diff options
-rw-r--r-- | include/sound/pcm.h | 35 | ||||
-rw-r--r-- | include/sound/pcm_params.h | 16 | ||||
-rw-r--r-- | include/sound/soc-dapm.h | 6 | ||||
-rw-r--r-- | sound/core/pcm_lib.c | 13 | ||||
-rw-r--r-- | sound/pci/asihpi/asihpi.c | 21 | ||||
-rw-r--r-- | sound/pci/hda/Kconfig | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 114 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 15 | ||||
-rw-r--r-- | sound/pci/hda/hda_local.h | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_analog.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/patch_cirrus.c | 743 | ||||
-rw-r--r-- | sound/pci/hda/patch_conexant.c | 14 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 15 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 122 | ||||
-rw-r--r-- | sound/pci/hda/patch_via.c | 4 | ||||
-rw-r--r-- | sound/soc/codecs/sgtl5000.c | 70 | ||||
-rw-r--r-- | sound/soc/codecs/wm8962.c | 6 | ||||
-rw-r--r-- | sound/soc/davinci/davinci-vcif.c | 9 | ||||
-rw-r--r-- | sound/soc/samsung/i2s.c | 7 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 5 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 30 |
21 files changed, 1035 insertions, 217 deletions
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e1bad113061..57e71fa33f7 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -507,6 +507,18 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); + +#ifdef CONFIG_SND_DEBUG +void snd_pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len); +#else +static inline void +snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) +{ + *buf = 0; +} +#endif + /* * PCM library */ @@ -749,17 +761,18 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc return ¶ms->intervals[var - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL]; } -#define params_access(p) ((__force snd_pcm_access_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_ACCESS))) -#define params_format(p) ((__force snd_pcm_format_t)snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_FORMAT))) -#define params_subformat(p) snd_mask_min(hw_param_mask((p), SNDRV_PCM_HW_PARAM_SUBFORMAT)) -#define params_channels(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min -#define params_rate(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_RATE)->min -#define params_period_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min -#define params_period_bytes(p) ((params_period_size(p)*snd_pcm_format_physical_width(params_format(p))*params_channels(p))/8) -#define params_periods(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_PERIODS)->min -#define params_buffer_size(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min -#define params_buffer_bytes(p) hw_param_interval((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min - +#define params_channels(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_CHANNELS)->min) +#define params_rate(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_RATE)->min) +#define params_period_size(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIOD_SIZE)->min) +#define params_periods(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_PERIODS)->min) +#define params_buffer_size(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_SIZE)->min) +#define params_buffer_bytes(p) \ + (hw_param_interval_c((p), SNDRV_PCM_HW_PARAM_BUFFER_BYTES)->min) int snd_interval_refine(struct snd_interval *i, const struct snd_interval *v); void snd_interval_mul(const struct snd_interval *a, const struct snd_interval *b, struct snd_interval *c); diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 85cf1cf4f31..f494f1e3c90 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -337,5 +337,19 @@ static inline unsigned int sub(unsigned int a, unsigned int b) return 0; } -#endif /* __SOUND_PCM_PARAMS_H */ +#define params_access(p) ((__force snd_pcm_access_t)\ + snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS))) +#define params_format(p) ((__force snd_pcm_format_t)\ + snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_FORMAT))) +#define params_subformat(p) \ + snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_SUBFORMAT)) +static inline unsigned int +params_period_bytes(const struct snd_pcm_hw_params *p) +{ + return (params_period_size(p) * + snd_pcm_format_physical_width(params_format(p)) * + params_channels(p)) / 8; +} + +#endif /* __SOUND_PCM_PARAMS_H */ diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c5a49..e0583b7769c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -266,6 +266,12 @@ .get = snd_soc_dapm_get_enum_virt, \ .put = snd_soc_dapm_put_enum_virt, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = xget, \ + .put = xput, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f1341308bed..86d0caf91b3 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -128,7 +128,8 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } -static void pcm_debug_name(struct snd_pcm_substream *substream, +#ifdef CONFIG_SND_DEBUG +void snd_pcm_debug_name(struct snd_pcm_substream *substream, char *name, size_t len) { snprintf(name, len, "pcmC%dD%d%c:%d", @@ -137,6 +138,8 @@ static void pcm_debug_name(struct snd_pcm_substream *substream, substream->stream ? 'c' : 'p', substream->number); } +EXPORT_SYMBOL(snd_pcm_debug_name); +#endif #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ @@ -168,7 +171,7 @@ static void xrun(struct snd_pcm_substream *substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); dump_stack_on_xrun(substream); } @@ -243,7 +246,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) return; if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) return; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { entry = &log->entries[idx]; if (entry->period_size == 0) @@ -319,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (pos >= runtime->buffer_size) { if (printk_ratelimit()) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = %ld, " "buffer size = %ld, period size = %ld\n", @@ -364,7 +367,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (xrun_debug(substream, in_interrupt ? XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd("%s_update: %s: pos=%u/%u/%u, " "hwptr=%ld/%ld/%ld/%ld\n", in_interrupt ? "period" : "hwptr", diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index b941d2541dd..eae62ebbd29 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -41,31 +41,10 @@ #include <sound/tlv.h> #include <sound/hwdep.h> - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. <support@audioscience.com>"); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); -#if defined CONFIG_SND_DEBUG -/* copied from pcm_lib.c, hope later patch will make that version public -and this copy can be removed */ -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - snprintf(buf, size, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} -#else -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - *buf = 0; -} -#endif - #if defined CONFIG_SND_DEBUG_VERBOSE /** * snd_printddd - very verbose debug printk diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7489b460855..bb7e102d672 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -243,6 +243,7 @@ config SND_HDA_GENERIC config SND_HDA_POWER_SAVE bool "Aggressive power-saving on HD-audio" + depends on PM help Say Y here to enable more aggressive power-saving mode on HD-audio driver. The power-saving timeout can be configured diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9c27a3a4c4d..3e7850c238c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -91,8 +91,10 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); +#define hda_codec_is_power_on(codec) ((codec)->power_on) #else static inline void hda_keep_power_on(struct hda_codec *codec) {} +#define hda_codec_is_power_on(codec) 1 #endif /** @@ -1101,7 +1103,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { @@ -1499,7 +1501,7 @@ static void purify_inactive_streams(struct hda_codec *codec) } } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { @@ -1838,7 +1840,7 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /** * snd_hda_codec_resume_amp - Resume all AMP commands from the cache * @codec: HD-audio codec @@ -1868,7 +1870,7 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } } EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ static u32 get_amp_max_value(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int ofs) @@ -3082,7 +3084,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_in_ctls); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* * command cache */ @@ -3199,53 +3201,32 @@ void snd_hda_sequence_write_cache(struct hda_codec *codec, seq->param); } EXPORT_SYMBOL_HDA(snd_hda_sequence_write_cache); -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ -/* - * set power state of the codec - */ -static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state) +void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state, + bool eapd_workaround) { - hda_nid_t nid; + hda_nid_t nid = codec->start_nid; int i; - /* this delay seems necessary to avoid click noise at power-down */ - if (power_state == AC_PWRST_D3) - msleep(100); - snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, - power_state); - /* partial workaround for "azx_get_response timeout" */ - if (power_state == AC_PWRST_D0 && - (codec->vendor_id & 0xffff0000) == 0x14f10000) - msleep(10); - - nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int wcaps = get_wcaps(codec, nid); - if (wcaps & AC_WCAP_POWER) { - unsigned int wid_type = get_wcaps_type(wcaps); - if (power_state == AC_PWRST_D3 && - wid_type == AC_WID_PIN) { - unsigned int pincap; - /* - * don't power down the widget if it controls - * eapd and EAPD_BTLENABLE is set. - */ - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_EAPD) { - int eapd = snd_hda_codec_read(codec, - nid, 0, + if (!(wcaps & AC_WCAP_POWER)) + continue; + /* don't power down the widget if it controls eapd and + * EAPD_BTLENABLE is set. + */ + if (eapd_workaround && power_state == AC_PWRST_D3 && + get_wcaps_type(wcaps) == AC_WID_PIN && + (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD)) { + int eapd = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_EAPD_BTLENABLE, 0); - eapd &= 0x02; - if (eapd) - continue; - } - } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - power_state); + if (eapd & 0x02) + continue; } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, + power_state); } if (power_state == AC_PWRST_D0) { @@ -3262,6 +3243,26 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, } while (time_after_eq(end_time, jiffies)); } } +EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); + +/* + * set power state of the codec + */ +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->patch_ops.set_power_state) { + codec->patch_ops.set_power_state(codec, fg, power_state); + return; + } + + /* this delay seems necessary to avoid click noise at power-down */ + if (power_state == AC_PWRST_D3) + msleep(100); + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); +} #ifdef CONFIG_SND_HDA_HWDEP /* execute additional init verbs */ @@ -3274,7 +3275,7 @@ static void hda_exec_init_verbs(struct hda_codec *codec) static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM /* * call suspend and power-down; used both from PM and power-save */ @@ -3315,7 +3316,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); } } -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ /** @@ -4071,9 +4072,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); #ifdef CONFIG_SND_HDA_POWER_SAVE -static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, - unsigned int power_state); - static void hda_power_work(struct work_struct *work) { struct hda_codec *codec = @@ -4376,11 +4374,8 @@ void snd_hda_bus_reboot_notify(struct hda_bus *bus) if (!bus) return; list_for_each_entry(codec, &bus->codec_list, list) { -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!codec->power_on) - continue; -#endif - if (codec->patch_ops.reboot_notify) + if (hda_codec_is_power_on(codec) && + codec->patch_ops.reboot_notify) codec->patch_ops.reboot_notify(codec); } } @@ -5079,11 +5074,10 @@ int snd_hda_suspend(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!codec->power_on) - continue; -#endif - hda_call_codec_suspend(codec); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec); + if (codec->patch_ops.post_suspend) + codec->patch_ops.post_suspend(codec); } return 0; } @@ -5103,6 +5097,8 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { + if (codec->patch_ops.pre_resume) + codec->patch_ops.pre_resume(codec); if (snd_hda_codec_needs_resume(codec)) hda_call_codec_resume(codec); } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index f465e07a487..755f2b0f9d8 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -26,10 +26,6 @@ #include <sound/pcm.h> #include <sound/hwdep.h> -#if defined(CONFIG_PM) || defined(CONFIG_SND_HDA_POWER_SAVE) -#define SND_HDA_NEEDS_RESUME /* resume control code is required */ -#endif - /* * nodes */ @@ -704,8 +700,12 @@ struct hda_codec_ops { int (*init)(struct hda_codec *codec); void (*free)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); -#ifdef SND_HDA_NEEDS_RESUME + void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); +#ifdef CONFIG_PM int (*suspend)(struct hda_codec *codec, pm_message_t state); + int (*post_suspend)(struct hda_codec *codec); + int (*pre_resume)(struct hda_codec *codec); int (*resume)(struct hda_codec *codec); #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -927,7 +927,7 @@ void snd_hda_sequence_write(struct hda_codec *codec, int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex); /* cached write */ -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm); void snd_hda_sequence_write_cache(struct hda_codec *codec, @@ -1008,6 +1008,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); void snd_hda_bus_reboot_notify(struct hda_bus *bus); +void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state, + bool eapd_workaround); /* * power management diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 88b277e9740..2e7ac31afa8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -131,7 +131,7 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val); int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int dir, int idx, int mask, int val); -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM void snd_hda_codec_resume_amp(struct hda_codec *codec); #endif diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1362c8ba4d1..8648917acff 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -563,7 +563,7 @@ static void ad198x_free(struct hda_codec *codec) snd_hda_detach_beep_device(codec); } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) { ad198x_shutup(codec); @@ -579,7 +579,7 @@ static const struct hda_codec_ops ad198x_patch_ops = { #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = ad198x_check_power_status, #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = ad198x_suspend, #endif .reboot_notify = ad198x_shutup, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7f93739b1e3..47d6ffc9b5b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include <sound/tlv.h> /* */ @@ -61,9 +62,15 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; -/* available models */ +/* available models with CS420x */ enum { CS420X_MBP53, CS420X_MBP55, @@ -72,6 +79,12 @@ enum { CS420X_MODELS }; +/* CS421x boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + /* Vendor-specific processing widget */ #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 @@ -111,21 +124,42 @@ enum { /* 0x0009 - 0x0014 -> 12 test regs */ /* 0x0015 - visibility reg */ +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -347,15 +381,12 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int type; - int idx; type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; - idx = snd_hda_get_conn_index(codec, nid, pin, 0); - if (idx >= 0) { - *idxp = idx; + *idxp = snd_hda_get_conn_index(codec, nid, pin, false); + if (*idxp >= 0) return nid; - } } return 0; } @@ -835,6 +866,8 @@ static int build_digital_input(struct hda_codec *codec) /* * auto-mute and auto-mic switching + * CS421x auto-output redirecting + * HP/SPK/SPDIF */ static void cs_automute(struct hda_codec *codec) @@ -842,9 +875,25 @@ static void cs_automute(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int hp_present; + unsigned int spdif_present; hda_nid_t nid; int i; + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + if (is_jack_detectable(codec, nid)) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; @@ -854,11 +903,19 @@ static void cs_automute(struct hda_codec *codec) if (hp_present) break; } + + /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { nid = cfg->speaker_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); + /* detect on spdif is specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? 0 : PIN_OUT); + } } if (spec->board_config == CS420X_MBP53 || spec->board_config == CS420X_MBP55 || @@ -867,21 +924,62 @@ static void cs_automute(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } + + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present && spec->sense_b) ? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT : 0); + + } + /* Update board GPIOs if neccessary ... */ + } } +/* + * Auto-input redirect for CS421x + * Switch max 3 inputs of a single ADC (nid 3) +*/ + static void cs_automic(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; unsigned int present; - + nid = cfg->inputs[spec->automic_idx].pin; present = snd_hda_jack_detect(codec, nid); - if (present) - change_cur_input(codec, spec->automic_idx, 0); - else - change_cur_input(codec, !spec->automic_idx, 0); + + /* specific to CS421x, single ADC */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } else { + if (present) + change_cur_input(codec, spec->automic_idx, 0); + else + change_cur_input(codec, !spec->automic_idx, 0); + } } /* @@ -911,23 +1009,28 @@ static void init_output(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); spec->hp_detect = 1; } } + + /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (spec->hp_detect) + + /* SPDIF is enabled on presence detect for CS421x */ + if (spec->hp_detect || spec->spdif_detect) cs_automute(codec); } @@ -961,19 +1064,31 @@ static void init_input(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | MIC_EVENT); } - change_cur_input(codec, spec->cur_input, 1); - if (spec->mic_detect) - cs_automic(codec); - - coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ - if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ - if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is selected in - * IDX_SPDIF_CTL. - */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (spec->mic_detect) + cs_automic(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + } else { + change_cur_input(codec, spec->cur_input, 1); + if (spec->mic_detect) + cs_automic(codec); + + coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + if (is_active_pin(codec, CS_DMIC2_PIN_NID)) + coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + if (is_active_pin(codec, CS_DMIC1_PIN_NID)) + coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + * No effect if SPDIF_OUT2 is + * selected in IDX_SPDIF_CTL. + */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + } } static const struct hda_verb cs_coef_init_verbs[] = { @@ -1221,16 +1336,16 @@ static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + const struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1241,11 +1356,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1272,6 +1389,562 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static const struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static const struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static const struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static const struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: + * 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: + * The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static const struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + unsigned int original_coef = coef; + + coef &= ~0x0003; + coef |= (vol & 0x0003); + if (original_coef == coef) + return 0; + else { + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + return 1; + } +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) { + + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_output(codec); + init_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3)) { + + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char *name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs_automute(codec); + break; + + case MIC_EVENT: + cs_automic(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef CONFIG_PM +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef CONFIG_PM + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", + cs421x_models[spec->board_config]); +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries @@ -1279,11 +1952,13 @@ static int patch_cs420x(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 884f67b8f4e..502fc949945 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -446,6 +446,19 @@ static int conexant_init_jacks(struct hda_codec *codec) return 0; } +static void conexant_set_power(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (power_state == AC_PWRST_D3) + msleep(100); + snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE, + power_state); + /* partial workaround for "azx_get_response timeout" */ + if (power_state == AC_PWRST_D0) + msleep(10); + snd_hda_codec_set_power_to_all(codec, fg, power_state, true); +} + static int conexant_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -588,6 +601,7 @@ static const struct hda_codec_ops conexant_patch_ops = { .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, + .set_power_state = conexant_set_power, #ifdef CONFIG_SND_HDA_POWER_SAVE .suspend = conexant_suspend, #endif diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52ce07534e5..694327ae8b7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2386,7 +2386,7 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) } #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { msleep(150); /* to avoid pop noise */ @@ -2406,7 +2406,7 @@ static const struct hda_codec_ops alc_patch_ops = { .init = alc_init, .free = alc_free, .unsol_event = alc_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .resume = alc_resume, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -2801,7 +2801,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) int i; again: - spec->multiout.num_dacs = 0; + /* set num_dacs once to full for alc_auto_look_for_dac() */ + spec->multiout.num_dacs = cfg->line_outs; spec->multiout.hp_nid = 0; spec->multiout.extra_out_nid[0] = 0; memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); @@ -2834,6 +2835,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) } } + /* re-count num_dacs and squash invalid entries */ + spec->multiout.num_dacs = 0; for (i = 0; i < cfg->line_outs; i++) { if (spec->private_dac_nids[i]) spec->multiout.num_dacs++; @@ -4410,7 +4413,7 @@ static void alc269_shutup(struct hda_codec *codec) } } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int alc269_resume(struct hda_codec *codec) { if ((alc_read_coef_idx(codec, 0) & 0x00ff) == 0x018) { @@ -4433,7 +4436,7 @@ static int alc269_resume(struct hda_codec *codec) hda_call_check_power_status(codec, 0x01); return 0; } -#endif /* SND_HDA_NEEDS_RESUME */ +#endif /* CONFIG_PM */ static void alc269_fixup_hweq(struct hda_codec *codec, const struct alc_fixup *fix, int action) @@ -4725,7 +4728,7 @@ static int patch_alc269(struct hda_codec *codec) spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; #endif if (board_config == ALC_MODEL_AUTO) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 56425a53cf1..fcf4c714210 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -95,6 +95,7 @@ enum { STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, STAC_92HD83XXX_HP, + STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, STAC_92HD83XXX_MODELS }; @@ -1636,10 +1637,17 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = { 0x40f000f0, 0x40f000f0, }; +static const unsigned int hp_cNB11_intquad_pin_configs[10] = { + 0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110, + 0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130, + 0x40f000f0, 0x40f000f0, +}; + static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, [STAC_DELL_S14] = dell_s14_pin_configs, + [STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs, [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, }; @@ -1649,6 +1657,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", [STAC_92HD83XXX_HP] = "hp", + [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", }; @@ -1661,7 +1670,47 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, - "HP", STAC_92HD83XXX_HP), + "HP", STAC_92HD83XXX_HP), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1656, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1657, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1658, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1659, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165A, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3388, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3389, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355B, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355C, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355D, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355E, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x355F, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3560, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358B, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358C, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x358D, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3591, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3592, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593, + "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), {} /* terminator */ }; @@ -4885,7 +4934,18 @@ static void stac927x_proc_hook(struct snd_info_buffer *buffer, #define stac927x_proc_hook NULL #endif -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM +static int stac92xx_pre_resume(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + /* sync mute LED */ + if (spec->gpio_led) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data); + return 0; +} + static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4901,29 +4961,19 @@ static int stac92xx_resume(struct hda_codec *codec) stac_issue_unsol_event(codec, spec->autocfg.line_out_pins[0]); } - /* sync mute LED */ - if (spec->gpio_led) - hda_call_check_power_status(codec, 0x01); return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE /* - * using power check for controlling mute led of HP notebooks - * check for mute state only on Speakers (nid = 0x10) - * - * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise - * the LED is NOT working properly ! - * - * Changed name to reflect that it now works for any designated - * model, not just HP HDX. + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed + * as mute LED state is updated in check_power_status hook */ - -#ifdef CONFIG_SND_HDA_POWER_SAVE -static int stac92xx_hp_check_power_status(struct hda_codec *codec, - hda_nid_t nid) +static int stac92xx_update_led_status(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i, muted = 1; + int i, num_ext_dacs, muted = 1; + hda_nid_t nid; for (i = 0; i < spec->multiout.num_dacs; i++) { nid = spec->multiout.dac_nids[i]; @@ -4933,6 +4983,22 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, break; } } + if (muted && spec->multiout.hp_nid) + if (!(snd_hda_codec_amp_read(codec, + spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* HP is not muted */ + } + num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid); + for (i = 0; muted && i < num_ext_dacs; i++) { + nid = spec->multiout.extra_out_nid[i]; + if (nid == 0) + break; + if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE)) { + muted = 0; /* extra output is not muted */ + } + } if (muted) spec->gpio_data &= ~spec->gpio_led; /* orange */ else @@ -4946,6 +5012,17 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); return 0; } + +/* + * use power check for controlling mute led of HP notebooks + */ +static int stac92xx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + stac92xx_update_led_status(codec); + + return 0; +} #endif static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) @@ -4953,7 +5030,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) stac92xx_shutup(codec); return 0; } -#endif +#endif /* CONFIG_PM */ static const struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, @@ -4961,9 +5038,10 @@ static const struct hda_codec_ops stac92xx_patch_ops = { .init = stac92xx_init, .free = stac92xx_free, .unsol_event = stac92xx_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = stac92xx_suspend, .resume = stac92xx_resume, + .pre_resume = stac92xx_pre_resume, #endif .reboot_notify = stac92xx_shutup, }; @@ -5482,7 +5560,7 @@ again: spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; + stac92xx_check_power_status; } #endif @@ -5810,7 +5888,7 @@ again: spec->gpio_data |= spec->gpio_led; /* register check_power_status callback. */ codec->patch_ops.check_power_status = - stac92xx_hp_check_power_status; + stac92xx_check_power_status; } #endif diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index f38160b00e1..84d8798bf33 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1708,7 +1708,7 @@ static void via_unsol_event(struct hda_codec *codec, via_gpio_control(codec); } -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM static int via_suspend(struct hda_codec *codec, pm_message_t state) { struct via_spec *spec = codec->spec; @@ -1736,7 +1736,7 @@ static const struct hda_codec_ops via_patch_ops = { .init = via_init, .free = via_free, .unsol_event = via_unsol_event, -#ifdef SND_HDA_NEEDS_RESUME +#ifdef CONFIG_PM .suspend = via_suspend, #endif #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ff29380c9ed..76258f2a2ff 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -907,6 +907,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, struct regulator_init_data *init_data, int voltage) { + dev_err(codec->dev, "this setup needs regulator support in the kernel\n"); return -EINVAL; } @@ -1218,6 +1219,34 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) return 0; } +static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int ret; + + /* set internal ldo to 1.2v */ + ret = ldo_regulator_register(codec, &ldo_init_data, LDO_VOLTAGE); + if (ret) { + dev_err(codec->dev, + "Failed to register vddd internal supplies: %d\n", ret); + return ret; + } + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + dev_info(codec->dev, "Using internal LDO instead of VDDD\n"); + return 0; +} + static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { u16 reg; @@ -1235,30 +1264,9 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) if (!ret) external_vddd = 1; else { - /* set internal ldo to 1.2v */ - int voltage = LDO_VOLTAGE; - - ret = ldo_regulator_register(codec, &ldo_init_data, voltage); - if (ret) { - dev_err(codec->dev, - "Failed to register vddd internal supplies: %d\n", - ret); - return ret; - } - - sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - - ret = regulator_bulk_get(codec->dev, - ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, - "Failed to request supplies: %d\n", ret); - + ret = sgtl5000_replace_vddd_with_ldo(codec); + if (ret) return ret; - } } ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), @@ -1287,7 +1295,6 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) * roll back to use internal LDO */ if (external_vddd && rev >= 0x11) { - int voltage = LDO_VOLTAGE; /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); @@ -1295,23 +1302,10 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + ret = sgtl5000_replace_vddd_with_ldo(codec); if (ret) return ret; - sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; - - ret = regulator_bulk_get(codec->dev, - ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); - if (ret) { - ldo_regulator_remove(codec); - dev_err(codec->dev, - "Failed to request supplies: %d\n", ret); - - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8499c563a9b..60d740ebeb5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3409,6 +3409,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; + /* Acknowledge the interrupts */ + snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); + if (active & WM8962_FLL_LOCK_EINT) { dev_dbg(codec->dev, "FLL locked\n"); complete(&wm8962->fll_lock); @@ -3433,9 +3436,6 @@ static irqreturn_t wm8962_irq(int irq, void *data) msecs_to_jiffies(250)); } - /* Acknowledge the interrupts */ - snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); - return IRQ_HANDLED; } diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f3489..1f11525d97e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1568eea31f4..c086b78539e 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -21,6 +21,7 @@ #include <plat/audio.h> #include "dma.h" +#include "idma.h" #include "i2s.h" #include "i2s-regs.h" @@ -60,6 +61,7 @@ struct i2s_dai { /* DMA parameters */ struct s3c_dma_params dma_playback; struct s3c_dma_params dma_capture; + struct s3c_dma_params idma_playback; u32 quirks; u32 suspend_i2smod; u32 suspend_i2scon; @@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); + if (i2s->quirks & QUIRK_SEC_DAI) + idma_reg_addr_init((void *)i2s->addr, + i2s->sec_dai->idma_playback.dma_addr); + probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; @@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; + sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f6621..83ad8ca2749 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -577,6 +577,7 @@ int snd_soc_suspend(struct device *dev) case SND_SOC_BIAS_OFF: codec->driver->suspend(codec, PMSG_SUSPEND); codec->suspended = 1; + codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); @@ -1140,7 +1141,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) } } cpu_dai->probed = 1; - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark cpu_dai as probed and add to card dai list */ list_add(&cpu_dai->card_list, &card->dai_dev_list); } @@ -1171,7 +1172,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) } } - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark codec_dai as probed and add to card dai list */ codec_dai->probed = 1; list_add(&codec_dai->card_list, &card->dai_dev_list); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda06283..7e15914b363 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + else + BUG(); + + /* unreachable */ + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + else + BUG(); + + /* unreachable */ + return NULL; +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) |