{
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HDAAudioStream *st = opaque;
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int recv = 0;
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int len;
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bool rc;
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while (avail - recv >= sizeof(st->buf)) {
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if (st->bpos != sizeof(st->buf)) {
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len = AUD_read(st->voice.in, st->buf + st->bpos,
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sizeof(st->buf) - st->bpos);
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st->bpos += len;
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recv += len;
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if (st->bpos != sizeof(st->buf)) {
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break;
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}
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}
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rc = hda_codec_xfer(&st->state->hda, st->stream, false,
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st->buf, sizeof(st->buf));
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if (!rc) {
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break;
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}
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st->bpos = 0;
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}
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}
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static void hda_audio_output_cb(void *opaque, int avail)
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{
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HDAAudioStream *st = opaque;
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int sent = 0;
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int len;
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bool rc;
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while (avail - sent >= sizeof(st->buf)) {
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if (st->bpos == sizeof(st->buf)) {
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rc = hda_codec_xfer(&st->state->hda, st->stream, true,
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st->buf, sizeof(st->buf));
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if (!rc) {
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break;
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}
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st->bpos = 0;
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}
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len = AUD_write(st->voice.out, st->buf + st->bpos,
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sizeof(st->buf) - st->bpos);
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st->bpos += len;
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sent += len;
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if (st->bpos != sizeof(st->buf)) {
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break;
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}
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}
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}
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static void hda_audio_set_running(HDAAudioStream *st, bool running)
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{
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if (st->node == NULL) {
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return;
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}
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if (st->running == running) {
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return;
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}
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st->running = running;
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dprint(st->state, 1, "%s: %s (stream %d)\n", st->node->name,
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st->running ? "on" : "off", st->stream);
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if (st->output) {
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AUD_set_active_out(st->voice.out, st->running);
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} else {
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AUD_set_active_in(st->voice.in, st->running);
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}
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}
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static void hda_audio_set_amp(HDAAudioStream *st)
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{
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bool muted;
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uint32_t left, right;
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if (st->node == NULL) {
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return;
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}
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muted = st->mute_left && st->mute_right;
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left = st->mute_left ? 0 : st->gain_left;
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right = st->mute_right ? 0 : st->gain_right;
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left = left * 255 / QEMU_HDA_AMP_STEPS;
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right = right * 255 / QEMU_HDA_AMP_STEPS;
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if (!st->state->mixer) {
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return;
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}
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if (st->output) {
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AUD_set_volume_out(st->voice.out, muted, left, right);
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} else {
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AUD_set_volume_in(st->voice.in, muted, left, right);
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}
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}
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static void hda_audio_setup(HDAAudioStream *st)
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{
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if (st->node == NULL) {
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return;
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}
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dprint(st->state, 1, "%s: format: %d x %s @ %d Hz\n",
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st->node->name, st->as.nchannels,
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fmt2name[st->as.fmt], st->as.freq);
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if (st->output) {
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st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
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st->node->name, st,
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hda_audio_output_cb, &st->as);
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} else {
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st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
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st->node->name, st,
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hda_audio_input_cb, &st->as);
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}
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}
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static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
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{
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HDAAudioState *a = HDA_AUDIO(hda);
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HDAAudioStream *st;
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const desc_node *node = NULL;
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const desc_param *param;
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uint32_t verb, payload, response, count, shift;
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if ((data & 0x70000) == 0x70000) {
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/* 12/8 id/payload */
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verb = (data >> 8) & 0xfff;
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payload = data & 0x00ff;
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} else {
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/* 4/16 id/payload */
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verb = (data >> 8) & 0xf00;
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payload = data & 0xffff;
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}
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node = hda_codec_find_node(a->desc, nid);
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if (node == NULL) {
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goto fail;
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}
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dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
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__FUNCTION__, nid, node->name, verb, payload);
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switch (verb) {
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/* all nodes */
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case AC_VERB_PARAMETERS:
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param = hda_codec_find_param(node, payload);
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if (param == NULL) {
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goto fail;
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}
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hda_codec_response(hda, true, param->val);
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break;
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case AC_VERB_GET_SUBSYSTEM_ID:
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hda_codec_response(hda, true, a->desc->iid);
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break;
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/* all functions */
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case AC_VERB_GET_CONNECT_LIST:
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param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
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count = param ? param->val : 0;
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response = 0;
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shift = 0;
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while (payload < count && shift < 32) {
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response |= node->conn[payload] << shift;
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payload++;
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shift += 8;
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}
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hda_codec_response(hda, true, response);
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break;
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/* pin widget */
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case AC_VERB_GET_CONFIG_DEFAULT:
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hda_codec_response(hda, true, node->config);
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break;
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case AC_VERB_GET_PIN_WIDGET_CONTROL:
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hda_codec_response(hda, true, node->pinctl);
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break;
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case AC_VERB_SET_PIN_WIDGET_CONTROL:
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if (node->pinctl != payload) {
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dprint(a, 1, "unhandled pin control bit\n");
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}
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hda_codec_response(hda, true, 0);
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break;
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/* audio in/out widget */
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case AC_VERB_SET_CHANNEL_STREAMID:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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hda_audio_set_running(st, false);
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st->stream = (payload >> 4) & 0x0f;
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st->channel = payload & 0x0f;
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dprint(a, 2, "%s: stream %d, channel %d\n",
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st->node->name, st->stream, st->channel);
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hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
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hda_codec_response(hda, true, 0);
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break;
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case AC_VERB_GET_CONV:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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response = st->stream << 4 | st->channel;
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hda_codec_response(hda, true, response);
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break;
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case AC_VERB_SET_STREAM_FORMAT:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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st->format = payload;
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hda_codec_parse_fmt(st->format, &st->as);
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hda_audio_setup(st);
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hda_codec_response(hda, true, 0);
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break;
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case AC_VERB_GET_STREAM_FORMAT:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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hda_codec_response(hda, true, st->format);
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break;
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case AC_VERB_GET_AMP_GAIN_MUTE:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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if (payload & AC_AMP_GET_LEFT) {
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response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
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} else {
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response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
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}
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hda_codec_response(hda, true, response);
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break;
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case AC_VERB_SET_AMP_GAIN_MUTE:
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st = a->st + node->stindex;
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if (st->node == NULL) {
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goto fail;
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}
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dprint(a, 1, "amp (%s): %s%s%s%s index %d gain %3d %s\n",
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st->node->name,
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(payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
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(payload & AC_AMP_SET_INPUT) ? "i" : "-",
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(payload & AC_AMP_SET_LEFT) ? "l" : "-",
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(payload & AC_AMP_SET_RIGHT) ? "r" : "-",
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(payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
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(payload & AC_AMP_GAIN),
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(payload & AC_AMP_MUTE) ? "muted" : "");
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if (payload & AC_AMP_SET_LEFT) {
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st->gain_left = payload & AC_AMP_GAIN;
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st->mute_left = payload & AC_AMP_MUTE;
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}
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if (payload & AC_AMP_SET_RIGHT) {
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st->gain_right = payload & AC_AMP_GAIN;
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st->mute_right = payload & AC_AMP_MUTE;
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}
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hda_audio_set_amp(st);
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hda_codec_response(hda, true, 0);
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break;
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|
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/* not supported */
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case AC_VERB_SET_POWER_STATE:
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case AC_VERB_GET_POWER_STATE:
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case AC_VERB_GET_SDI_SELECT:
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hda_codec_response(hda, true, 0);
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break;
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default:
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goto fail;
|
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}
|
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return;
|
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fail:
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dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
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__FUNCTION__, nid, node ? node->name : "?", verb, payload);
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hda_codec_response(hda, true, 0);
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}
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|
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static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
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{
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HDAAudioState *a = HDA_AUDIO(hda);
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int s;
|
|
|
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a->running_compat[stnr] = running;
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a->running_real[output * 16 + stnr] = running;
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for (s = 0; s < ARRAY_SIZE(a->st); s++) {
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if (a->st[s].node == NULL) {
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continue;
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}
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if (a->st[s].output != output) {
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continue;
|
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}
|
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if (a->st[s].stream != stnr) {
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continue;
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}
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hda_audio_set_running(&a->st[s], running);
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}
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}
|
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|
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static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
|
|
{
|
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HDAAudioState *a = HDA_AUDIO(hda);
|
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HDAAudioStream *st;
|
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const desc_node *node;
|
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const desc_param *param;
|
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uint32_t i, type;
|
|
|
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a->desc = desc;
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a->name = object_get_typename(OBJECT(a));
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|
dprint(a, 1, "%s: cad %d\n", __FUNCTION__, a->hda.cad);
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|
|
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AUD_register_card("hda", &a->card);
|
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for (i = 0; i < a->desc->nnodes; i++) {
|
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node = a->desc->nodes + i;
|
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param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
|
|
if (param == NULL) {
|
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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)
|