summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rw-r--r--sound/core/control.c2
-rw-r--r--sound/core/info.c4
-rw-r--r--sound/core/init.c8
-rw-r--r--sound/core/pcm.c10
-rw-r--r--sound/core/pcm_native.c4
-rw-r--r--sound/core/rawmidi.c2
-rw-r--r--sound/core/rtctimer.c4
-rw-r--r--sound/core/sound_oss.c2
-rw-r--r--sound/core/timer.c4
-rw-r--r--sound/drivers/opl3/opl3_synth.c2
-rw-r--r--sound/drivers/pcsp/pcsp_input.c1
-rw-r--r--sound/oss/pas2.h3
-rw-r--r--sound/oss/pas2_card.c2
-rw-r--r--sound/pci/hda/hda_generic.c6
-rw-r--r--sound/pci/hda/hda_generic.h1
-rw-r--r--sound/pci/hda/hda_intel.c25
-rw-r--r--sound/pci/hda/patch_cmedia.c31
-rw-r--r--sound/pci/hda/patch_conexant.c810
-rw-r--r--sound/pci/hda/patch_realtek.c78
-rw-r--r--sound/pci/hda/patch_via.c8
-rw-r--r--sound/pci/lx6464es/lx_core.c84
-rw-r--r--sound/usb/card.c18
-rw-r--r--sound/usb/mixer.c99
-rw-r--r--sound/usb/mixer.h7
24 files changed, 608 insertions, 607 deletions
diff --git a/sound/core/control.c b/sound/core/control.c
index d8aa206..c228f00 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -151,7 +151,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
if (snd_BUG_ON(!card || !id))
return;
read_lock(&card->ctl_files_rwlock);
-#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
card->mixer_oss_change_count++;
#endif
list_for_each_entry(ctl, &card->ctl_files, list) {
diff --git a/sound/core/info.c b/sound/core/info.c
index e79baa1..7916c07 100644
--- a/sound/core/info.c
+++ b/sound/core/info.c
@@ -540,7 +540,7 @@ int __init snd_info_init(void)
snd_oss_root = entry;
}
#endif
-#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#if IS_ENABLED(CONFIG_SND_SEQUENCER)
{
struct snd_info_entry *entry;
if ((entry = snd_info_create_module_entry(THIS_MODULE, "seq", NULL)) == NULL)
@@ -567,7 +567,7 @@ int __exit snd_info_done(void)
snd_minor_info_done();
snd_info_version_done();
if (snd_proc_root) {
-#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#if IS_ENABLED(CONFIG_SND_SEQUENCER)
snd_info_free_entry(snd_seq_root);
#endif
#ifdef CONFIG_SND_OSSEMUL
diff --git a/sound/core/init.c b/sound/core/init.c
index 0d42fcd..a16d765 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -94,7 +94,7 @@ static int module_slot_match(struct module *module, int idx)
return match;
}
-#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag);
EXPORT_SYMBOL(snd_mixer_oss_notify_callback);
#endif
@@ -394,7 +394,7 @@ int snd_card_disconnect(struct snd_card *card)
/* phase 3: notify all connected devices about disconnection */
/* at this point, they cannot respond to any calls except release() */
-#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
if (snd_mixer_oss_notify_callback)
snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_DISCONNECT);
#endif
@@ -430,7 +430,7 @@ EXPORT_SYMBOL(snd_card_disconnect);
*/
static int snd_card_do_free(struct snd_card *card)
{
-#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
if (snd_mixer_oss_notify_callback)
snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_FREE);
#endif
@@ -723,7 +723,7 @@ int snd_card_register(struct snd_card *card)
snd_cards[card->number] = card;
mutex_unlock(&snd_card_mutex);
init_info_for_card(card);
-#if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
if (snd_mixer_oss_notify_callback)
snd_mixer_oss_notify_callback(card, SND_MIXER_OSS_NOTIFY_REGISTER);
#endif
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index e1e9e0c..091a05c 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -295,7 +295,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state)
return snd_pcm_state_names[(__force int)state];
}
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
#include <linux/soundcard.h>
static const char *snd_pcm_oss_format_name(int format)
@@ -398,7 +398,7 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry,
snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den);
snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size);
snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size);
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
if (substream->oss.oss) {
snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format));
snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels);
@@ -651,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
struct snd_pcm_str *pstr = &pcm->streams[stream];
struct snd_pcm_substream *substream, *prev;
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
mutex_init(&pstr->oss.setup_mutex);
#endif
pstr->stream = stream;
@@ -807,7 +807,7 @@ EXPORT_SYMBOL(snd_pcm_new_internal);
static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
{
struct snd_pcm_substream *substream, *substream_next;
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
struct snd_pcm_oss_setup *setup, *setupn;
#endif
substream = pstr->substream;
@@ -819,7 +819,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
substream = substream_next;
}
snd_pcm_stream_proc_done(pstr);
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
for (setup = pstr->oss.setup_list; setup; setup = setupn) {
setupn = setup->next;
kfree(setup->task_name);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 01a5e05..e366411 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -399,7 +399,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
return -EBADFD;
}
snd_pcm_stream_unlock_irq(substream);
-#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+#if IS_ENABLED(CONFIG_SND_PCM_OSS)
if (!substream->oss.oss)
#endif
if (atomic_read(&substream->mmap_count))
@@ -954,7 +954,7 @@ static struct action_ops snd_pcm_action_stop = {
*
* The state of each stream is then changed to the given state unconditionally.
*
- * Return: Zero if succesful, or a negative error code.
+ * Return: Zero if successful, or a negative error code.
*/
int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state)
{
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 7b596b5..f016be7 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1101,7 +1101,7 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
/**
* snd_rawmidi_transmit_ack - acknowledge the transmission
* @substream: the rawmidi substream
- * @count: the tranferred count
+ * @count: the transferred count
*
* Advances the hardware pointer for the internal output buffer with
* the given size and updates the condition.
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
index e85e72b..b272237 100644
--- a/sound/core/rtctimer.c
+++ b/sound/core/rtctimer.c
@@ -27,7 +27,7 @@
#include <sound/core.h>
#include <sound/timer.h>
-#if defined(CONFIG_RTC) || defined(CONFIG_RTC_MODULE)
+#if IS_ENABLED(CONFIG_RTC)
#include <linux/mc146818rtc.h>
@@ -185,4 +185,4 @@ MODULE_LICENSE("GPL");
MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC));
-#endif /* CONFIG_RTC || CONFIG_RTC_MODULE */
+#endif /* IS_ENABLED(CONFIG_RTC) */
diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c
index 726a49a..5391c5e 100644
--- a/sound/core/sound_oss.c
+++ b/sound/core/sound_oss.c
@@ -21,7 +21,7 @@
#ifdef CONFIG_SND_OSSEMUL
-#if !defined(CONFIG_SOUND) && !(defined(MODULE) && defined(CONFIG_SOUND_MODULE))
+#if !IS_ENABLED(CONFIG_SOUND)
#error "Enable the OSS soundcore multiplexer (CONFIG_SOUND) in the kernel."
#endif
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 6ddcf06..cbec514 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -35,9 +35,9 @@
#include <sound/initval.h>
#include <linux/kmod.h>
-#if defined(CONFIG_SND_HRTIMER) || defined(CONFIG_SND_HRTIMER_MODULE)
+#if IS_ENABLED(CONFIG_SND_HRTIMER)
#define DEFAULT_TIMER_LIMIT 4
-#elif defined(CONFIG_SND_RTCTIMER) || defined(CONFIG_SND_RTCTIMER_MODULE)
+#elif IS_ENABLED(CONFIG_SND_RTCTIMER)
#define DEFAULT_TIMER_LIMIT 2
#else
#define DEFAULT_TIMER_LIMIT 1
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index 742a4b6..ddcc1a3 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -24,7 +24,7 @@
#include <sound/opl3.h>
#include <sound/asound_fm.h>
-#if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE)
+#if IS_ENABLED(CONFIG_SND_SEQUENCER)
#define OPL3_SUPPORT_SYNTH
#endif
diff --git a/sound/drivers/pcsp/pcsp_input.c b/sound/drivers/pcsp/pcsp_input.c
index b874b0a..0ecf8a4 100644
--- a/sound/drivers/pcsp/pcsp_input.c
+++ b/sound/drivers/pcsp/pcsp_input.c
@@ -16,6 +16,7 @@
#include <linux/input.h>
#include <asm/io.h>
#include "pcsp.h"
+#include "pcsp_input.h"
static void pcspkr_do_sound(unsigned int count)
{
diff --git a/sound/oss/pas2.h b/sound/oss/pas2.h
index fa12c55..d19f757 100644
--- a/sound/oss/pas2.h
+++ b/sound/oss/pas2.h
@@ -15,3 +15,6 @@ int pas_init_mixer(void);
/* From pas_midi.c */
void pas_midi_init(void);
void pas_midi_interrupt(void);
+
+/* From pas2_mixer.c*/
+void mix_write(unsigned char data, int ioaddr);
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
index 7004e24..b07954a 100644
--- a/sound/oss/pas2_card.c
+++ b/sound/oss/pas2_card.c
@@ -74,8 +74,6 @@ static char *pas_model_names[] = {
* to support other than the default base address
*/
-extern void mix_write(unsigned char data, int ioaddr);
-
unsigned char pas_read(int ioaddr)
{
return inb(ioaddr + pas_translate_code);
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index d9a09bd..bcd9c71 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -79,7 +79,7 @@ static void free_kctls(struct hda_gen_spec *spec)
snd_array_free(&spec->kctls);
}
-void snd_hda_gen_spec_free(struct hda_gen_spec *spec)
+static void snd_hda_gen_spec_free(struct hda_gen_spec *spec)
{
if (!spec)
return;
@@ -87,7 +87,6 @@ void snd_hda_gen_spec_free(struct hda_gen_spec *spec)
snd_array_free(&spec->paths);
snd_array_free(&spec->loopback_list);
}
-EXPORT_SYMBOL_GPL(snd_hda_gen_spec_free);
/*
* store user hints
@@ -762,7 +761,7 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path,
AC_PWRST_D0);
}
if (enable && path->multi[i])
- snd_hda_codec_write_cache(codec, nid, 0,
+ snd_hda_codec_update_cache(codec, nid, 0,
AC_VERB_SET_CONNECT_SEL,
path->idx[i]);
if (has_amp_in(codec, path, i))
@@ -5350,6 +5349,7 @@ EXPORT_SYMBOL_GPL(snd_hda_gen_init);
*/
void snd_hda_gen_free(struct hda_codec *codec)
{
+ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE);
snd_hda_detach_beep_device(codec);
snd_hda_gen_spec_free(codec->spec);
kfree(codec->spec);
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index c908afb..bb2dea7 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -297,7 +297,6 @@ struct hda_gen_spec {
};
int snd_hda_gen_spec_init(struct hda_gen_spec *spec);
-void snd_hda_gen_spec_free(struct hda_gen_spec *spec);
int snd_hda_gen_init(struct hda_codec *codec);
void snd_hda_gen_free(struct hda_codec *codec);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e354ab1..d8d9bf3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -834,18 +834,6 @@ static unsigned int azx_command_addr(u32 cmd)
return addr;
}
-static unsigned int azx_response_addr(u32 res)
-{
- unsigned int addr = res & 0xf;
-
- if (addr >= AZX_MAX_CODECS) {
- snd_BUG();
- addr = 0;
- }
-
- return addr;
-}
-
/* send a command */
static int azx_corb_send_cmd(struct hda_bus *bus, u32 val)
{
@@ -907,8 +895,15 @@ static void azx_update_rirb(struct azx *chip)
rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
res = le32_to_cpu(chip->rirb.buf[rp]);
- addr = azx_response_addr(res_ex);
- if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
+ addr = res_ex & 0xf;
+ if ((addr >= AZX_MAX_CODECS) || !(chip->codec_mask & (1 << addr))) {
+ snd_printk(KERN_ERR SFX "%s: spurious response %#x:%#x, rp = %d, wp = %d",
+ pci_name(chip->pci),
+ res, res_ex,
+ chip->rirb.rp, wp);
+ snd_BUG();
+ }
+ else if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
snd_hda_queue_unsol_event(chip->bus, res, res_ex);
else if (chip->rirb.cmds[addr]) {
chip->rirb.res[addr] = res;
@@ -4142,7 +4137,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
.driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
{ PCI_DEVICE(0x1102, 0x0012),
.driver_data = AZX_DRIVER_CTHDA | AZX_DCAPS_PRESET_CTHDA },
-#if !defined(CONFIG_SND_CTXFI) && !defined(CONFIG_SND_CTXFI_MODULE)
+#if !IS_ENABLED(CONFIG_SND_CTXFI)
/* the following entry conflicts with snd-ctxfi driver,
* as ctxfi driver mutates from HD-audio to native mode with
* a special command sequence.
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 9c6ce73..139ef30 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -32,6 +32,9 @@
#include "hda_jack.h"
#include "hda_generic.h"
+#undef ENABLE_CMI_STATIC_QUIRKS
+
+#ifdef ENABLE_CMI_STATIC_QUIRKS
#define NUM_PINS 11
@@ -45,10 +48,12 @@ enum {
CMI_AUTO, /* let driver guess it */
CMI_MODELS
};
+#endif /* ENABLE_CMI_STATIC_QUIRKS */
struct cmi_spec {
struct hda_gen_spec gen;
+#ifdef ENABLE_CMI_STATIC_QUIRKS
/* below are only for static models */
int board_config;
@@ -81,8 +86,10 @@ struct cmi_spec {
/* multichannel pins */
struct hda_verb multi_init[9]; /* 2 verbs for each pin + terminator */
+#endif /* ENABLE_CMI_STATIC_QUIRKS */
};
+#ifdef ENABLE_CMI_STATIC_QUIRKS
/*
* input MUX
*/
@@ -566,6 +573,7 @@ static const struct hda_codec_ops cmi9880_patch_ops = {
.init = cmi9880_init,
.free = cmi9880_free,
};
+#endif /* ENABLE_CMI_STATIC_QUIRKS */
/*
* stuff for auto-parser
@@ -588,15 +596,20 @@ static int cmi_parse_auto_config(struct hda_codec *codec)
err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0);
if (err < 0)
- return err;
+ goto error;
err = snd_hda_gen_parse_auto_config(codec, cfg);
if (err < 0)
- return err;
+ goto error;
codec->patch_ops = cmi_auto_patch_ops;
return 0;
+
+ error:
+ snd_hda_gen_free(codec);
+ return err;
}
+
static int patch_cmi9880(struct hda_codec *codec)
{
struct cmi_spec *spec;
@@ -606,6 +619,7 @@ static int patch_cmi9880(struct hda_codec *codec)
return -ENOMEM;
codec->spec = spec;
+#ifdef ENABLE_CMI_STATIC_QUIRKS
spec->board_config = snd_hda_check_board_config(codec, CMI_MODELS,
cmi9880_models,
cmi9880_cfg_tbl);
@@ -615,14 +629,8 @@ static int patch_cmi9880(struct hda_codec *codec)
spec->board_config = CMI_AUTO; /* try everything */
}
- if (spec->board_config == CMI_AUTO) {
- int err = cmi_parse_auto_config(codec);
- if (err < 0) {
- snd_hda_gen_free(codec);
- return err;
- }
- return 0;
- }
+ if (spec->board_config == CMI_AUTO)
+ return cmi_parse_auto_config(codec);
/* copy default DAC NIDs */
memcpy(spec->dac_nids, cmi9880_dac_nids, sizeof(spec->dac_nids));
@@ -669,6 +677,9 @@ static int patch_cmi9880(struct hda_codec *codec)
codec->patch_ops = cmi9880_patch_ops;
return 0;
+#else
+ return cmi_parse_auto_config(codec);
+#endif
}
/*
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index bcf91be..59e3aea 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -35,7 +35,7 @@
#include "hda_jack.h"
#include "hda_generic.h"
-#define ENABLE_CXT_STATIC_QUIRKS
+#undef ENABLE_CXT_STATIC_QUIRKS
#define CXT_PIN_DIR_IN 0x00
#define CXT_PIN_DIR_OUT 0x01
@@ -68,6 +68,12 @@ struct conexant_spec {
unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+ /* OPLC XO specific */
+ bool recording;
+ bool dc_enable;
+ unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */
+ struct nid_path *dc_mode_path;
+
#ifdef ENABLE_CXT_STATIC_QUIRKS
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
@@ -123,19 +129,6 @@ struct conexant_spec {
unsigned int hp_laptop:1;
unsigned int asus:1;
- unsigned int ext_mic_present;
- unsigned int recording;
- void (*capture_prepare)(struct hda_codec *codec);
- void (*capture_cleanup)(struct hda_codec *codec);
-
- /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors)
- * through the microphone jack.
- * When the user enables this through a mixer switch, both internal and
- * external microphones are disabled. Gain is fixed at 0dB. In this mode,
- * we also allow the bias to be configured through a separate mixer
- * control. */
- unsigned int dc_enable;
- unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */
unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
#endif /* ENABLE_CXT_STATIC_QUIRKS */
};
@@ -253,8 +246,6 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct snd_pcm_substream *substream)
{
struct conexant_spec *spec = codec->spec;
- if (spec->capture_prepare)
- spec->capture_prepare(codec);
snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
stream_tag, 0, format);
return 0;
@@ -266,8 +257,6 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
{
struct conexant_spec *spec = codec->spec;
snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
- if (spec->capture_cleanup)
- spec->capture_cleanup(codec);
return 0;
}
@@ -673,14 +662,6 @@ static const struct hda_input_mux cxt5045_capture_source_benq = {
}
};
-static const struct hda_input_mux cxt5045_capture_source_hp530 = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "Internal Mic", 0x2 },
- }
-};
-
/* turn on/off EAPD (+ mute HP) as a master switch */
static int cxt5045_hp_master_sw_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
@@ -796,28 +777,6 @@ static const struct snd_kcontrol_new cxt5045_benq_mixers[] = {
{}
};
-static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
- HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x17, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x17, 0x1, HDA_INPUT),
- HDA_BIND_VOL("Master Playback Volume", &cxt5045_hp_bind_master_vol),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Switch",
- .info = cxt_eapd_info,
- .get = cxt_eapd_get,
- .put = cxt5045_hp_master_sw_put,
- .private_value = 0x10,
- },
-
- {}
-};
-
static const struct hda_verb cxt5045_init_verbs[] = {
/* Line in, Mic */
{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_80 },
@@ -1000,7 +959,6 @@ enum {
CXT5045_LAPTOP_MICSENSE,
CXT5045_LAPTOP_HPMICSENSE,
CXT5045_BENQ,
- CXT5045_LAPTOP_HP530,
#ifdef CONFIG_SND_DEBUG
CXT5045_TEST,
#endif
@@ -1013,7 +971,6 @@ static const char * const cxt5045_models[CXT5045_MODELS] = {
[CXT5045_LAPTOP_MICSENSE] = "laptop-micsense",
[CXT5045_LAPTOP_HPMICSENSE] = "laptop-hpmicsense",
[CXT5045_BENQ] = "benq",
- [CXT5045_LAPTOP_HP530] = "laptop-hp530",
#ifdef CONFIG_SND_DEBUG
[CXT5045_TEST] = "test",
#endif
@@ -1021,8 +978,6 @@ static const char * const cxt5045_models[CXT5045_MODELS] = {
};
static const struct snd_pci_quirk cxt5045_cfg_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
- SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE),
SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP_HPMICSENSE),
@@ -1113,14 +1068,6 @@ static int patch_cxt5045(struct hda_codec *codec)
spec->num_mixers = 2;
codec->patch_ops.init = cxt5045_init;
break;
- case CXT5045_LAPTOP_HP530:
- codec->patch_ops.unsol_event = cxt5045_hp_unsol_event;
- spec->input_mux = &cxt5045_capture_source_hp530;
- spec->num_init_verbs = 2;
- spec->init_verbs[1] = cxt5045_hp_sense_init_verbs;
- spec->mixers[0] = cxt5045_mixers_hp530;
- codec->patch_ops.init = cxt5045_init;
- break;
#ifdef CONFIG_SND_DEBUG
case CXT5045_TEST:
spec->input_mux = &cxt5045_test_capture_source;
@@ -1940,11 +1887,6 @@ static const hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
static const hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
static const hda_nid_t cxt5066_digout_pin_nids[2] = { 0x20, 0x22 };
-/* OLPC's microphone port is DC coupled for use with external sensors,
- * therefore we use a 50% mic bias in order to center the input signal with
- * the DC input range of the codec. */
-#define CXT5066_OLPC_EXT_MIC_BIAS PIN_VREF50
-
static const struct hda_channel_mode cxt5066_modes[1] = {
{ 2, NULL },
};
@@ -1997,88 +1939,6 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
return 1;
}
-static const struct hda_input_mux cxt5066_olpc_dc_bias = {
- .num_items = 3,
- .items = {
- { "Off", PIN_IN },
- { "50%", PIN_VREF50 },
- { "80%", PIN_VREF80 },
- },
-};
-
-static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- /* Even though port F is the DC input, the bias is controlled on port B.
- * we also leave that port as an active input (but unselected) in DC mode
- * just in case that is necessary to make the bias setting take effect. */
- return snd_hda_set_pin_ctl_cache(codec, 0x1a,
- cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index);
-}
-
-/* OLPC defers mic widget control until when capture is started because the
- * microphone LED comes on as soon as these settings are put in place. if we
- * did this before recording, it would give the false indication that recording
- * is happening when it is not. */
-static void cxt5066_olpc_select_mic(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- if (!spec->recording)
- return;
-
- if (spec->dc_enable) {
- /* in DC mode we ignore presence detection and just use the jack
- * through our special DC port */
- const struct hda_verb enable_dc_mode[] = {
- /* disble internal mic, port C */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* enable DC capture, port F */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {},
- };
-
- snd_hda_sequence_write(codec, enable_dc_mode);
- /* port B input disabled (and bias set) through the following call */
- cxt5066_set_olpc_dc_bias(codec);
- return;
- }
-
- /* disable DC (port F) */
- snd_hda_set_pin_ctl(codec, 0x1e, 0);
-
- /* external mic, port B */
- snd_hda_set_pin_ctl(codec, 0x1a,
- spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
-
- /* internal mic, port C */
- snd_hda_set_pin_ctl(codec, 0x1b,
- spec->ext_mic_present ? 0 : PIN_VREF80);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void cxt5066_olpc_automic(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- unsigned int present;
-
- if (spec->dc_enable) /* don't do presence detection in DC mode */
- return;
-
- present = snd_hda_codec_read(codec, 0x1a, 0,
- AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
- if (present)
- snd_printdd("CXT5066: external microphone detected\n");
- else
- snd_printdd("CXT5066: external microphone absent\n");
-
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
- present ? 0 : 1);
- spec->ext_mic_present = !!present;
-
- cxt5066_olpc_select_mic(codec);
-}
-
/* toggle input of built-in digital mic and mic jack appropriately */
static void cxt5066_vostro_automic(struct hda_codec *codec)
{
@@ -2252,23 +2112,6 @@ static void cxt5066_automic(struct hda_codec *codec)
}
/* unsolicited event for jack sensing */
-static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- struct conexant_spec *spec = codec->spec;
- snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
- switch (res >> 26) {
- case CONEXANT_HP_EVENT:
- cxt5066_hp_automute(codec);
- break;
- case CONEXANT_MIC_EVENT:
- /* ignore mic events in DC mode; we're always using the jack */
- if (!spec->dc_enable)
- cxt5066_olpc_automic(codec);
- break;
- }
-}
-
-/* unsolicited event for jack sensing */
static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
{
snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
@@ -2338,124 +2181,10 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
idx = imux->num_items - 1;
spec->mic_boost = idx;
- if (!spec->dc_enable)
- cxt5066_set_mic_boost(codec);
- return 1;
-}
-
-static void cxt5066_enable_dc(struct hda_codec *codec)
-{
- const struct hda_verb enable_dc_mode[] = {
- /* disable gain */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* switch to DC input */
- {0x17, AC_VERB_SET_CONNECT_SEL, 3},
- {}
- };
-
- /* configure as input source */
- snd_hda_sequence_write(codec, enable_dc_mode);
- cxt5066_olpc_select_mic(codec); /* also sets configured bias */
-}
-
-static void cxt5066_disable_dc(struct hda_codec *codec)
-{
- /* reconfigure input source */
cxt5066_set_mic_boost(codec);
- /* automic also selects the right mic if we're recording */
- cxt5066_olpc_automic(codec);
-}
-
-static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- ucontrol->value.integer.value[0] = spec->dc_enable;
- return 0;
-}
-
-static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- int dc_enable = !!ucontrol->value.integer.value[0];
-
- if (dc_enable == spec->dc_enable)
- return 0;
-
- spec->dc_enable = dc_enable;
- if (dc_enable)
- cxt5066_enable_dc(codec);
- else
- cxt5066_disable_dc(codec);
-
- return 1;
-}
-
-static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo);
-}
-
-static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- ucontrol->value.enumerated.item[0] = spec->dc_input_bias;
- return 0;
-}
-
-static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct conexant_spec *spec = codec->spec;
- const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
- unsigned int idx;
-
- idx = ucontrol->value.enumerated.item[0];
- if (idx >= imux->num_items)
- idx = imux->num_items - 1;
-
- spec->dc_input_bias = idx;
- if (spec->dc_enable)
- cxt5066_set_olpc_dc_bias(codec);
return 1;
}
-static void cxt5066_olpc_capture_prepare(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- /* mark as recording and configure the microphone widget so that the
- * recording LED comes on. */
- spec->recording = 1;
- cxt5066_olpc_select_mic(codec);
-}
-
-static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- const struct hda_verb disable_mics[] = {
- /* disable external mic, port B */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* disble internal mic, port C */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* disable DC capture, port F */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {},
- };
-
- snd_hda_sequence_write(codec, disable_mics);
- spec->recording = 0;
-}
-
static void conexant_check_dig_outs(struct hda_codec *codec,
const hda_nid_t *dig_pins,
int num_pins)
@@ -2506,43 +2235,6 @@ static const struct snd_kcontrol_new cxt5066_mixer_master[] = {
{}
};
-static const struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Master Playback Volume",
- .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
- SNDRV_CTL_ELEM_ACCESS_TLV_READ |
- SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
- .subdevice = HDA_SUBDEV_AMP_FLAG,
- .info = snd_hda_mixer_amp_volume_info,
- .get = snd_hda_mixer_amp_volume_get,
- .put = snd_hda_mixer_amp_volume_put,
- .tlv = { .c = snd_hda_mixer_amp_tlv },
- /* offset by 28 volume steps to limit minimum gain to -46dB */
- .private_value =
- HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28),
- },
- {}
-};
-
-static const struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "DC Mode Enable Switch",
- .info = snd_ctl_boolean_mono_info,
- .get = cxt5066_olpc_dc_get,
- .put = cxt5066_olpc_dc_put,
- },
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "DC Input Bias Enum",
- .info = cxt5066_olpc_dc_bias_enum_info,
- .get = cxt5066_olpc_dc_bias_enum_get,
- .put = cxt5066_olpc_dc_bias_enum_put,
- },
- {}
-};
-
static const struct snd_kcontrol_new cxt5066_mixers[] = {
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -2633,67 +2325,6 @@ static const struct hda_verb cxt5066_init_verbs[] = {
{ } /* end */
};
-static const struct hda_verb cxt5066_init_verbs_olpc[] = {
- /* Port A: headphones */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* Port B: external microphone */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port C: internal microphone */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port D: unused */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port E: unused, but has primary EAPD */
- {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
-
- /* Port F: external DC input through microphone port */
- {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Port G: internal speakers */
- {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
-
- /* DAC1 */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* DAC2: unused */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-
- /* Disable digital microphone port */
- {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* Audio input selectors */
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-
- /* Disable SPDIF */
- {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-
- /* enable unsolicited events for Port A and B */
- {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
- {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
- { } /* end */
-};
-
static const struct hda_verb cxt5066_init_verbs_vostro[] = {
/* Port A: headphones */
{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
@@ -2889,25 +2520,9 @@ static int cxt5066_init(struct hda_codec *codec)
return 0;
}
-static int cxt5066_olpc_init(struct hda_codec *codec)
-{
- struct conexant_spec *spec = codec->spec;
- snd_printdd("CXT5066: init\n");
- conexant_init(codec);
- cxt5066_hp_automute(codec);
- if (!spec->dc_enable) {
- cxt5066_set_mic_boost(codec);
- cxt5066_olpc_automic(codec);
- } else {
- cxt5066_enable_dc(codec);
- }
- return 0;
-}
-
enum {
CXT5066_LAPTOP, /* Laptops w/ EAPD support */
CXT5066_DELL_LAPTOP, /* Dell Laptop */
- CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */
CXT5066_DELL_VOSTRO, /* Dell Vostro 1015i */
CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */
CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */
@@ -2920,7 +2535,6 @@ enum {
static const char * const cxt5066_models[CXT5066_MODELS] = {
[CXT5066_LAPTOP] = "laptop",
[CXT5066_DELL_LAPTOP] = "dell-laptop",
- [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5",
[CXT5066_DELL_VOSTRO] = "dell-vostro",
[CXT5066_IDEAPAD] = "ideapad",
[CXT5066_THINKPAD] = "thinkpad",
@@ -2941,10 +2555,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS),
SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS),
SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD),
- SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
CXT5066_LAPTOP),
- SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
@@ -3030,32 +2642,11 @@ static int patch_cxt5066(struct hda_codec *codec)
spec->mic_boost = 3; /* default 30dB gain */
break;
- case CXT5066_OLPC_XO_1_5:
- codec->patch_ops.init = cxt5066_olpc_init;
- codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event;
- spec->init_verbs[0] = cxt5066_init_verbs_olpc;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc;
- spec->mixers[spec->num_mixers++] = cxt5066_mixers;
- spec->port_d_mode = 0;
- spec->mic_boost = 3; /* default 30dB gain */
-
- /* no S/PDIF out */
- spec->multiout.dig_out_nid = 0;
-
- /* input source automatically selected */
- spec->input_mux = NULL;
-
- /* our capture hooks which allow us to turn on the microphone LED
- * at the right time */
- spec->capture_prepare = cxt5066_olpc_capture_prepare;
- spec->capture_cleanup = cxt5066_olpc_capture_cleanup;
- break;
case CXT5066_DELL_VOSTRO:
codec->patch_ops.init = cxt5066_init;
codec->patch_ops.unsol_event = cxt5066_unsol_event;
spec->init_verbs[0] = cxt5066_init_verbs_vostro;
- spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
+ spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
spec->mixers[spec->num_mixers++] = cxt5066_mixers;
spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
spec->port_d_mode = 0;
@@ -3207,11 +2798,7 @@ static int cx_auto_init(struct hda_codec *codec)
return 0;
}
-static void cx_auto_free(struct hda_codec *codec)
-{
- snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE);
- snd_hda_gen_free(codec);
-}
+#define cx_auto_free snd_hda_gen_free
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
@@ -3238,6 +2825,11 @@ enum {
CXT_FIXUP_HEADPHONE_MIC,
CXT_FIXUP_GPIO1,
CXT_FIXUP_THINKPAD_ACPI,
+ CXT_FIXUP_OLPC_XO,
+ CXT_FIXUP_CAP_MIX_AMP,
+ CXT_FIXUP_TOSHIBA_P105,
+ CXT_FIXUP_HP_530,
+ CXT_FIXUP_CAP_MIX_AMP_5047,
};
/* for hda_fixup_thinkpad_acpi() */
@@ -3316,6 +2908,288 @@ static void cxt_fixup_headphone_mic(struct hda_codec *codec,
}
}
+/* OPLC XO 1.5 fixup */
+
+/* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors)
+ * through the microphone jack.
+ * When the user enables this through a mixer switch, both internal and
+ * external microphones are disabled. Gain is fixed at 0dB. In this mode,
+ * we also allow the bias to be configured through a separate mixer
+ * control. */
+
+#define update_mic_pin(codec, nid, val) \
+ snd_hda_codec_update_cache(codec, nid, 0, \
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val)
+
+static const struct hda_input_mux olpc_xo_dc_bias = {
+ .num_items = 3,
+ .items = {
+ { "Off", PIN_IN },
+ { "50%", PIN_VREF50 },
+ { "80%", PIN_VREF80 },
+ },
+};
+
+static void olpc_xo_update_mic_boost(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ int ch, val;
+
+ for (ch = 0; ch < 2; ch++) {
+ val = AC_AMP_SET_OUTPUT |
+ (ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT);
+ if (!spec->dc_enable)
+ val |= snd_hda_codec_amp_read(codec, 0x17, ch, HDA_OUTPUT, 0);
+ snd_hda_codec_write(codec, 0x17, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, val);
+ }
+}
+
+static void olpc_xo_update_mic_pins(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ int cur_input, val;
+ struct nid_path *path;
+
+ cur_input = spec->gen.input_paths[0][spec->gen.cur_mux[0]];
+
+ /* Set up mic pins for port-B, C and F dynamically as the recording
+ * LED is turned on/off by these pin controls
+ */
+ if (!spec->dc_enable) {
+ /* disable DC bias path and pin for port F */
+ update_mic_pin(codec, 0x1e, 0);
+ snd_hda_activate_path(codec, spec->dc_mode_path, false, false);
+
+ /* update port B (ext mic) and C (int mic) */
+ /* OLPC defers mic widget control until when capture is
+ * started because the microphone LED comes on as soon as
+ * these settings are put in place. if we did this before
+ * recording, it would give the false indication that
+ * recording is happening when it is not.
+ */
+ update_mic_pin(codec, 0x1a, spec->recording ?
+ snd_hda_codec_get_pin_target(codec, 0x1a) : 0);
+ update_mic_pin(codec, 0x1b, spec->recording ?
+ snd_hda_codec_get_pin_target(codec, 0x1b) : 0);
+ /* enable normal mic path */
+ path = snd_hda_get_path_from_idx(codec, cur_input);
+ if (path)
+ snd_hda_activate_path(codec, path, true, false);
+ } else {
+ /* disable normal mic path */
+ path = snd_hda_get_path_from_idx(codec, cur_input);
+ if (path)
+ snd_hda_activate_path(codec, path, false, false);
+
+ /* Even though port F is the DC input, the bias is controlled
+ * on port B. We also leave that port as an active input (but
+ * unselected) in DC mode just in case that is necessary to
+ * make the bias setting take effect.
+ */
+ if (spec->recording)
+ val = olpc_xo_dc_bias.items[spec->dc_input_bias].index;
+ else
+ val = 0;
+ update_mic_pin(codec, 0x1a, val);
+ update_mic_pin(codec, 0x1b, 0);
+ /* enable DC bias path and pin */
+ update_mic_pin(codec, 0x1e, spec->recording ? PIN_IN : 0);
+ snd_hda_activate_path(codec, spec->dc_mode_path, true, false);
+ }
+}
+
+/* mic_autoswitch hook */
+static void olpc_xo_automic(struct hda_codec *codec, struct hda_jack_tbl *jack)
+{
+ struct conexant_spec *spec = codec->spec;
+ int saved_cached_write = codec->cached_write;
+
+ codec->cached_write = 1;
+ /* in DC mode, we don't handle automic */
+ if (!spec->dc_enable)
+ snd_hda_gen_mic_autoswitch(codec, jack);
+ olpc_xo_update_mic_pins(codec);
+ snd_hda_codec_flush_cache(codec);
+ codec->cached_write = saved_cached_write;
+ if (spec->dc_enable)
+ olpc_xo_update_mic_boost(codec);
+}
+
+/* pcm_capture hook */
+static void olpc_xo_capture_hook(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream,
+ int action)
+{
+ struct conexant_spec *spec = codec->spec;
+
+ /* toggle spec->recording flag and update mic pins accordingly
+ * for turning on/off LED
+ */
+ switch (action) {
+ case HDA_GEN_PCM_ACT_PREPARE:
+ spec->recording = 1;
+ olpc_xo_update_mic_pins(codec);
+ break;
+ case HDA_GEN_PCM_ACT_CLEANUP:
+ spec->recording = 0;
+ olpc_xo_update_mic_pins(codec);
+ break;
+ }
+}
+
+static int olpc_xo_dc_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ ucontrol->value.integer.value[0] = spec->dc_enable;
+ return 0;
+}
+
+static int olpc_xo_dc_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ int dc_enable = !!ucontrol->value.integer.value[0];
+
+ if (dc_enable == spec->dc_enable)
+ return 0;
+
+ spec->dc_enable = dc_enable;
+ olpc_xo_update_mic_pins(codec);
+ olpc_xo_update_mic_boost(codec);
+ return 1;
+}
+
+static int olpc_xo_dc_bias_enum_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ ucontrol->value.enumerated.item[0] = spec->dc_input_bias;
+ return 0;
+}
+
+static int olpc_xo_dc_bias_enum_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ return snd_hda_input_mux_info(&olpc_xo_dc_bias, uinfo);
+}
+
+static int olpc_xo_dc_bias_enum_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ const struct hda_input_mux *imux = &olpc_xo_dc_bias;
+ unsigned int idx;
+
+ idx = ucontrol->value.enumerated.item[0];
+ if (idx >= imux->num_items)
+ idx = imux->num_items - 1;
+ if (spec->dc_input_bias == idx)
+ return 0;
+
+ spec->dc_input_bias = idx;
+ if (spec->dc_enable)
+ olpc_xo_update_mic_pins(codec);
+ return 1;
+}
+
+static const struct snd_kcontrol_new olpc_xo_mixers[] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DC Mode Enable Switch",
+ .info = snd_ctl_boolean_mono_info,
+ .get = olpc_xo_dc_mode_get,
+ .put = olpc_xo_dc_mode_put,
+ },
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "DC Input Bias Enum",
+ .info = olpc_xo_dc_bias_enum_info,
+ .get = olpc_xo_dc_bias_enum_get,
+ .put = olpc_xo_dc_bias_enum_put,
+ },
+ {}
+};
+
+/* overriding mic boost put callback; update mic boost volume only when
+ * DC mode is disabled
+ */
+static int olpc_xo_mic_boost_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct conexant_spec *spec = codec->spec;
+ int ret = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+ if (ret > 0 && spec->dc_enable)
+ olpc_xo_update_mic_boost(codec);
+ return ret;
+}
+
+static void cxt_fixup_olpc_xo(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct conexant_spec *spec = codec->spec;
+ int i;
+
+ if (action != HDA_FIXUP_ACT_PROBE)
+ return;
+
+ spec->gen.mic_autoswitch_hook = olpc_xo_automic;
+ spec->gen.pcm_capture_hook = olpc_xo_capture_hook;
+ spec->dc_mode_path = snd_hda_add_new_path(codec, 0x1e, 0x14, 0);
+
+ snd_hda_add_new_ctls(codec, olpc_xo_mixers);
+
+ /* OLPC's microphone port is DC coupled for use with external sensors,
+ * therefore we use a 50% mic bias in order to center the input signal
+ * with the DC input range of the codec.
+ */
+ snd_hda_codec_set_pin_target(codec, 0x1a, PIN_VREF50);
+
+ /* override mic boost control */
+ for (i = 0; i < spec->gen.kctls.used; i++) {
+ struct snd_kcontrol_new *kctl =
+ snd_array_elem(&spec->gen.kctls, i);
+ if (!strcmp(kctl->name, "Mic Boost Volume")) {
+ kctl->put = olpc_xo_mic_boost_put;
+ break;
+ }
+ }
+}
+
+/*
+ * Fix max input level on mixer widget to 0dB
+ * (originally it has 0x2b steps with 0dB offset 0x14)
+ */
+static void cxt_fixup_cap_mix_amp(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT,
+ (0x14 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x14 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+}
+
+/*
+ * Fix max input level on mixer widget to 0dB
+ * (originally it has 0x1e steps with 0 dB offset 0x17)
+ */
+static void cxt_fixup_cap_mix_amp_5047(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ snd_hda_override_amp_caps(codec, 0x10, HDA_INPUT,
+ (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
+ (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
+ (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
+ (1 << AC_AMPCAP_MUTE_SHIFT));
+}
/* ThinkPad X200 & co with cxt5051 */
static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
@@ -3401,6 +3275,68 @@ static const struct hda_fixup cxt_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = hda_fixup_thinkpad_acpi,
},
+ [CXT_FIXUP_OLPC_XO] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_olpc_xo,
+ },
+ [CXT_FIXUP_CAP_MIX_AMP] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_cap_mix_amp,
+ },
+ [CXT_FIXUP_TOSHIBA_P105] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x10, 0x961701f0 }, /* speaker/hp */
+ { 0x12, 0x02a1901e }, /* ext mic */
+ { 0x14, 0x95a70110 }, /* int mic */
+ {}
+ },
+ },
+ [CXT_FIXUP_HP_530] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ { 0x12, 0x90a60160 }, /* int mic */
+ {}
+ },
+ .chained = true,
+ .chain_id = CXT_FIXUP_CAP_MIX_AMP,
+ },
+ [CXT_FIXUP_CAP_MIX_AMP_5047] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = cxt_fixup_cap_mix_amp_5047,
+ },
+};
+
+static const struct snd_pci_quirk cxt5045_fixups[] = {
+ SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT_FIXUP_HP_530),
+ SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT_FIXUP_TOSHIBA_P105),
+ /* HP, Packard Bell, Fujitsu-Siemens & Lenovo laptops have
+ * really bad sound over 0dB on NID 0x17.
+ */
+ SND_PCI_QUIRK_VENDOR(0x103c, "HP", CXT_FIXUP_CAP_MIX_AMP),
+ SND_PCI_QUIRK_VENDOR(0x1631, "Packard Bell", CXT_FIXUP_CAP_MIX_AMP),
+ SND_PCI_QUIRK_VENDOR(0x1734, "Fujitsu", CXT_FIXUP_CAP_MIX_AMP),
+ SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT_FIXUP_CAP_MIX_AMP),
+ {}
+};
+
+static const struct hda_model_fixup cxt5045_fixup_models[] = {
+ { .id = CXT_FIXUP_CAP_MIX_AMP, .name = "cap-mix-amp" },
+ { .id = CXT_FIXUP_TOSHIBA_P105, .name = "toshiba-p105" },
+ { .id = CXT_FIXUP_HP_530, .name = "hp-530" },
+ {}
+};
+
+static const struct snd_pci_quirk cxt5047_fixups[] = {
+ /* HP laptops have really bad sound over 0 dB on NID 0x10.
+ */
+ SND_PCI_QUIRK_VENDOR(0x103c, "HP", CXT_FIXUP_CAP_MIX_AMP_5047),
+ {}
+};
+
+static const struct hda_model_fixup cxt5047_fixup_models[] = {
+ { .id = CXT_FIXUP_CAP_MIX_AMP_5047, .name = "cap-mix-amp" },
+ {}
};
static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3408,10 +3344,16 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
{}
};
+static const struct hda_model_fixup cxt5051_fixup_models[] = {
+ { .id = CXT_PINCFG_LENOVO_X200, .name = "lenovo-x200" },
+ {}
+};
+
static const struct snd_pci_quirk cxt5066_fixups[] = {
SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1),
SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
+ SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3428,6 +3370,17 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
{}
};
+static const struct hda_model_fixup cxt5066_fixup_models[] = {
+ { .id = CXT_FIXUP_STEREO_DMIC, .name = "stereo-dmic" },
+ { .id = CXT_FIXUP_GPIO1, .name = "gpio1" },
+ { .id = CXT_FIXUP_HEADPHONE_MIC_PIN, .name = "headphone-mic-pin" },
+ { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" },
+ { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" },
+ { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" },
+ { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" },
+ {}
+};
+
/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
* can be created (bko#42825)
*/
@@ -3467,19 +3420,28 @@ static int patch_conexant_auto(struct hda_codec *codec)
switch (codec->vendor_id) {
case 0x14f15045:
codec->single_adc_amp = 1;
+ spec->gen.mixer_nid = 0x17;
+ spec->gen.add_stereo_mix_input = 1;
+ snd_hda_pick_fixup(codec, cxt5045_fixup_models,
+ cxt5045_fixups, cxt_fixups);
break;
case 0x14f15047:
codec->pin_amp_workaround = 1;
spec->gen.mixer_nid = 0x19;
+ spec->gen.add_stereo_mix_input = 1;
+ snd_hda_pick_fixup(codec, cxt5047_fixup_models,
+ cxt5047_fixups, cxt_fixups);
break;
case 0x14f15051:
add_cx5051_fake_mutes(codec);
codec->pin_amp_workaround = 1;
- snd_hda_pick_fixup(codec, NULL, cxt5051_fixups, cxt_fixups);
+ snd_hda_pick_fixup(codec, cxt5051_fixup_models,
+ cxt5051_fixups, cxt_fixups);
break;
default:
codec->pin_amp_workaround = 1;
- snd_hda_pick_fixup(codec, NULL, cxt5066_fixups, cxt_fixups);
+ snd_hda_pick_fixup(codec, cxt5066_fixup_models,
+ cxt5066_fixups, cxt_fixups);
break;
}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0ab0b9e..86857b6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -845,11 +845,7 @@ static inline void alc_shutup(struct hda_codec *codec)
snd_hda_shutup_pins(codec);
}
-static void alc_free(struct hda_codec *codec)
-{
- snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_FREE);
- snd_hda_gen_free(codec);
-}
+#define alc_free snd_hda_gen_free
#ifdef CONFIG_PM
static void alc_power_eapd(struct hda_codec *codec)
@@ -4875,8 +4871,42 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec,
}
}
+/* turn on/off mute LED per vmaster hook */
+static void alc662_led_gpio1_mute_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct alc_spec *spec = codec->spec;
+ unsigned int oldval = spec->gpio_led;
+
+ if (enabled)
+ spec->gpio_led &= ~0x01;
+ else
+ spec->gpio_led |= 0x01;
+ if (spec->gpio_led != oldval)
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->gpio_led);
+}
+
+static void alc662_fixup_led_gpio1(struct hda_codec *codec,
+ const struct hda_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static const struct hda_verb gpio_init[] = {
+ { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 },
+ { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 },
+ {}
+ };
+
+ if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+ spec->gen.vmaster_mute.hook = alc662_led_gpio1_mute_hook;
+ spec->gpio_led = 0;
+ snd_hda_add_verbs(codec, gpio_init);
+ }
+}
+
enum {
ALC662_FIXUP_ASPIRE,
+ ALC662_FIXUP_LED_GPIO1,
ALC662_FIXUP_IDEAPAD,
ALC272_FIXUP_MARIO,
ALC662_FIXUP_CZC_P10T,
@@ -4895,9 +4925,10 @@ enum {
ALC662_FIXUP_INV_DMIC,
ALC668_FIXUP_DELL_MIC_NO_PRESENCE,
ALC668_FIXUP_HEADSET_MODE,
- ALC662_FIXUP_BASS_CHMAP,
+ ALC662_FIXUP_BASS_MODE4_CHMAP,
+ ALC662_FIXUP_BASS_16,
ALC662_FIXUP_BASS_1A,
- ALC662_FIXUP_BASS_1A_CHMAP,
+ ALC662_FIXUP_BASS_CHMAP,
ALC668_FIXUP_AUTO_MUTE,
};
@@ -4909,12 +4940,18 @@ static const struct hda_fixup alc662_fixups[] = {
{ }
}
},
+ [ALC662_FIXUP_LED_GPIO1] = {
+ .type = HDA_FIXUP_FUNC,
+ .v.func = alc662_fixup_led_gpio1,
+ },
[ALC662_FIXUP_IDEAPAD] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
{ 0x17, 0x99130112 }, /* subwoofer */
{ }
- }
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_LED_GPIO1,
},
[ALC272_FIXUP_MARIO] = {
.type = HDA_FIXUP_FUNC,
@@ -5079,24 +5116,33 @@ static const struct hda_fixup alc662_fixups[] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_headset_mode_alc668,
},
- [ALC662_FIXUP_BASS_CHMAP] = {
+ [ALC662_FIXUP_BASS_MODE4_CHMAP] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_bass_chmap,
.chained = true,
.chain_id = ALC662_FIXUP_ASUS_MODE4
},
+ [ALC662_FIXUP_BASS_16] = {
+ .type = HDA_FIXUP_PINS,
+ .v.pins = (const struct hda_pintbl[]) {
+ {0x16, 0x80106111}, /* bass speaker */
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC662_FIXUP_BASS_CHMAP,
+ },
[ALC662_FIXUP_BASS_1A] = {
.type = HDA_FIXUP_PINS,
.v.pins = (const struct hda_pintbl[]) {
{0x1a, 0x80106111}, /* bass speaker */
{}
},
+ .chained = true,
+ .chain_id = ALC662_FIXUP_BASS_CHMAP,
},
- [ALC662_FIXUP_BASS_1A_CHMAP] = {
+ [ALC662_FIXUP_BASS_CHMAP] = {
.type = HDA_FIXUP_FUNC,
.v.func = alc_fixup_bass_chmap,
- .chained = true,
- .chain_id = ALC662_FIXUP_BASS_1A,
},
};
@@ -5118,9 +5164,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE),
SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
- SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP),
- SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP),
- SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP),
+ SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A),
+ SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
+ SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
+ SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16),
+ SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index f84195f..7781662 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -465,14 +465,8 @@ static void via_playback_pcm_hook(struct hda_pcm_stream *hinfo,
static void via_free(struct hda_codec *codec)
{
- struct via_spec *spec = codec->spec;
-
- if (!spec)
- return;
-
vt1708_stop_hp_work(codec);
- snd_hda_gen_spec_free(&spec->gen);
- kfree(spec);
+ snd_hda_gen_free(codec);
}
#ifdef CONFIG_PM
diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c
index 626ecad..df4044d 100644
--- a/sound/pci/lx6464es/lx_core.c
+++ b/sound/pci/lx6464es/lx_core.c
@@ -141,63 +141,6 @@ void lx_plx_reg_write(struct lx6464es *chip, int port, u32 data)
iowrite32(data, address);
}
-u32 lx_plx_mbox_read(struct lx6464es *chip, int mbox_nr)
-{
- int index;
-
- switch (mbox_nr) {
- case 1:
- index = ePLX_MBOX1; break;
- case 2:
- index = ePLX_MBOX2; break;
- case 3:
- index = ePLX_MBOX3; break;
- case 4:
- index = ePLX_MBOX4; break;
- case 5:
- index = ePLX_MBOX5; break;
- case 6:
- index = ePLX_MBOX6; break;
- case 7:
- index = ePLX_MBOX7; break;
- case 0: /* reserved for HF flags */
- snd_BUG();
- default:
- return 0xdeadbeef;
- }
-
- return lx_plx_reg_read(chip, index);
-}
-
-int lx_plx_mbox_write(struct lx6464es *chip, int mbox_nr, u32 value)
-{
- int index = -1;
-
- switch (mbox_nr) {
- case 1:
- index = ePLX_MBOX1; break;
- case 3:
- index = ePLX_MBOX3; break;
- case 4:
- index = ePLX_MBOX4; break;
- case 5:
- index = ePLX_MBOX5; break;
- case 6:
- index = ePLX_MBOX6; break;
- case 7:
- index = ePLX_MBOX7; break;
- case 0: /* reserved for HF flags */
- case 2: /* reserved for Pipe States
- * the DSP keeps an image of it */
- snd_BUG();
- return -EBADRQC;
- }
-
- lx_plx_reg_write(chip, index, value);
- return 0;
-}
-
-
/* rmh */
#ifdef CONFIG_SND_DEBUG
@@ -491,33 +434,6 @@ int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data)
#define CSES_BROADCAST 0x0002
#define CSES_UPDATE_LDSV 0x0004
-int lx_dsp_es_check_pipeline(struct lx6464es *chip)
-{
- int i;
-
- for (i = 0; i != CSES_TIMEOUT; ++i) {
- /*
- * le bit CSES_UPDATE_LDSV est à 1 dés que le macprog
- * est pret. il re-passe à 0 lorsque le premier read a
- * été fait. pour l'instant on retire le test car ce bit
- * passe a 1 environ 200 à 400 ms aprés que le registre
- * confES à été écrit (kick du xilinx ES).
- *
- * On ne teste que le bit CE.
- * */
-
- u32 cses = lx_dsp_reg_read(chip, eReg_CSES);
-
- if ((cses & CSES_CE) == 0)
- return 0;
-
- udelay(1);
- }
-
- return -ETIMEDOUT;
-}
-
-
#define PIPE_INFO_TO_CMD(capture, pipe) \
((u32)((u32)(pipe) | ((capture) ? ID_IS_CAPTURE : 0L)) << ID_OFFSET)
diff --git a/sound/usb/card.c b/sound/usb/card.c
index d979050..0252241 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -691,12 +691,12 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
}
list_for_each_entry(mixer, &chip->mixer_list, list)
- snd_usb_mixer_inactivate(mixer);
+ snd_usb_mixer_suspend(mixer);
return 0;
}
-static int usb_audio_resume(struct usb_interface *intf)
+static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
{
struct snd_usb_audio *chip = usb_get_intfdata(intf);
struct usb_mixer_interface *mixer;
@@ -711,7 +711,7 @@ static int usb_audio_resume(struct usb_interface *intf)
* we just notify and restart the mixers
*/
list_for_each_entry(mixer, &chip->mixer_list, list) {
- err = snd_usb_mixer_activate(mixer);
+ err = snd_usb_mixer_resume(mixer, reset_resume);
if (err < 0)
goto err_out;
}
@@ -723,9 +723,20 @@ static int usb_audio_resume(struct usb_interface *intf)
err_out:
return err;
}
+
+static int usb_audio_resume(struct usb_interface *intf)
+{
+ return __usb_audio_resume(intf, false);
+}
+
+static int usb_audio_reset_resume(struct usb_interface *intf)
+{
+ return __usb_audio_resume(intf, true);
+}
#else
#define usb_audio_suspend NULL
#define usb_audio_resume NULL
+#define usb_audio_reset_resume NULL
#endif /* CONFIG_PM */
static struct usb_device_id usb_audio_ids [] = {
@@ -747,6 +758,7 @@ static struct usb_driver usb_audio_driver = {
.disconnect = usb_audio_disconnect,
.suspend = usb_audio_suspend,
.resume = usb_audio_resume,
+ .reset_resume = usb_audio_reset_resume,
.id_table = usb_audio_ids,
.supports_autosuspend = 1,
};
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 44b0ba4..aa9bc19 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -2299,26 +2299,6 @@ requeue:
}
}
-/* stop any bus activity of a mixer */
-void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer)
-{
- usb_kill_urb(mixer->urb);
- usb_kill_urb(mixer->rc_urb);
-}
-
-int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
-{
- int err;
-
- if (mixer->urb) {
- err = usb_submit_urb(mixer->urb, GFP_NOIO);
- if (err < 0)
- return err;
- }
-
- return 0;
-}
-
/* create the handler for the optional status interrupt endpoint */
static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer)
{
@@ -2417,3 +2397,82 @@ void snd_usb_mixer_disconnect(struct list_head *p)
usb_kill_urb(mixer->urb);
usb_kill_urb(mixer->rc_urb);
}
+
+#ifdef CONFIG_PM
+/* stop any bus activity of a mixer */
+static void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer)
+{
+ usb_kill_urb(mixer->urb);
+ usb_kill_urb(mixer->rc_urb);
+}
+
+static int snd_usb_mixer_activate(struct usb_mixer_interface *mixer)
+{
+ int err;
+
+ if (mixer->urb) {
+ err = usb_submit_urb(mixer->urb, GFP_NOIO);
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer)
+{
+ snd_usb_mixer_inactivate(mixer);
+ return 0;
+}
+
+static int restore_mixer_value(struct usb_mixer_elem_info *cval)
+{
+ int c, err, idx;
+
+ if (cval->cmask) {
+ idx = 0;
+ for (c = 0; c < MAX_CHANNELS; c++) {
+ if (!(cval->cmask & (1 << c)))
+ continue;
+ if (cval->cached & (1 << c)) {
+ err = set_cur_mix_value(cval, c + 1, idx,
+ cval->cache_val[idx]);
+ if (err < 0)
+ return err;
+ }
+ idx++;
+ }
+ } else {
+ /* master */
+ if (cval->cached) {
+ err = set_cur_mix_value(cval, 0, 0, *cval->cache_val);
+ if (err < 0)
+ return err;
+ }
+ }
+
+ return 0;
+}
+
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
+{
+ struct usb_mixer_elem_info *cval;
+ int id, err;
+
+ /* FIXME: any mixer quirks? */
+
+ if (reset_resume) {
+ /* restore cached mixer values */
+ for (id = 0; id < MAX_ID_ELEMS; id++) {
+ for (cval = mixer->id_elems[id]; cval;
+ cval = cval->next_id_elem) {
+ err = restore_mixer_value(cval);
+ if (err < 0)
+ return err;
+ }
+ }
+ }
+
+ return snd_usb_mixer_activate(mixer);
+}
+#endif
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index aab80df..73b1f64 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -63,8 +63,6 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid);
int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval,
int request, int validx, int value_set);
-void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer);
-int snd_usb_mixer_activate(struct usb_mixer_interface *mixer);
int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
struct snd_kcontrol *kctl);
@@ -72,4 +70,9 @@ int snd_usb_mixer_add_control(struct usb_mixer_interface *mixer,
int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *_tlv);
+#ifdef CONFIG_PM
+int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer);
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume);
+#endif
+
#endif /* __USBMIXER_H */
generated by cgit v0.10.2 at 2024-09-10 19:29:25 (GMT)