diff options
Diffstat (limited to 'sound/pci')
37 files changed, 3200 insertions, 277 deletions
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 6320bf084e47..e120a11c69e8 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -448,7 +448,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97) if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) { snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read); #ifdef CONFIG_SND_DEBUG - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = snd_ac97_proc_regs_write; #endif if (snd_info_register(entry) < 0) { @@ -474,7 +474,7 @@ void snd_ac97_bus_proc_init(struct snd_ac97_bus * bus) sprintf(name, "codec97#%d", bus->num); if ((entry = snd_info_create_card_entry(bus->card, name, bus->card->proc_root)) != NULL) { - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); entry = NULL; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 0bf2c04eeada..d9c54c08e2db 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -258,7 +258,7 @@ snd_ad1889_ac97_ready(struct snd_ad1889 *chip) while (!(ad1889_readw(chip, AD_AC97_ACIC) & AD_AC97_ACIC_ACRDY) && --retry) - mdelay(1); + usleep_range(1000, 2000); if (!retry) { dev_err(chip->card->dev, "[%s] Link is not ready.\n", __func__); @@ -872,7 +872,7 @@ snd_ad1889_init(struct snd_ad1889 *chip) ad1889_writew(chip, AD_DS_CCS, AD_DS_CCS_CLKEN); /* turn on clock */ ad1889_readw(chip, AD_DS_CCS); /* flush posted write */ - mdelay(10); + usleep_range(10000, 11000); /* enable Master and Target abort interrupts */ ad1889_writel(chip, AD_DMA_DISR, AD_DMA_DISR_PMAE | AD_DMA_DISR_PTAE); diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 720361455c60..64e0961f93ba 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -69,27 +69,27 @@ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; static bool enable_hpi_hwdep = 1; -module_param_array(index, int, NULL, S_IRUGO); +module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "ALSA index value for AudioScience soundcard."); -module_param_array(id, charp, NULL, S_IRUGO); +module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ALSA ID string for AudioScience soundcard."); -module_param_array(enable, bool, NULL, S_IRUGO); +module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "ALSA enable AudioScience soundcard."); -module_param(enable_hpi_hwdep, bool, S_IRUGO|S_IWUSR); +module_param(enable_hpi_hwdep, bool, 0644); MODULE_PARM_DESC(enable_hpi_hwdep, "ALSA enable HPI hwdep for AudioScience soundcard "); /* identify driver */ #ifdef KERNEL_ALSA_BUILD static char *build_info = "Built using headers from kernel source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built using headers from kernel source"); #else static char *build_info = "Built within ALSA source"; -module_param(build_info, charp, S_IRUGO); +module_param(build_info, charp, 0444); MODULE_PARM_DESC(build_info, "Built within ALSA source"); #endif diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index b1a2a7ea4172..7d049569012c 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -46,14 +46,14 @@ MODULE_FIRMWARE("asihpi/dsp8900.bin"); #endif static int prealloc_stream_buf; -module_param(prealloc_stream_buf, int, S_IRUGO); +module_param(prealloc_stream_buf, int, 0444); MODULE_PARM_DESC(prealloc_stream_buf, "Preallocate size for per-adapter stream buffer"); /* Allow the debug level to be changed after module load. E.g. echo 2 > /sys/module/asihpi/parameters/hpiDebugLevel */ -module_param(hpi_debug_level, int, S_IRUGO | S_IWUSR); +module_param(hpi_debug_level, int, 0644); MODULE_PARM_DESC(hpi_debug_level, "debug verbosity 0..5"); /* List of adapters found */ diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 9b2b8b38122f..a2c85cc37972 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -431,7 +431,7 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); entry->c.text.write = snd_ca0106_proc_reg_write32; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); @@ -440,12 +440,12 @@ int snd_ca0106_proc_init(struct snd_ca0106 *emu) if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); entry->c.text.write = snd_ca0106_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { entry->c.text.write = snd_ca0106_proc_i2c_write; entry->private_data = emu; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 26a657870664..452cc79b44af 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1139,7 +1139,7 @@ static int save_mixer_state(struct cmipci *cm) struct snd_ctl_elem_value *val; unsigned int i; - val = kmalloc(sizeof(*val), GFP_ATOMIC); + val = kmalloc(sizeof(*val), GFP_KERNEL); if (!val) return -ENOMEM; for (i = 0; i < CM_SAVED_MIXERS; i++) { diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 655fbea1692c..4910d3f46d4b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -58,7 +58,7 @@ MODULE_PARM_DESC(id, "ID string for the CS46xx soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable CS46xx soundcard."); module_param_array(external_amp, bool, NULL, 0444); -MODULE_PARM_DESC(external_amp, "Force to enable external amplifer."); +MODULE_PARM_DESC(external_amp, "Force to enable external amplifier."); module_param_array(thinkpad, bool, NULL, 0444); MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 0020fd0efc46..ed1251c5f449 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2849,7 +2849,7 @@ static int snd_cs46xx_proc_init(struct snd_card *card, struct snd_cs46xx *chip) entry->private_data = chip; entry->c.ops = &snd_cs46xx_proc_io_ops; entry->size = region->size; - entry->mode = S_IFREG | S_IRUSR; + entry->mode = S_IFREG | 0400; } } #ifdef CONFIG_SND_CS46XX_NEW_DSP diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index aa61615288ff..c44eadef64ae 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -798,7 +798,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; - entry->mode = S_IFDIR | S_IRUGO | S_IXUGO; + entry->mode = S_IFDIR | 0555; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -814,7 +814,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_symbols", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_symbol_table_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -826,7 +826,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "spos_modules", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_modules_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -838,7 +838,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "parameter", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -850,7 +850,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "sample", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_sample_dump_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -862,7 +862,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "task_tree", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_task_tree_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); @@ -874,7 +874,7 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) if ((entry = snd_info_create_card_entry(card, "scb_info", ins->proc_dsp_dir)) != NULL) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = chip; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_read; if (snd_info_register(entry) < 0) { snd_info_free_entry(entry); diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index 7488e1b7a770..abb01ce66983 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -271,7 +271,7 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip, entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = scb_info; - entry->mode = S_IFREG | S_IRUGO | S_IWUSR; + entry->mode = S_IFREG | 0644; entry->c.text.read = cs46xx_dsp_proc_scb_info_read; diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index 08e874e9a7f6..2099e9ce441a 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -17,7 +17,7 @@ static bool use_system_timer; MODULE_PARM_DESC(use_system_timer, "Force to use system-timer"); -module_param(use_system_timer, bool, S_IRUGO); +module_param(use_system_timer, bool, 0444); struct ct_timer_ops { void (*init)(struct ct_timer_instance *); diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index f2f32779de98..b2874220be1b 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -26,9 +26,9 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs, Sound Blaster X-Fi}"); static unsigned int reference_rate = 48000; static unsigned int multiple = 2; MODULE_PARM_DESC(reference_rate, "Reference rate (default=48000)"); -module_param(reference_rate, uint, S_IRUGO); +module_param(reference_rate, uint, 0444); MODULE_PARM_DESC(multiple, "Rate multiplier (default=2)"); -module_param(multiple, uint, S_IRUGO); +module_param(multiple, uint, 0444); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0935a5c8741f..358ef7dcf410 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -59,7 +59,7 @@ static int get_firmware(const struct firmware **fw_entry, dev_dbg(chip->card->dev, "firmware requested: %s\n", card_fw[fw_index].data); snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); - err = request_firmware(fw_entry, name, pci_device(chip)); + err = request_firmware(fw_entry, name, &chip->pci->dev); if (err < 0) dev_err(chip->card->dev, "get_firmware(): Firmware not available (%d)\n", err); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 152ce158583c..44b390a667d5 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -559,10 +559,4 @@ static inline int monitor_index(const struct echoaudio *chip, int out, int in) return out * num_busses_in(chip) + in; } - -#ifndef pci_device -#define pci_device(chip) (&chip->pci->dev) -#endif - - #endif /* _ECHOAUDIO_H_ */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 2c2b12a06177..611589cbdad6 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1070,7 +1070,7 @@ static int snd_emu10k1x_proc_init(struct emu10k1x *emu) if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read); entry->c.text.write = snd_emu10k1x_proc_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->private_data = emu; } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index a2b56b188be4..b45a01bb73e5 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -170,7 +170,7 @@ static char *audigy_outs[32] = { /* 0x0f */ "Rear Right", /* 0x10 */ "AC97 Front Left", /* 0x11 */ "AC97 Front Right", - /* 0x12 */ "ADC Caputre Left", + /* 0x12 */ "ADC Capture Left", /* 0x13 */ "ADC Capture Right", /* 0x14 */ NULL, /* 0x15 */ NULL, @@ -421,14 +421,10 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu, snd_fx8010_irq_handler_t *handler, unsigned char gpr_running, void *private_data, - struct snd_emu10k1_fx8010_irq **r_irq) + struct snd_emu10k1_fx8010_irq *irq) { - struct snd_emu10k1_fx8010_irq *irq; unsigned long flags; - irq = kmalloc(sizeof(*irq), GFP_ATOMIC); - if (irq == NULL) - return -ENOMEM; irq->handler = handler; irq->gpr_running = gpr_running; irq->private_data = private_data; @@ -443,8 +439,6 @@ int snd_emu10k1_fx8010_register_irq_handler(struct snd_emu10k1 *emu, emu->fx8010.irq_handlers = irq; } spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags); - if (r_irq) - *r_irq = irq; return 0; } @@ -468,7 +462,6 @@ int snd_emu10k1_fx8010_unregister_irq_handler(struct snd_emu10k1 *emu, tmp->next = tmp->next->next; } spin_unlock_irqrestore(&emu->fx8010.irq_lock, flags); - kfree(irq); return 0; } diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cefe613ef7b7..d39458ab251f 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1724,7 +1724,7 @@ static int snd_emu10k1_fx8010_playback_trigger(struct snd_pcm_substream *substre case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - snd_emu10k1_fx8010_unregister_irq_handler(emu, pcm->irq); pcm->irq = NULL; + snd_emu10k1_fx8010_unregister_irq_handler(emu, &pcm->irq); snd_emu10k1_ptr_write(emu, emu->gpr_base + pcm->gpr_trigger, 0, 0); pcm->tram_pos = INITIAL_TRAM_POS(pcm->buffer_size); pcm->tram_shift = 0; diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index bde0d1954f56..b57008031792 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -135,7 +135,7 @@ static void snd_emu10k1_proc_read(struct snd_info_entry *entry, /* 15 */ "Rear Right", /* 16 */ "AC97 Front Left", /* 17 */ "AC97 Front Right", - /* 18 */ "ADC Caputre Left", + /* 18 */ "ADC Capture Left", /* 19 */ "ADC Capture Right", /* 20 */ "???", /* 21 */ "???", @@ -574,32 +574,32 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "io_regs", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read); entry->c.text.write = snd_emu_proc_io_reg_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b); entry->c.text.write = snd_emu_proc_ptr_reg_write00; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) { snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c); entry->c.text.write = snd_emu_proc_ptr_reg_write20; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } #endif @@ -621,35 +621,35 @@ int snd_emu10k1_proc_init(struct snd_emu10k1 *emu) if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_GPR : TOTAL_SIZE_GPR; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_data", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_DATA : TOTAL_SIZE_TANKMEM_DATA ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_tram_addr", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_TANKMEM_ADDR : TOTAL_SIZE_TANKMEM_ADDR ; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_code", &entry)) { entry->content = SNDRV_INFO_CONTENT_DATA; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->size = emu->audigy ? A_TOTAL_SIZE_CODE : TOTAL_SIZE_CODE; entry->c.ops = &snd_emu10k1_proc_ops_fx8010; } if (! snd_card_proc_new(emu->card, "fx8010_acode", &entry)) { entry->content = SNDRV_INFO_CONTENT_TEXT; entry->private_data = emu; - entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/; + entry->mode = S_IFREG | 0444 /*| S_IWUSR*/; entry->c.text.read = snd_emu10k1_proc_acode_read; } return 0; diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 5865f3b90b34..dbc7d8d0e1c4 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,13 +248,13 @@ __found_pages: static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) { if (addr & ~emu->dma_mask) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "max memory size is 0x%lx (addr = 0x%lx)!!\n", emu->dma_mask, (unsigned long)addr); return 0; } if (addr & (EMUPAGESIZE-1)) { - dev_err(emu->card->dev, "page is not aligned\n"); + dev_err_ratelimited(emu->card->dev, "page is not aligned\n"); return 0; } return 1; @@ -345,7 +345,7 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst else addr = snd_pcm_sgbuf_get_addr(substream, ofs); if (! is_valid_page(emu, addr)) { - dev_err(emu->card->dev, + dev_err_ratelimited(emu->card->dev, "emu: failure page = %d\n", idx); mutex_unlock(&hdr->block_mutex); return NULL; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f7a492c382d9..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -127,11 +127,15 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" + select SND_DYNAMIC_MINORS help Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. + Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS + to assure the multiple streams for DP-MST support. + comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index d3ea73171a3d..b9a6b66aeb0e 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -793,11 +793,11 @@ EXPORT_SYMBOL_GPL(snd_hda_add_verbs); */ void snd_hda_apply_verbs(struct hda_codec *codec) { + const struct hda_verb **v; int i; - for (i = 0; i < codec->verbs.used; i++) { - struct hda_verb **v = snd_array_elem(&codec->verbs, i); + + snd_array_for_each(&codec->verbs, i, v) snd_hda_sequence_write(codec, *v); - } } EXPORT_SYMBOL_GPL(snd_hda_apply_verbs); @@ -890,10 +890,10 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; u32 cfg = pin->cfg; const struct hda_pintbl *t_pins; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5bc3a7468e17..08151f3c0b13 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -481,9 +481,10 @@ static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, struct snd_array *array, hda_nid_t nid) { + struct hda_pincfg *pin; int i; - for (i = 0; i < array->used; i++) { - struct hda_pincfg *pin = snd_array_elem(array, i); + + snd_array_for_each(array, i, pin) { if (pin->nid == nid) return pin; } @@ -618,14 +619,15 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_get_pin_target); */ void snd_hda_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + /* don't shut up pins when unloading the driver; otherwise it breaks * the default pin setup at the next load of the driver */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ snd_hda_codec_read(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); @@ -638,13 +640,14 @@ EXPORT_SYMBOL_GPL(snd_hda_shutup_pins); /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; + if (!codec->pins_shutup) return; if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { snd_hda_codec_write(codec, pin->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pin->ctrl); @@ -697,8 +700,7 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->nid == nid) return p; } @@ -1076,8 +1078,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, /* make other inactive cvts with the same stream-tag dirty */ type = get_wcaps_type(get_wcaps(codec, nid)); list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (!p->active && p->stream_tag == stream_tag && get_wcaps_type(get_wcaps(c, p->nid)) == type) p->dirty = 1; @@ -1140,12 +1141,11 @@ static void really_cleanup_stream(struct hda_codec *codec, static void purify_inactive_streams(struct hda_codec *codec) { struct hda_codec *c; + struct hda_cvt_setup *p; int i; list_for_each_codec(c, codec->bus) { - for (i = 0; i < c->cvt_setups.used; i++) { - struct hda_cvt_setup *p; - p = snd_array_elem(&c->cvt_setups, i); + snd_array_for_each(&c->cvt_setups, i, p) { if (p->dirty) really_cleanup_stream(c, p); } @@ -1156,10 +1156,10 @@ static void purify_inactive_streams(struct hda_codec *codec) /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { + struct hda_cvt_setup *p; int i; - for (i = 0; i < codec->cvt_setups.used; i++) { - struct hda_cvt_setup *p = snd_array_elem(&codec->cvt_setups, i); + snd_array_for_each(&codec->cvt_setups, i, p) { if (p->stream_tag) really_cleanup_stream(codec, p); } @@ -1493,10 +1493,10 @@ static void get_ctl_amp_tlv(struct snd_kcontrol *kcontrol, unsigned int *tlv) val1 = ((int)val1) * ((int)val2); if (min_mute || (caps & AC_AMPCAP_MIN_MUTE)) val2 |= TLV_DB_SCALE_MUTE; - tlv[0] = SNDRV_CTL_TLVT_DB_SCALE; - tlv[1] = 2 * sizeof(unsigned int); - tlv[2] = val1; - tlv[3] = val2; + tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE; + tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int); + tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = val1; + tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = val2; } /** @@ -1544,10 +1544,10 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; step = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; step = (step + 1) * 25; - tlv[0] = SNDRV_CTL_TLVT_DB_SCALE; - tlv[1] = 2 * sizeof(unsigned int); - tlv[2] = -nums * step; - tlv[3] = step; + tlv[SNDRV_CTL_TLVO_TYPE] = SNDRV_CTL_TLVT_DB_SCALE; + tlv[SNDRV_CTL_TLVO_LEN] = 2 * sizeof(unsigned int); + tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] = -nums * step; + tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] = step; } EXPORT_SYMBOL_GPL(snd_hda_set_vmaster_tlv); @@ -1845,10 +1845,10 @@ static int init_slave_0dB(struct snd_kcontrol *slave, } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) tlv = kctl->tlv.p; - if (!tlv || tlv[0] != SNDRV_CTL_TLVT_DB_SCALE) + if (!tlv || tlv[SNDRV_CTL_TLVO_TYPE] != SNDRV_CTL_TLVT_DB_SCALE) return 0; - step = tlv[3]; + step = tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP]; step &= ~TLV_DB_SCALE_MUTE; if (!step) return 0; @@ -1860,7 +1860,7 @@ static int init_slave_0dB(struct snd_kcontrol *slave, } arg->step = step; - val = -tlv[2] / step; + val = -tlv[SNDRV_CTL_TLVO_DB_SCALE_MIN] / step; if (val > 0) { put_kctl_with_value(slave, val); return val; @@ -2175,6 +2175,8 @@ static int snd_hda_spdif_default_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.iec958.status[0] = spdif->status & 0xff; @@ -2282,6 +2284,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2308,6 +2312,8 @@ static int snd_hda_spdif_out_switch_get(struct snd_kcontrol *kcontrol, int idx = kcontrol->private_value; struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); ucontrol->value.integer.value[0] = spdif->ctls & AC_DIG1_ENABLE; @@ -2336,6 +2342,8 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, unsigned short val; int change; + if (WARN_ON(codec->spdif_out.used <= idx)) + return -EINVAL; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); nid = spdif->nid; @@ -2461,10 +2469,10 @@ EXPORT_SYMBOL_GPL(snd_hda_create_dig_out_ctls); struct hda_spdif_out *snd_hda_spdif_out_of_nid(struct hda_codec *codec, hda_nid_t nid) { + struct hda_spdif_out *spdif; int i; - for (i = 0; i < codec->spdif_out.used; i++) { - struct hda_spdif_out *spdif = - snd_array_elem(&codec->spdif_out, i); + + snd_array_for_each(&codec->spdif_out, i, spdif) { if (spdif->nid == nid) return spdif; } @@ -2483,6 +2491,8 @@ void snd_hda_spdif_ctls_unassign(struct hda_codec *codec, int idx) { struct hda_spdif_out *spdif; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); spdif->nid = (u16)-1; @@ -2503,6 +2513,8 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid) struct hda_spdif_out *spdif; unsigned short val; + if (WARN_ON(codec->spdif_out.used <= idx)) + return; mutex_lock(&codec->spdif_mutex); spdif = snd_array_elem(&codec->spdif_out, idx); if (spdif->nid != nid) { diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5cc65093d941..db773e219aaa 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -264,10 +264,10 @@ static struct nid_path *get_nid_path(struct hda_codec *codec, int anchor_nid) { struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->depth <= 0) continue; if ((!from_nid || path->path[0] == from_nid) && @@ -325,10 +325,10 @@ EXPORT_SYMBOL_GPL(snd_hda_get_path_from_idx); static bool is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if (path->path[0] == nid) return true; } @@ -351,11 +351,11 @@ static bool is_reachable_path(struct hda_codec *codec, static bool is_ctl_used(struct hda_codec *codec, unsigned int val, int type) { struct hda_gen_spec *spec = codec->spec; + const struct nid_path *path; int i; val &= AMP_VAL_COMPARE_MASK; - for (i = 0; i < spec->paths.used; i++) { - struct nid_path *path = snd_array_elem(&spec->paths, i); + snd_array_for_each(&spec->paths, i, path) { if ((path->ctls[type] & AMP_VAL_COMPARE_MASK) == val) return true; } @@ -638,13 +638,13 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, { struct hda_gen_spec *spec = codec->spec; int type = get_wcaps_type(get_wcaps(codec, nid)); + const struct nid_path *path; int i, n; if (nid == codec->core.afg) return true; - for (n = 0; n < spec->paths.used; n++) { - struct nid_path *path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->active) continue; if (codec->power_save_node) { @@ -2065,7 +2065,7 @@ static int parse_output_paths(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, spec->vmaster_tlv); if (spec->dac_min_mute) - spec->vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; + spec->vmaster_tlv[SNDRV_CTL_TLVO_DB_SCALE_MUTE_AND_STEP] |= TLV_DB_SCALE_MUTE; } } @@ -2696,10 +2696,10 @@ static const struct snd_kcontrol_new out_jack_mode_enum = { static bool find_kctl_name(struct hda_codec *codec, const char *name, int idx) { struct hda_gen_spec *spec = codec->spec; + const struct snd_kcontrol_new *kctl; int i; - for (i = 0; i < spec->kctls.used; i++) { - struct snd_kcontrol_new *kctl = snd_array_elem(&spec->kctls, i); + snd_array_for_each(&spec->kctls, i, kctl) { if (!strcmp(kctl->name, name) && kctl->index == idx) return true; } @@ -4021,8 +4021,7 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path; int n; - for (n = 0; n < spec->paths.used; n++) { - path = snd_array_elem(&spec->paths, n); + snd_array_for_each(&spec->paths, n, path) { if (!path->depth) continue; if (path->path[0] == nid || @@ -5831,10 +5830,10 @@ static void init_digital(struct hda_codec *codec) */ static void clear_unsol_on_unused_pins(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { hda_nid_t nid = pin->nid; if (is_jack_detectable(codec, nid) && !snd_hda_jack_tbl_get(codec, nid)) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a0c93b9c9a28..1ae1850b3bfd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2209,7 +2209,18 @@ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1581607 */ + SND_PCI_QUIRK(0x1558, 0x3501, "Clevo W35xSS_370SS", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ + /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P55A-UD3 / Z87-D3HP", 0), + /* https://bugzilla.kernel.org/show_bug.cgi?id=199607 */ + SND_PCI_QUIRK(0x8086, 0x2057, "Intel NUC5i7RYB", 0), + /* https://bugzilla.redhat.com/show_bug.cgi?id=1520902 */ + SND_PCI_QUIRK(0x8086, 0x2068, "Intel NUC7i3BNB", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1572975 */ SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugzilla.kernel.org/show_bug.cgi?id=198611 */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 9b7efece4484..6ec79c58d48d 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -80,10 +80,10 @@ static ssize_t pin_configs_show(struct hda_codec *codec, struct snd_array *list, char *buf) { + const struct hda_pincfg *pin; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < list->used; i++) { - struct hda_pincfg *pin = snd_array_elem(list, i); + snd_array_for_each(list, i, pin) { len += sprintf(buf + len, "0x%02x 0x%08x\n", pin->nid, pin->cfg); } @@ -217,10 +217,10 @@ static ssize_t init_verbs_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_verb *v; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->init_verbs.used; i++) { - struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + snd_array_for_each(&codec->init_verbs, i, v) { len += snprintf(buf + len, PAGE_SIZE - len, "0x%02x 0x%03x 0x%04x\n", v->nid, v->verb, v->param); @@ -267,10 +267,10 @@ static ssize_t hints_show(struct device *dev, char *buf) { struct hda_codec *codec = dev_get_drvdata(dev); + const struct hda_hint *hint; int i, len = 0; mutex_lock(&codec->user_mutex); - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { len += snprintf(buf + len, PAGE_SIZE - len, "%s = %s\n", hint->key, hint->val); } @@ -280,10 +280,10 @@ static ssize_t hints_show(struct device *dev, static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) { + struct hda_hint *hint; int i; - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { if (!strcmp(hint->key, key)) return hint; } @@ -783,13 +783,13 @@ void snd_hda_sysfs_init(struct hda_codec *codec) void snd_hda_sysfs_clear(struct hda_codec *codec) { #ifdef CONFIG_SND_HDA_RECONFIG + struct hda_hint *hint; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - for (i = 0; i < codec->hints.used; i++) { - struct hda_hint *hint = snd_array_elem(&codec->hints, i); + snd_array_for_each(&codec->hints, i, hint) { kfree(hint->key); /* we don't need to free hint->val */ } snd_array_free(&codec->hints); diff --git a/sound/pci/hda/hp_x360_helper.c b/sound/pci/hda/hp_x360_helper.c new file mode 100644 index 000000000000..969542c57358 --- /dev/null +++ b/sound/pci/hda/hp_x360_helper.c @@ -0,0 +1,95 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for HP X360 laptops with top B&O speakers + * to be included from codec driver + */ + +static void alc295_fixup_hp_top_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170110 }, + { } + }; + static const struct coef_fw alc295_hp_speakers_coefs[] = { + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0600), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0xc0c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0008), WRITE_COEF(0x28, 0xb000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x002e), WRITE_COEF(0x28, 0x0800), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x00c1), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0320), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0039), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003b), WRITE_COEF(0x28, 0xffff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003c), WRITE_COEF(0x28, 0xffd0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0080), WRITE_COEF(0x28, 0x0880), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x0dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0018), WRITE_COEF(0x28, 0x0219), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x005d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x9142), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c0), WRITE_COEF(0x28, 0x01ce), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c1), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c2), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c3), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c4), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c5), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c6), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c7), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c8), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00c9), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ca), WRITE_COEF(0x28, 0x01c0), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cb), WRITE_COEF(0x28, 0xed0c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cc), WRITE_COEF(0x28, 0x1c00), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cd), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00ce), WRITE_COEF(0x28, 0x0200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00cf), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d0), WRITE_COEF(0x28, 0x0399), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d1), WRITE_COEF(0x28, 0x2330), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d2), WRITE_COEF(0x28, 0x1e5d), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x00d3), WRITE_COEF(0x28, 0x6eff), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0062), WRITE_COEF(0x28, 0x8000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0063), WRITE_COEF(0x28, 0x5f5f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0064), WRITE_COEF(0x28, 0x1000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0065), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0066), WRITE_COEF(0x28, 0x4004), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0067), WRITE_COEF(0x28, 0x0802), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0068), WRITE_COEF(0x28, 0x890f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0069), WRITE_COEF(0x28, 0xe021), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0070), WRITE_COEF(0x28, 0x8012), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0071), WRITE_COEF(0x28, 0x3450), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0072), WRITE_COEF(0x28, 0x0123), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0073), WRITE_COEF(0x28, 0x4543), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0074), WRITE_COEF(0x28, 0x2100), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0075), WRITE_COEF(0x28, 0x4321), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0076), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x8200), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003a), WRITE_COEF(0x28, 0x1dfe), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0051), WRITE_COEF(0x28, 0x0707), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0052), WRITE_COEF(0x28, 0x4090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0090), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x721f), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0012), WRITE_COEF(0x28, 0xebeb), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x009e), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0060), WRITE_COEF(0x28, 0x2213), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006a), WRITE_COEF(0x28, 0x0006), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x006c), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x003f), WRITE_COEF(0x28, 0x3000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0004), WRITE_COEF(0x28, 0x0500), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0040), WRITE_COEF(0x28, 0x800c), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0046), WRITE_COEF(0x28, 0xc22e), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x004b), WRITE_COEF(0x28, 0x0000), WRITE_COEF(0x29, 0xb024), + WRITE_COEF(0x24, 0x0012), WRITE_COEF(0x26, 0x0050), WRITE_COEF(0x28, 0x82ec), WRITE_COEF(0x29, 0xb024), + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + alc295_fixup_disable_dac3(codec, fix, action); + break; + case HDA_FIXUP_ACT_INIT: + alc_process_coef_fw(codec, alc295_hp_speakers_coefs); + break; + } +} diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h deleted file mode 100644 index 3b8b7d78f9e0..000000000000 --- a/sound/pci/hda/local.h +++ /dev/null @@ -1,40 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - */ - -#ifndef __HDAC_LOCAL_H -#define __HDAC_LOCAL_H - -int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); - -#define get_wcaps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) -/* get the widget type from widget capability bits */ -static inline int get_wcaps_type(unsigned int wcaps) -{ - if (!wcaps) - return -1; /* invalid type */ - return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -} - -#define get_pin_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) - -static inline -unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) -{ - unsigned int val; - - if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) - return -1; - return val; -} - -#define get_amp_caps(codec, nid, dir) \ - hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) - -#define get_power_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) - -#endif /* __HDAC_LOCAL_H */ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea8651993..292e2c592c17 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,9 @@ #include <linux/module.h> #include <linux/firmware.h> #include <linux/kernel.h> +#include <linux/types.h> +#include <linux/io.h> +#include <linux/pci.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -39,9 +42,15 @@ /* Enable this to see controls for tuning purpose. */ /*#define ENABLE_TUNING_CONTROLS*/ +#ifdef ENABLE_TUNING_CONTROLS +#include <sound/tlv.h> +#endif + #define FLOAT_ZERO 0x00000000 #define FLOAT_ONE 0x3f800000 #define FLOAT_TWO 0x40000000 +#define FLOAT_THREE 0x40400000 +#define FLOAT_EIGHT 0x41000000 #define FLOAT_MINUS_5 0xc0a00000 #define UNSOL_TAG_DSP 0x16 @@ -72,16 +81,22 @@ #define SCP_GET 1 #define EFX_FILE "ctefx.bin" +#define SBZ_EFX_FILE "ctefx-sbz.bin" +#define R3DI_EFX_FILE "ctefx-r3di.bin" #ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP MODULE_FIRMWARE(EFX_FILE); +MODULE_FIRMWARE(SBZ_EFX_FILE); +MODULE_FIRMWARE(R3DI_EFX_FILE); #endif -static char *dirstr[2] = { "Playback", "Capture" }; +static const char *const dirstr[2] = { "Playback", "Capture" }; +#define NUM_OF_OUTPUTS 3 enum { SPEAKER_OUT, - HEADPHONE_OUT + HEADPHONE_OUT, + SURROUND_OUT }; enum { @@ -89,6 +104,15 @@ enum { LINE_MIC_IN }; +/* Strings for Input Source Enum Control */ +static const char *const in_src_str[3] = {"Rear Mic", "Line", "Front Mic" }; +#define IN_SRC_NUM_OF_INPUTS 3 +enum { + REAR_MIC, + REAR_LINE_IN, + FRONT_MIC, +}; + enum { #define VNODE_START_NID 0x80 VNID_SPK = VNODE_START_NID, /* Speaker vnid */ @@ -122,13 +146,28 @@ enum { VOICEFX = IN_EFFECT_END_NID, PLAY_ENHANCEMENT, CRYSTAL_VOICE, - EFFECT_END_NID + EFFECT_END_NID, + OUTPUT_SOURCE_ENUM, + INPUT_SOURCE_ENUM, + XBASS_XOVER, + EQ_PRESET_ENUM, + SMART_VOLUME_ENUM, + MIC_BOOST_ENUM #define EFFECTS_COUNT (EFFECT_END_NID - EFFECT_START_NID) }; /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* + * Default values for the effect slider controls, they are in order of their + * effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then + * X-bass. + */ +static const unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50}; +/* Amount of effect level sliders for ca0132_alt controls. */ +#define EFFECT_LEVEL_SLIDERS 5 + /* Latency introduced by DSP blocks in milliseconds. */ #define DSP_CAPTURE_INIT_LATENCY 0 #define DSP_CRYSTAL_VOICE_LATENCY 124 @@ -150,7 +189,7 @@ struct ct_effect { #define EFX_DIR_OUT 0 #define EFX_DIR_IN 1 -static struct ct_effect ca0132_effects[EFFECTS_COUNT] = { +static const struct ct_effect ca0132_effects[EFFECTS_COUNT] = { { .name = "Surround", .nid = SURROUND, .mid = 0x96, @@ -277,7 +316,7 @@ struct ct_tuning_ctl { unsigned int def_val;/*effect default values*/ }; -static struct ct_tuning_ctl ca0132_tuning_ctls[] = { +static const struct ct_tuning_ctl ca0132_tuning_ctls[] = { { .name = "Wedge Angle", .parent_nid = VOICE_FOCUS, .nid = WEDGE_ANGLE, @@ -392,14 +431,14 @@ struct ct_voicefx_preset { unsigned int vals[VOICEFX_MAX_PARAM_COUNT]; }; -static struct ct_voicefx ca0132_voicefx = { +static const struct ct_voicefx ca0132_voicefx = { .name = "VoiceFX Capture Switch", .nid = VOICEFX, .mid = 0x95, .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18} }; -static struct ct_voicefx_preset ca0132_voicefx_presets[] = { +static const struct ct_voicefx_preset ca0132_voicefx_presets[] = { { .name = "Neutral", .vals = { 0x00000000, 0x43C80000, 0x44AF0000, 0x44FA0000, 0x3F800000, 0x3F800000, @@ -472,6 +511,161 @@ static struct ct_voicefx_preset ca0132_voicefx_presets[] = { } }; +/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */ + +#define EQ_PRESET_MAX_PARAM_COUNT 11 + +struct ct_eq { + char *name; + hda_nid_t nid; + int mid; + int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/ +}; + +struct ct_eq_preset { + char *name; /*preset name*/ + unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT]; +}; + +static const struct ct_eq ca0132_alt_eq_enum = { + .name = "FX: Equalizer Preset Switch", + .nid = EQ_PRESET_ENUM, + .mid = 0x96, + .reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20} +}; + + +static const struct ct_eq_preset ca0132_alt_eq_presets[] = { + { .name = "Flat", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + }, + { .name = "Acoustic", + .vals = { 0x00000000, 0x00000000, 0x3F8CCCCD, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Classical", + .vals = { 0x00000000, 0x00000000, 0x40C00000, + 0x40C00000, 0x40466666, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x40466666, 0x40466666 } + }, + { .name = "Country", + .vals = { 0x00000000, 0xBF99999A, 0x00000000, + 0x3FA66666, 0x3FA66666, 0x3F8CCCCD, + 0x00000000, 0x00000000, 0x40000000, + 0x40466666, 0x40800000 } + }, + { .name = "Dance", + .vals = { 0x00000000, 0xBF99999A, 0x40000000, + 0x40466666, 0x40866666, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Jazz", + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x3F8CCCCD, 0x40800000, 0x40800000, + 0x40800000, 0x00000000, 0x3F8CCCCD, + 0x40466666, 0x40466666 } + }, + { .name = "New Age", + .vals = { 0x00000000, 0x00000000, 0x40000000, + 0x40000000, 0x00000000, 0x00000000, + 0x00000000, 0x3F8CCCCD, 0x40000000, + 0x40000000, 0x40000000 } + }, + { .name = "Pop", + .vals = { 0x00000000, 0xBFCCCCCD, 0x00000000, + 0x40000000, 0x40000000, 0x00000000, + 0xBF99999A, 0xBF99999A, 0x00000000, + 0x40466666, 0x40C00000 } + }, + { .name = "Rock", + .vals = { 0x00000000, 0xBF99999A, 0xBF99999A, + 0x3F8CCCCD, 0x40000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x00000000, + 0x40800000, 0x40800000 } + }, + { .name = "Vocal", + .vals = { 0x00000000, 0xC0000000, 0xBF99999A, + 0xBF99999A, 0x00000000, 0x40466666, + 0x40800000, 0x40466666, 0x00000000, + 0x00000000, 0x3F8CCCCD } + } +}; + +/* DSP command sequences for ca0132_alt_select_out */ +#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */ +struct ca0132_alt_out_set { + char *name; /*preset name*/ + unsigned char commands; + unsigned int mids[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS]; + unsigned int vals[ALT_OUT_SET_MAX_COMMANDS]; +}; + +static const struct ca0132_alt_out_set alt_out_presets[] = { + { .name = "Line Out", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Headphone", + .commands = 7, + .mids = { 0x96, 0x96, 0x96, 0x8F, + 0x96, 0x96, 0x96 }, + .reqs = { 0x19, 0x17, 0x18, 0x01, + 0x1F, 0x15, 0x3A }, + .vals = { 0x3F000000, 0x42A00000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000 } + }, + { .name = "Surround", + .commands = 8, + .mids = { 0x96, 0x8F, 0x96, 0x96, + 0x96, 0x96, 0x96, 0x96 }, + .reqs = { 0x18, 0x01, 0x1F, 0x15, + 0x3A, 0x1A, 0x1B, 0x1C }, + .vals = { 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000, 0x00000000, + 0x00000000, 0x00000000 } + } +}; + +/* + * DSP volume setting structs. Req 1 is left volume, req 2 is right volume, + * and I don't know what the third req is, but it's always zero. I assume it's + * some sort of update or set command to tell the DSP there's new volume info. + */ +#define DSP_VOL_OUT 0 +#define DSP_VOL_IN 1 + +struct ct_dsp_volume_ctl { + hda_nid_t vnid; + int mid; /* module ID*/ + unsigned int reqs[3]; /* scp req ID */ +}; + +static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = { + { .vnid = VNID_SPK, + .mid = 0x32, + .reqs = {3, 4, 2} + }, + { .vnid = VNID_MIC, + .mid = 0x37, + .reqs = {2, 3, 1} + } +}; + enum hda_cmd_vendor_io { /* for DspIO node */ VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000, @@ -698,11 +892,12 @@ enum dsp_download_state { */ struct ca0132_spec { - struct snd_kcontrol_new *mixers[5]; + const struct snd_kcontrol_new *mixers[5]; unsigned int num_mixers; const struct hda_verb *base_init_verbs; const struct hda_verb *base_exit_verbs; const struct hda_verb *chip_init_verbs; + const struct hda_verb *sbz_init_verbs; struct hda_verb *spec_init_verbs; struct auto_pin_cfg autocfg; @@ -719,6 +914,7 @@ struct ca0132_spec { hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; hda_nid_t unsol_tag_hp; + hda_nid_t unsol_tag_front_hp; /* for desktop ca0132 codecs */ hda_nid_t unsol_tag_amic1; /* chip access */ @@ -734,6 +930,9 @@ struct ca0132_spec { unsigned int scp_resp_header; unsigned int scp_resp_data[4]; unsigned int scp_resp_count; + bool alt_firmware_present; + bool startup_check_entered; + bool dsp_reload; /* mixer and effects related */ unsigned char dmic_ctl; @@ -746,6 +945,17 @@ struct ca0132_spec { long effects_switch[EFFECTS_COUNT]; long voicefx_val; long cur_mic_boost; + /* ca0132_alt control related values */ + unsigned char in_enum_val; + unsigned char out_enum_val; + unsigned char mic_boost_enum_val; + unsigned char smart_volume_setting; + long fx_ctl_val[EFFECT_LEVEL_SLIDERS]; + long xbass_xover_freq; + long eq_preset_val; + unsigned int tlv[4]; + struct hda_vmaster_mute_hook vmaster_mute; + struct hda_codec *codec; struct delayed_work unsol_hp_work; @@ -754,6 +964,25 @@ struct ca0132_spec { #ifdef ENABLE_TUNING_CONTROLS long cur_ctl_vals[TUNING_CTLS_COUNT]; #endif + /* + * Sound Blaster Z PCI region 2 iomem, used for input and output + * switching, and other unknown commands. + */ + void __iomem *mem_base; + + /* + * Whether or not to use the alt functions like alt_select_out, + * alt_select_in, etc. Only used on desktop codecs for now, because of + * surround sound support. + */ + bool use_alt_functions; + + /* + * Whether or not to use alt controls: volume effect sliders, EQ + * presets, smart volume presets, and new control names with FX prefix. + * Renames PlayEnhancement and CrystalVoice too. + */ + bool use_alt_controls; }; /* @@ -762,6 +991,8 @@ struct ca0132_spec { enum { QUIRK_NONE, QUIRK_ALIENWARE, + QUIRK_SBZ, + QUIRK_R3DI, }; static const struct hda_pintbl alienware_pincfgs[] = { @@ -778,10 +1009,44 @@ static const struct hda_pintbl alienware_pincfgs[] = { {} }; +/* Sound Blaster Z pin configs taken from Windows Driver */ +static const struct hda_pintbl sbz_pincfgs[] = { + { 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x01c510f0 }, /* SPDIF In */ + { 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x50d000f0 }, /* N/A */ + {} +}; + +/* Recon3D integrated pin configs taken from Windows Driver */ +static const struct hda_pintbl r3di_pincfgs[] = { + { 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */ + { 0x0c, 0x014510f0 }, /* SPDIF Out 1 */ + { 0x0d, 0x014510f0 }, /* Digital Out */ + { 0x0e, 0x41c520f0 }, /* SPDIF In */ + { 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */ + { 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */ + { 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */ + { 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */ + { 0x13, 0x908700f0 }, /* What U Hear In*/ + { 0x18, 0x500000f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE), SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE), + SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ), + SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI), + SND_PCI_QUIRK(0x1458, 0xA036, "Recon3Di", QUIRK_R3DI), {} }; @@ -965,6 +1230,29 @@ exit: } /* + * Write given value to the given address through the chip I/O widget. + * not protected by the Mutex + */ +static int chipio_write_no_mutex(struct hda_codec *codec, + unsigned int chip_addx, const unsigned int data) +{ + int err; + + + /* write the address, and if successful proceed to write data */ + err = chipio_write_address(codec, chip_addx); + if (err < 0) + goto exit; + + err = chipio_write_data(codec, data); + if (err < 0) + goto exit; + +exit: + return err; +} + +/* * Write multiple values to the given address through the chip I/O widget. * protected by the Mutex */ @@ -1058,6 +1346,81 @@ static void chipio_set_control_param(struct hda_codec *codec, } /* + * Set chip parameters through the chip I/O widget. NO MUTEX. + */ +static void chipio_set_control_param_no_mutex(struct hda_codec *codec, + enum control_param_id param_id, int param_val) +{ + int val; + + if ((param_id < 32) && (param_val < 8)) { + val = (param_val << 5) | (param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_SET, val); + } else { + if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) { + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, + param_id); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, + param_val); + } + } +} +/* + * Connect stream to a source point, and then connect + * that source point to a destination point. + */ +static void chipio_set_stream_source_dest(struct hda_codec *codec, + int streamid, int source_point, int dest_point) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point); +} + +/* + * Set number of channels in the selected stream. + */ +static void chipio_set_stream_channels(struct hda_codec *codec, + int streamid, unsigned int channels) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAMS_CHANNELS, channels); +} + +/* + * Enable/Disable audio stream. + */ +static void chipio_set_stream_control(struct hda_codec *codec, + int streamid, int enable) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_ID, streamid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_STREAM_CONTROL, enable); +} + + +/* + * Set sampling rate of the connection point. NO MUTEX. + */ +static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec, + int connid, enum ca0132_sample_rate rate) +{ + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_ID, connid); + chipio_set_control_param_no_mutex(codec, + CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate); +} + +/* * Set sampling rate of the connection point. */ static void chipio_set_conn_rate(struct hda_codec *codec, @@ -1420,8 +1783,8 @@ static int dspio_send_scp_message(struct hda_codec *codec, * Returns zero or a negative error code. */ static int dspio_scp(struct hda_codec *codec, - int mod_id, int req, int dir, void *data, unsigned int len, - void *reply, unsigned int *reply_len) + int mod_id, int src_id, int req, int dir, const void *data, + unsigned int len, void *reply, unsigned int *reply_len) { int status = 0; struct scp_msg scp_send, scp_reply; @@ -1445,7 +1808,7 @@ static int dspio_scp(struct hda_codec *codec, return -EINVAL; } - scp_send.hdr = make_scp_header(mod_id, 0x20, (dir == SCP_GET), req, + scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req, 0, 0, 0, len/sizeof(unsigned int)); if (data != NULL && len > 0) { len = min((unsigned int)(sizeof(scp_send.data)), len); @@ -1502,15 +1865,24 @@ static int dspio_scp(struct hda_codec *codec, * Set DSP parameters */ static int dspio_set_param(struct hda_codec *codec, int mod_id, - int req, void *data, unsigned int len) + int src_id, int req, const void *data, unsigned int len) { - return dspio_scp(codec, mod_id, req, SCP_SET, data, len, NULL, NULL); + return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, NULL, + NULL); } static int dspio_set_uint_param(struct hda_codec *codec, int mod_id, - int req, unsigned int data) + int req, const unsigned int data) { - return dspio_set_param(codec, mod_id, req, &data, sizeof(unsigned int)); + return dspio_set_param(codec, mod_id, 0x20, req, &data, + sizeof(unsigned int)); +} + +static int dspio_set_uint_param_no_source(struct hda_codec *codec, int mod_id, + int req, const unsigned int data) +{ + return dspio_set_param(codec, mod_id, 0x00, req, &data, + sizeof(unsigned int)); } /* @@ -1522,8 +1894,9 @@ static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan) unsigned int size = sizeof(dma_chan); codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n"); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_GET, NULL, 0, dma_chan, &size); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0, + dma_chan, &size); if (status < 0) { codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n"); @@ -1552,8 +1925,9 @@ static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan) codec_dbg(codec, " dspio_free_dma_chan() -- begin\n"); codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan); - status = dspio_scp(codec, MASTERCONTROL, MASTERCONTROL_ALLOC_DMA_CHAN, - SCP_SET, &dma_chan, sizeof(dma_chan), NULL, &dummy); + status = dspio_scp(codec, MASTERCONTROL, 0x20, + MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan, + sizeof(dma_chan), NULL, &dummy); if (status < 0) { codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n"); @@ -2575,14 +2949,16 @@ exit: */ static void dspload_post_setup(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "---- dspload_post_setup ------\n"); + if (!spec->use_alt_functions) { + /*set DSP speaker to 2.0 configuration*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - /*set DSP speaker to 2.0 configuration*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080); - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000); - - /*update write pointer*/ - chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + /*update write pointer*/ + chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002); + } } /** @@ -2690,6 +3066,170 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* + * Setup GPIO for the other variants of Core3D. + */ + +/* + * Sets up the GPIO pins so that they are discoverable. If this isn't done, + * the card shows as having no GPIO pins. + */ +static void ca0132_gpio_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53); + snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00); + snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B); + break; + } + +} + +/* Sets the GPIO for audio output. */ +static void ca0132_gpio_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + switch (spec->quirk) { + case QUIRK_SBZ: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x07); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x04); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x06); + break; + case QUIRK_R3DI: + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, 0x1E); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, 0x1F); + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, 0x0C); + break; + } +} + +/* + * GPIO control functions for the Recon3D integrated. + */ + +enum r3di_gpio_bit { + /* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */ + R3DI_MIC_SELECT_BIT = 1, + /* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */ + R3DI_OUT_SELECT_BIT = 2, + /* + * I dunno what this actually does, but it stays on until the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADING = 3, + /* + * Same as above, no clue what it does, but it comes on after the dsp + * is downloaded. + */ + R3DI_GPIO_DSP_DOWNLOADED = 4 +}; + +enum r3di_mic_select { + /* Set GPIO bit 1 to 0 for rear mic */ + R3DI_REAR_MIC = 0, + /* Set GPIO bit 1 to 1 for front microphone*/ + R3DI_FRONT_MIC = 1 +}; + +enum r3di_out_select { + /* Set GPIO bit 2 to 0 for headphone */ + R3DI_HEADPHONE_OUT = 0, + /* Set GPIO bit 2 to 1 for speaker */ + R3DI_LINE_OUT = 1 +}; +enum r3di_dsp_status { + /* Set GPIO bit 3 to 1 until DSP is downloaded */ + R3DI_DSP_DOWNLOADING = 0, + /* Set GPIO bit 4 to 1 once DSP is downloaded */ + R3DI_DSP_DOWNLOADED = 1 +}; + + +static void r3di_gpio_mic_set(struct hda_codec *codec, + enum r3di_mic_select cur_mic) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_mic) { + case R3DI_REAR_MIC: + cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT); + break; + case R3DI_FRONT_MIC: + cur_gpio |= (1 << R3DI_MIC_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +static void r3di_gpio_out_set(struct hda_codec *codec, + enum r3di_out_select cur_out) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (cur_out) { + case R3DI_HEADPHONE_OUT: + cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT); + break; + case R3DI_LINE_OUT: + cur_gpio |= (1 << R3DI_OUT_SELECT_BIT); + break; + } + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +static void r3di_gpio_dsp_status_set(struct hda_codec *codec, + enum r3di_dsp_status dsp_status) +{ + unsigned int cur_gpio; + + /* Get the current GPIO Data setup */ + cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0); + + switch (dsp_status) { + case R3DI_DSP_DOWNLOADING: + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING); + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + break; + case R3DI_DSP_DOWNLOADED: + /* Set DOWNLOADING bit to 0. */ + cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING); + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); + + cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED); + break; + } + + snd_hda_codec_write(codec, codec->core.afg, 0, + AC_VERB_SET_GPIO_DATA, cur_gpio); +} + +/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2852,6 +3392,24 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, .tlv = { .c = ca0132_volume_tlv }, \ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) } +/* + * Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the + * volume put, which is used for setting the DSP volume. This was done because + * the ca0132 functions were taking too much time and causing lag. + */ +#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = snd_hda_mixer_amp_volume_info, \ + .get = snd_hda_mixer_amp_volume_get, \ + .put = ca0132_alt_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) } + #define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ @@ -2864,9 +3422,88 @@ static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, /* stereo */ #define CA0132_CODEC_VOL(xname, nid, dir) \ CA0132_CODEC_VOL_MONO(xname, nid, 3, dir) +#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \ + CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir) #define CA0132_CODEC_MUTE(xname, nid, dir) \ CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir) +/* lookup tables */ +/* + * Lookup table with decibel values for the DSP. When volume is changed in + * Windows, the DSP is also sent the dB value in floating point. In Windows, + * these values have decimal points, probably because the Windows driver + * actually uses floating point. We can't here, so I made a lookup table of + * values -90 to 9. -90 is the lowest decibel value for both the ADC's and the + * DAC's, and 9 is the maximum. + */ +static const unsigned int float_vol_db_lookup[] = { +0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000, +0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000, +0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000, +0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000, +0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000, +0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000, +0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000, +0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000, +0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000, +0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000, +0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000, +0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000, +0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000, +0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000, +0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000, +0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000, +0x40C00000, 0x40E00000, 0x41000000, 0x41100000 +}; + +/* + * This table counts from float 0 to 1 in increments of .01, which is + * useful for a few different sliders. + */ +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD, +0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE, +0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B, +0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F, +0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1, +0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333, +0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85, +0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7, +0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14, +0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D, +0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666, +0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F, +0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8, +0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1, +0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A, +0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333, +0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000 +}; + +/* + * This table counts from float 10 to 1000, which is the range of the x-bass + * crossover slider in Windows. + */ +static const unsigned int float_xbass_xover_lookup[] = { +0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000, +0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000, +0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000, +0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000, +0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000, +0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000, +0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000, +0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000, +0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000, +0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000, +0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000, +0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000, +0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000, +0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000, +0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000, +0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000, +0x44728000, 0x44750000, 0x44778000, 0x447A0000 +}; + /* The following are for tuning of products */ #ifdef ENABLE_TUNING_CONTROLS @@ -2942,7 +3579,7 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, break; snd_hda_power_up(codec); - dspio_set_param(codec, ca0132_tuning_ctls[i].mid, + dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, &(lookup[idx]), sizeof(unsigned int)); snd_hda_power_down(codec); @@ -3068,8 +3705,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); -static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0); static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, @@ -3207,7 +3844,7 @@ static int ca0132_select_out(struct hda_codec *codec) pin_ctl & ~PIN_HP); /* enable speaker node */ pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); snd_hda_set_pin_ctl(codec, spec->out_pins[0], pin_ctl | PIN_OUT); } else { @@ -3251,13 +3888,209 @@ exit: return err < 0 ? err : 0; } +/* + * This function behaves similarly to the ca0132_select_out funciton above, + * except with a few differences. It adds the ability to select the current + * output with an enumerated control "output source" if the auto detect + * mute switch is set to off. If the auto detect mute switch is enabled, it + * will detect either headphone or lineout(SPEAKER_OUT) from jack detection. + * It also adds the ability to auto-detect the front headphone port. The only + * way to select surround is to disable auto detect, and set Surround with the + * enumerated control. + */ +static int ca0132_alt_select_out(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int pin_ctl; + int jack_present; + int auto_jack; + unsigned int i; + unsigned int tmp; + int err; + /* Default Headphone is rear headphone */ + hda_nid_t headphone_nid = spec->out_pins[1]; + + codec_dbg(codec, "%s\n", __func__); + + snd_hda_power_up_pm(codec); + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + /* + * If headphone rear or front is plugged in, set to headphone. + * If neither is plugged in, set to rear line out. Only if + * hp/speaker auto detect is enabled. + */ + if (auto_jack) { + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) || + snd_hda_jack_detect(codec, spec->unsol_tag_front_hp); + + if (jack_present) + spec->cur_out_type = HEADPHONE_OUT; + else + spec->cur_out_type = SPEAKER_OUT; + } else + spec->cur_out_type = spec->out_enum_val; + + /* Begin DSP output switch */ + tmp = FLOAT_ONE; + err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp); + if (err < 0) + goto exit; + + switch (spec->cur_out_type) { + case SPEAKER_OUT: + codec_dbg(codec, "%s speaker\n", __func__); + /*speaker out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + + /* disable headphone node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* enable line-out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Enable EAPD */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + + /* If PlayEnhancement is enabled, set different source */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + break; + case HEADPHONE_OUT: + codec_dbg(codec, "%s hp\n", __func__); + /* Headphone out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0107, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x12); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x21); + r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT); + break; + } + + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + /* disable speaker*/ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl & ~PIN_HP); + + /* enable headphone, either front or rear */ + + if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp)) + headphone_nid = spec->out_pins[2]; + else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp)) + headphone_nid = spec->out_pins[1]; + + pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, headphone_nid, + pin_ctl | PIN_HP); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO); + break; + case SURROUND_OUT: + codec_dbg(codec, "%s surround\n", __func__); + /* Surround out config*/ + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0007, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0101, spec->mem_base + 0x320); + chipio_set_control_param(codec, 0x0D, 0x18); + break; + case QUIRK_R3DI: + chipio_set_control_param(codec, 0x0D, 0x24); + r3di_gpio_out_set(codec, R3DI_LINE_OUT); + break; + } + /* enable line out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[0], + pin_ctl | PIN_OUT); + /* Disable headphone out */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[1], + pin_ctl & ~PIN_HP); + /* Enable EAPD on line out */ + snd_hda_codec_write(codec, spec->out_pins[0], 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x01); + /* enable center/lfe out node */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[2], + pin_ctl | PIN_OUT); + /* Now set rear surround node as out. */ + pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + snd_hda_set_pin_ctl(codec, spec->out_pins[3], + pin_ctl | PIN_OUT); + + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE); + else + dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT); + break; + } + + /* run through the output dsp commands for line-out */ + for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) { + err = dspio_set_uint_param(codec, + alt_out_presets[spec->cur_out_type].mids[i], + alt_out_presets[spec->cur_out_type].reqs[i], + alt_out_presets[spec->cur_out_type].vals[i]); + + if (err < 0) + goto exit; + } + +exit: + snd_hda_power_down_pm(codec); + + return err < 0 ? err : 0; +} + static void ca0132_unsol_hp_delayed(struct work_struct *work) { struct ca0132_spec *spec = container_of( to_delayed_work(work), struct ca0132_spec, unsol_hp_work); struct hda_jack_tbl *jack; - ca0132_select_out(spec->codec); + if (spec->use_alt_functions) + ca0132_alt_select_out(spec->codec); + else + ca0132_select_out(spec->codec); + jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp); if (jack) { jack->block_report = 0; @@ -3268,6 +4101,10 @@ static void ca0132_unsol_hp_delayed(struct work_struct *work) static void ca0132_set_dmic(struct hda_codec *codec, int enable); static int ca0132_mic_boost_set(struct hda_codec *codec, long val); static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val); +static void resume_mic1(struct hda_codec *codec, unsigned int oldval); +static int stop_mic1(struct hda_codec *codec); +static int ca0132_cvoice_switch_set(struct hda_codec *codec); +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val); /* * Select the active VIP source @@ -3310,6 +4147,71 @@ static int ca0132_set_vipsource(struct hda_codec *codec, int val) return 1; } +static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + codec_dbg(codec, "%s\n", __func__); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* if CrystalVoice is off, vipsource should be 0 */ + if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] || + (val == 0) || spec->in_enum_val == REAR_LINE_IN) { + codec_dbg(codec, "%s: off.", __func__); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + + if (spec->in_enum_val == REAR_LINE_IN) + tmp = FLOAT_ZERO; + else { + if (spec->quirk == QUIRK_SBZ) + tmp = FLOAT_THREE; + else + tmp = FLOAT_ONE; + } + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + } else { + codec_dbg(codec, "%s: on.", __func__); + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_16_000); + + if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID]) + tmp = FLOAT_TWO; + else + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x05, tmp); + + msleep(20); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + return 1; +} + /* * Select the active microphone. * If autodetect is enabled, mic will be selected based on jack detection. @@ -3363,6 +4265,125 @@ static int ca0132_select_mic(struct hda_codec *codec) } /* + * Select the active input. + * Mic detection isn't used, because it's kind of pointless on the SBZ. + * The front mic has no jack-detection, so the only way to switch to it + * is to do it manually in alsamixer. + */ +static int ca0132_alt_select_in(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + + codec_dbg(codec, "%s\n", __func__); + + snd_hda_power_up_pm(codec); + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + spec->cur_mic_type = spec->in_enum_val; + + switch (spec->cur_mic_type) { + case REAR_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + tmp = FLOAT_THREE; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + tmp = FLOAT_ONE; + break; + default: + tmp = FLOAT_ONE; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x0000000C); + } + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + case REAR_LINE_IN: + ca0132_mic_boost_set(codec, 0); + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0000, spec->mem_base + 0x320); + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_REAR_MIC); + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x00000000); + chipio_write(codec, 0x18B09C, 0x00000000); + } + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + break; + case FRONT_MIC: + switch (spec->quirk) { + case QUIRK_SBZ: + writew(0x0100, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + tmp = FLOAT_THREE; + break; + case QUIRK_R3DI: + r3di_gpio_mic_set(codec, R3DI_FRONT_MIC); + tmp = FLOAT_ONE; + break; + default: + tmp = FLOAT_ONE; + break; + } + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + if (spec->quirk == QUIRK_R3DI) + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + if (spec->quirk == QUIRK_SBZ) { + chipio_write(codec, 0x18B098, 0x0000000C); + chipio_write(codec, 0x18B09C, 0x000000CC); + } + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + break; + } + ca0132_cvoice_switch_set(codec); + + snd_hda_power_down_pm(codec); + return 0; + +} + +/* * Check if VNODE settings take effect immediately. */ static bool ca0132_is_vnode_effective(struct hda_codec *codec, @@ -3418,7 +4439,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable) static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) { struct ca0132_spec *spec = codec->spec; - unsigned int on; + unsigned int on, tmp; int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; int err = 0; int idx = nid - EFFECT_START_NID; @@ -3442,6 +4463,46 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val) /* Voice Focus applies to 2-ch Mic, Digital Mic */ if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC)) val = 0; + + /* If Voice Focus on SBZ, set to two channel. */ + if ((nid == VOICE_FOCUS) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + + if (spec->effects_switch[VOICE_FOCUS - + EFFECT_START_NID]) { + tmp = FLOAT_TWO; + val = 1; + } else + tmp = FLOAT_ONE; + + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + } + } + /* + * For SBZ noise reduction, there's an extra command + * to module ID 0x47. No clue why. + */ + if ((nid == NOISE_REDUCTION) && (spec->quirk == QUIRK_SBZ) + && (spec->cur_mic_type != REAR_LINE_IN)) { + if (spec->effects_switch[CRYSTAL_VOICE - + EFFECT_START_NID]) { + if (spec->effects_switch[NOISE_REDUCTION - + EFFECT_START_NID]) + tmp = FLOAT_ONE; + else + tmp = FLOAT_ZERO; + } else + tmp = FLOAT_ZERO; + + dspio_set_uint_param(codec, 0x47, 0x00, tmp); + } + + /* If rear line in disable effects. */ + if (spec->use_alt_functions && + spec->in_enum_val == REAR_LINE_IN) + val = 0; } codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n", @@ -3469,6 +4530,9 @@ static int ca0132_pe_switch_set(struct hda_codec *codec) codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n", spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]); + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + i = OUT_EFFECT_START_NID - EFFECT_START_NID; nid = OUT_EFFECT_START_NID; /* PE affects all out effects */ @@ -3526,7 +4590,10 @@ static int ca0132_cvoice_switch_set(struct hda_codec *codec) /* set correct vipsource */ oldval = stop_mic1(codec); - ret |= ca0132_set_vipsource(codec, 1); + if (spec->use_alt_functions) + ret |= ca0132_alt_set_vipsource(codec, 1); + else + ret |= ca0132_set_vipsource(codec, 1); resume_mic1(codec, oldval); return ret; } @@ -3546,6 +4613,16 @@ static int ca0132_mic_boost_set(struct hda_codec *codec, long val) return ret; } +static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val) +{ + struct ca0132_spec *spec = codec->spec; + int ret = 0; + + ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0, + HDA_INPUT, 0, HDA_AMP_VOLMASK, val); + return ret; +} + static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3560,8 +4637,12 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, if (nid == VNID_HP_SEL) { auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; - if (!auto_jack) - ca0132_select_out(codec); + if (!auto_jack) { + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + else + ca0132_select_out(codec); + } return 1; } @@ -3574,7 +4655,10 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, } if (nid == VNID_HP_ASEL) { - ca0132_select_out(codec); + if (spec->use_alt_functions) + ca0132_alt_select_out(codec); + else + ca0132_select_out(codec); return 1; } @@ -3602,6 +4686,432 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol, return ret; } /* End of control change helpers. */ +/* + * Below I've added controls to mess with the effect levels, I've only enabled + * them on the Sound Blaster Z, but they would probably also work on the + * Chromebook. I figured they were probably tuned specifically for it, and left + * out for a reason. + */ + +/* Sets DSP effect level from the sliders above the controls */ +static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid, + const unsigned int *lookup, int idx) +{ + int i = 0; + unsigned int y; + /* + * For X_BASS, req 2 is actually crossover freq instead of + * effect level + */ + if (nid == X_BASS) + y = 2; + else + y = 1; + + snd_hda_power_up(codec); + if (nid == XBASS_XOVER) { + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (ca0132_effects[i].nid == X_BASS) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[1], + &(lookup[idx - 1]), sizeof(unsigned int)); + } else { + /* Find the actual effect structure */ + for (i = 0; i < OUT_EFFECTS_COUNT; i++) + if (nid == ca0132_effects[i].nid) + break; + + dspio_set_param(codec, ca0132_effects[i].mid, 0x20, + ca0132_effects[i].reqs[y], + &(lookup[idx]), sizeof(unsigned int)); + } + + snd_hda_power_down(codec); + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + long *valp = ucontrol->value.integer.value; + + *valp = spec->xbass_xover_freq; + return 0; +} + +static int ca0132_alt_slider_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx = nid - OUT_EFFECT_START_NID; + + *valp = spec->fx_ctl_val[idx]; + return 0; +} + +/* + * The X-bass crossover starts at 10hz, so the min is 1. The + * frequency is set in multiples of 10. + */ +static int ca0132_alt_xbass_xover_slider_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 = 1; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int chs = get_amp_channels(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = chs == 3 ? 2 : 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + + return 0; +} + +static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + /* any change? */ + if (spec->xbass_xover_freq == *valp) + return 0; + + spec->xbass_xover_freq = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx); + + return 0; +} + +static int ca0132_alt_effect_slider_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + long *valp = ucontrol->value.integer.value; + int idx; + + idx = nid - EFFECT_START_NID; + /* any change? */ + if (spec->fx_ctl_val[idx] == *valp) + return 0; + + spec->fx_ctl_val[idx] = *valp; + + idx = *valp; + ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx); + + return 0; +} + + +/* + * Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original + * only has off or full 30 dB, and didn't like making a volume slider that has + * traditional 0-100 in alsamixer that goes in big steps. I like enum better. + */ +#define MIC_BOOST_NUM_OF_STEPS 4 +#define MIC_BOOST_ENUM_MAX_STRLEN 10 + +static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + char *sfx = "dB"; + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS; + if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS) + uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1; + sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx); + strcpy(uinfo->value.enumerated.name, namestr); + return 0; +} + +static int ca0132_alt_mic_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->mic_boost_enum_val; + return 0; +} + +static int ca0132_alt_mic_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = MIC_BOOST_NUM_OF_STEPS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n", + sel); + + spec->mic_boost_enum_val = sel; + + if (spec->in_enum_val != REAR_LINE_IN) + ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val); + + return 1; +} + + +/* + * Input Select Control for alternative ca0132 codecs. This exists because + * front microphone has no auto-detect, and we need a way to set the rear + * as line-in + */ +static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS; + if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS) + uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1; + strcpy(uinfo->value.enumerated.name, + in_src_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_input_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->in_enum_val; + return 0; +} + +static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = IN_SRC_NUM_OF_INPUTS; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n", + sel, in_src_str[sel]); + + spec->in_enum_val = sel; + + ca0132_alt_select_in(codec); + + return 1; +} + +/* Sound Blaster Z Output Select Control */ +static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = NUM_OF_OUTPUTS; + if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS) + uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1; + strcpy(uinfo->value.enumerated.name, + alt_out_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_output_select_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->out_enum_val; + return 0; +} + +static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = NUM_OF_OUTPUTS; + unsigned int auto_jack; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n", + sel, alt_out_presets[sel].name); + + spec->out_enum_val = sel; + + auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; + + if (!auto_jack) + ca0132_alt_select_out(codec); + + return 1; +} + +/* + * Smart Volume output setting control. Three different settings, Normal, + * which takes the value from the smart volume slider. The two others, loud + * and night, disregard the slider value and have uneditable values. + */ +#define NUM_OF_SVM_SETTINGS 3 +static const char *const out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" }; + +static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS; + if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS) + uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1; + strcpy(uinfo->value.enumerated.name, + out_svm_set_enum_str[uinfo->value.enumerated.item]); + return 0; +} + +static int ca0132_alt_svm_setting_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->smart_volume_setting; + return 0; +} + +static int ca0132_alt_svm_setting_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = NUM_OF_SVM_SETTINGS; + unsigned int idx = SMART_VOLUME - EFFECT_START_NID; + unsigned int tmp; + + if (sel >= items) + return 0; + + codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n", + sel, out_svm_set_enum_str[sel]); + + spec->smart_volume_setting = sel; + + switch (sel) { + case 0: + tmp = FLOAT_ZERO; + break; + case 1: + tmp = FLOAT_ONE; + break; + case 2: + tmp = FLOAT_TWO; + break; + default: + tmp = FLOAT_ZERO; + break; + } + /* Req 2 is the Smart Volume Setting req. */ + dspio_set_uint_param(codec, ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[2], tmp); + return 1; +} + +/* Sound Blaster Z EQ preset controls */ +static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + ca0132_alt_eq_presets[uinfo->value.enumerated.item].name); + return 0; +} + +static int ca0132_alt_eq_preset_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->eq_preset_val; + return 0; +} + +static int ca0132_alt_eq_preset_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + int i, err = 0; + int sel = ucontrol->value.enumerated.item[0]; + unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets); + + if (sel >= items) + return 0; + + codec_dbg(codec, "%s: sel=%d, preset=%s\n", __func__, sel, + ca0132_alt_eq_presets[sel].name); + /* + * Idx 0 is default. + * Default needs to qualify with CrystalVoice state. + */ + for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) { + err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid, + ca0132_alt_eq_enum.reqs[i], + ca0132_alt_eq_presets[sel].vals[i]); + if (err < 0) + break; + } + + if (err >= 0) + spec->eq_preset_val = sel; + + return 1; +} static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -3753,10 +5263,15 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol, /* mic boost */ if (nid == spec->input_pins[0]) { spec->cur_mic_boost = *valp; + if (spec->use_alt_functions) { + if (spec->in_enum_val != REAR_LINE_IN) + changed = ca0132_mic_boost_set(codec, *valp); + } else { + /* Mic boost does not apply to Digital Mic */ + if (spec->cur_mic_type != DIGITAL_MIC) + changed = ca0132_mic_boost_set(codec, *valp); + } - /* Mic boost does not apply to Digital Mic */ - if (spec->cur_mic_type != DIGITAL_MIC) - changed = ca0132_mic_boost_set(codec, *valp); goto exit; } @@ -3768,6 +5283,41 @@ exit: /* * Volume related */ +/* + * Sets the internal DSP decibel level to match the DAC for output, and the + * ADC for input. Currently only the SBZ sets dsp capture volume level, and + * all alternative codecs set DSP playback volume. + */ +static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_dir; + unsigned int lookup_val; + + if (nid == VNID_SPK) + dsp_dir = DSP_VOL_OUT; + else + dsp_dir = DSP_VOL_IN; + + lookup_val = spec->vnode_lvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[0], + float_vol_db_lookup[lookup_val]); + + lookup_val = spec->vnode_rvol[nid - VNODE_START_NID]; + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[1], + float_vol_db_lookup[lookup_val]); + + dspio_set_uint_param(codec, + ca0132_alt_vol_ctls[dsp_dir].mid, + ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO); +} + static int ca0132_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -3869,6 +5419,51 @@ static int ca0132_volume_put(struct snd_kcontrol *kcontrol, return changed; } +/* + * This function is the same as the one above, because using an if statement + * inside of the above volume control for the DSP volume would cause too much + * lag. This is a lot more smooth. + */ +static int ca0132_alt_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ca0132_spec *spec = codec->spec; + hda_nid_t nid = get_amp_nid(kcontrol); + int ch = get_amp_channels(kcontrol); + long *valp = ucontrol->value.integer.value; + hda_nid_t vnid = 0; + int changed = 1; + + switch (nid) { + case 0x02: + vnid = VNID_SPK; + break; + case 0x07: + vnid = VNID_MIC; + break; + } + + /* store the left and right volume */ + if (ch & 1) { + spec->vnode_lvol[vnid - VNODE_START_NID] = *valp; + valp++; + } + if (ch & 2) { + spec->vnode_rvol[vnid - VNODE_START_NID] = *valp; + valp++; + } + + snd_hda_power_up(codec); + ca0132_alt_dsp_volume_put(codec, vnid); + mutex_lock(&codec->control_mutex); + changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); + mutex_unlock(&codec->control_mutex); + snd_hda_power_down(codec); + + return changed; +} + static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -3907,14 +5502,59 @@ static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag, return err; } +/* Add volume slider control for effect level */ +static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid, + const char *pfx, int dir) +{ + char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int type = dir ? HDA_INPUT : HDA_OUTPUT; + struct snd_kcontrol_new knew = + HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type); + + sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]); + + knew.tlv.c = 0; + knew.tlv.p = 0; + + switch (nid) { + case XBASS_XOVER: + knew.info = ca0132_alt_xbass_xover_slider_info; + knew.get = ca0132_alt_xbass_xover_slider_ctl_get; + knew.put = ca0132_alt_xbass_xover_slider_put; + break; + default: + knew.info = ca0132_alt_effect_slider_info; + knew.get = ca0132_alt_slider_ctl_get; + knew.put = ca0132_alt_effect_slider_put; + knew.private_value = + HDA_COMPOSE_AMP_VAL(nid, 1, 0, type); + break; + } + + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); +} + +/* + * Added FX: prefix for the alternative codecs, because otherwise the surround + * effect would conflict with the Surround sound volume control. Also seems more + * clear as to what the switches do. Left alone for others. + */ static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, int dir) { + struct ca0132_spec *spec = codec->spec; char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int type = dir ? HDA_INPUT : HDA_OUTPUT; struct snd_kcontrol_new knew = CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type); - sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + /* If using alt_controls, add FX: prefix. But, don't add FX: + * prefix to OutFX or InFX enable controls. + */ + if ((spec->use_alt_controls) && (nid <= IN_EFFECT_END_NID)) + sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]); + else + sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } @@ -3929,11 +5569,141 @@ static int add_voicefx(struct hda_codec *codec) return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec)); } +/* Create the EQ Preset control */ +static int add_ca0132_alt_eq_presets(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name, + EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_eq_preset_info; + knew.get = ca0132_alt_eq_preset_get; + knew.put = ca0132_alt_eq_preset_put; + return snd_hda_ctl_add(codec, EQ_PRESET_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add enumerated control for the three different settings of the smart volume + * output effect. Normal just uses the slider value, and loud and night are + * their own things that ignore that value. + */ +static int ca0132_alt_add_svm_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting", + SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_svm_setting_info; + knew.get = ca0132_alt_svm_setting_get; + knew.put = ca0132_alt_svm_setting_put; + return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM, + snd_ctl_new1(&knew, codec)); + +} + +/* + * Create an Output Select enumerated control for codecs with surround + * out capabilities. + */ +static int ca0132_alt_add_output_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Output Select", + OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT); + knew.info = ca0132_alt_output_select_get_info; + knew.get = ca0132_alt_output_select_get; + knew.put = ca0132_alt_output_select_put; + return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Create an Input Source enumerated control for the alternate ca0132 codecs + * because the front microphone has no auto-detect, and Line-in has to be set + * somehow. + */ +static int ca0132_alt_add_input_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Input Source", + INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_input_source_info; + knew.get = ca0132_alt_input_source_get; + knew.put = ca0132_alt_input_source_put; + return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM, + snd_ctl_new1(&knew, codec)); +} + +/* + * Add mic boost enumerated control. Switches through 0dB to 30dB. This adds + * more control than the original mic boost, which is either full 30dB or off. + */ +static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec) +{ + struct snd_kcontrol_new knew = + HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch", + MIC_BOOST_ENUM, 1, 0, HDA_INPUT); + knew.info = ca0132_alt_mic_boost_info; + knew.get = ca0132_alt_mic_boost_get; + knew.put = ca0132_alt_mic_boost_put; + return snd_hda_ctl_add(codec, MIC_BOOST_ENUM, + snd_ctl_new1(&knew, codec)); + +} + +/* + * Need to create slave controls for the alternate codecs that have surround + * capabilities. + */ +static const char * const ca0132_alt_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", NULL, +}; + +/* + * Also need special channel map, because the default one is incorrect. + * I think this has to do with the pin for rear surround being 0x11, + * and the center/lfe being 0x10. Usually the pin order is the opposite. + */ +const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +/* Add the correct chmap for streams with 6 channels. */ +static void ca0132_alt_add_chmap_ctls(struct hda_codec *codec) +{ + int err = 0; + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + struct hda_pcm_stream *hinfo = + &pcm->stream[SNDRV_PCM_STREAM_PLAYBACK]; + struct snd_pcm_chmap *chmap; + const struct snd_pcm_chmap_elem *elem; + + elem = ca0132_alt_chmaps; + if (hinfo->channels_max == 6) { + err = snd_pcm_add_chmap_ctls(pcm->pcm, + SNDRV_PCM_STREAM_PLAYBACK, + elem, hinfo->channels_max, 0, &chmap); + if (err < 0) + codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!"); + } + } +} + /* * When changing Node IDs for Mixer Controls below, make sure to update * Node IDs in ca0132_config() as well. */ -static struct snd_kcontrol_new ca0132_mixer[] = { +static const struct snd_kcontrol_new ca0132_mixer[] = { CA0132_CODEC_VOL("Master Playback Volume", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_MUTE("Master Playback Switch", VNID_SPK, HDA_OUTPUT), CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), @@ -3955,10 +5725,55 @@ static struct snd_kcontrol_new ca0132_mixer[] = { { } /* end */ }; +/* + * SBZ specific control mixer. Removes auto-detect for mic, and adds surround + * controls. Also sets both the Front Playback and Capture Volume controls to + * alt so they set the DSP's decibel level. + */ +static const struct snd_kcontrol_new sbz_mixer[] = { + CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), + CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT), + CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT), + CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT), + HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT), + HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT), + CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch", + VNID_HP_ASEL, 1, HDA_OUTPUT), + { } /* end */ +}; + +/* + * Same as the Sound Blaster Z, except doesn't use the alt volume for capture + * because it doesn't set decibel levels for the DSP for capture. + */ +static const struct snd_kcontrol_new r3di_mixer[] = { + CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT), + CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT), + CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT), + CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT), + HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT), + HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT), + CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch", + VNID_HP_ASEL, 1, HDA_OUTPUT), + { } /* end */ +}; + static int ca0132_build_controls(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - int i, num_fx; + int i, num_fx, num_sliders; int err = 0; /* Add Mixer controls */ @@ -3967,29 +5782,94 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + /* Setup vmaster with surround slaves for desktop ca0132 devices */ + if (spec->use_alt_functions) { + snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT, + spec->tlv); + snd_hda_add_vmaster(codec, "Master Playback Volume", + spec->tlv, ca0132_alt_slave_pfxs, + "Playback Volume"); + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, ca0132_alt_slave_pfxs, + "Playback Switch", + true, &spec->vmaster_mute.sw_kctl); + + } /* Add in and out effects controls. * VoiceFX, PE and CrystalVoice are added separately. */ num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT; for (i = 0; i < num_fx; i++) { + /* SBZ breaks if Echo Cancellation is used */ + if (spec->quirk == QUIRK_SBZ) { + if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID + + OUT_EFFECTS_COUNT)) + continue; + } + err = add_fx_switch(codec, ca0132_effects[i].nid, ca0132_effects[i].name, ca0132_effects[i].direct); if (err < 0) return err; } + /* + * If codec has use_alt_controls set to true, add effect level sliders, + * EQ presets, and Smart Volume presets. Also, change names to add FX + * prefix, and change PlayEnhancement and CrystalVoice to match. + */ + if (spec->use_alt_controls) { + ca0132_alt_add_svm_enum(codec); + add_ca0132_alt_eq_presets(codec); + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "Enable OutFX", 0); + if (err < 0) + return err; - err = add_fx_switch(codec, PLAY_ENHANCEMENT, "PlayEnhancement", 0); - if (err < 0) - return err; + err = add_fx_switch(codec, CRYSTAL_VOICE, + "Enable InFX", 1); + if (err < 0) + return err; - err = add_fx_switch(codec, CRYSTAL_VOICE, "CrystalVoice", 1); - if (err < 0) - return err; + num_sliders = OUT_EFFECTS_COUNT - 1; + for (i = 0; i < num_sliders; i++) { + err = ca0132_alt_add_effect_slider(codec, + ca0132_effects[i].nid, + ca0132_effects[i].name, + ca0132_effects[i].direct); + if (err < 0) + return err; + } + + err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER, + "X-Bass Crossover", EFX_DIR_OUT); + if (err < 0) + return err; + } else { + err = add_fx_switch(codec, PLAY_ENHANCEMENT, + "PlayEnhancement", 0); + if (err < 0) + return err; + + err = add_fx_switch(codec, CRYSTAL_VOICE, + "CrystalVoice", 1); + if (err < 0) + return err; + } add_voicefx(codec); + /* + * If the codec uses alt_functions, you need the enumerated controls + * to select the new outputs and inputs, plus add the new mic boost + * setting control. + */ + if (spec->use_alt_functions) { + ca0132_alt_add_output_enum(codec); + ca0132_alt_add_input_enum(codec); + ca0132_alt_add_mic_boost_enum(codec); + } #ifdef ENABLE_TUNING_CONTROLS add_tuning_ctls(codec); #endif @@ -4014,6 +5894,10 @@ static int ca0132_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + if (spec->use_alt_functions) + ca0132_alt_add_chmap_ctls(codec); + return 0; } @@ -4068,6 +5952,11 @@ static int ca0132_build_pcms(struct hda_codec *codec) info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); if (!info) return -ENOMEM; + if (spec->use_alt_functions) { + info->own_chmap = true; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap + = ca0132_alt_chmaps; + } info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = @@ -4076,12 +5965,16 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); - if (!info) - return -ENOMEM; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + /* With the DSP enabled, desktops don't use this ADC. */ + if (spec->use_alt_functions) { + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = + ca0132_pcm_analog_capture; + info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; + } info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); if (!info) @@ -4288,6 +6181,196 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec) } /* + * Recon3Di r3di_setup_defaults sub functions. + */ + +static void r3di_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void r3di_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + /* This ConnPointID is unique to Recon3Di. Haven't seen it elsewhere */ + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + chipio_set_conn_rate(codec, 0x0F, SR_96_000); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); +} + +/* + * Initialize Sound Blaster Z analog microphones. + */ +static void sbz_init_analog_mics(struct hda_codec *codec) +{ + unsigned int tmp; + + /* Mic 1 Setup */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + tmp = FLOAT_THREE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + /* Mic 2 Setup, even though it isn't connected on SBZ */ + chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000); + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x80, 0x01, tmp); + +} + +/* + * Sets the source of stream 0x14 to connpointID 0x48, and the destination + * connpointID to 0x91. If this isn't done, the destination is 0x71, and + * you get no sound. I'm guessing this has to do with the Sound Blaster Z + * having an updated DAC, which changes the destination to that DAC. + */ +static void sbz_connect_streams(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + + codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n"); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + + /* This value is 0x43 for 96khz, and 0x83 for 192khz. */ + chipio_write_no_mutex(codec, 0x18a020, 0x00000043); + + /* Setup stream 0x14 with it's source and destination points */ + chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91); + chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000); + chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000); + chipio_set_stream_channels(codec, 0x14, 2); + chipio_set_stream_control(codec, 0x14, 1); + + codec_dbg(codec, "Connect Streams exited, mutex released.\n"); + + mutex_unlock(&spec->chipio_mutex); + +} + +/* + * Write data through ChipIO to setup proper stream destinations. + * Not sure how it exactly works, but it seems to direct data + * to different destinations. Example is f8 to c0, e0 to c0. + * All I know is, if you don't set these, you get no sound. + */ +static void sbz_chipio_startup_data(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + mutex_lock(&spec->chipio_mutex); + codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n"); + + /* These control audio output */ + chipio_write_no_mutex(codec, 0x190060, 0x0001f8c0); + chipio_write_no_mutex(codec, 0x190064, 0x0001f9c1); + chipio_write_no_mutex(codec, 0x190068, 0x0001fac6); + chipio_write_no_mutex(codec, 0x19006c, 0x0001fbc7); + /* Signal to update I think */ + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + chipio_set_stream_channels(codec, 0x0C, 6); + chipio_set_stream_control(codec, 0x0C, 1); + /* No clue what these control */ + chipio_write_no_mutex(codec, 0x190030, 0x0001e0c0); + chipio_write_no_mutex(codec, 0x190034, 0x0001e1c1); + chipio_write_no_mutex(codec, 0x190038, 0x0001e4c2); + chipio_write_no_mutex(codec, 0x19003c, 0x0001e5c3); + chipio_write_no_mutex(codec, 0x190040, 0x0001e2c4); + chipio_write_no_mutex(codec, 0x190044, 0x0001e3c5); + chipio_write_no_mutex(codec, 0x190048, 0x0001e8c6); + chipio_write_no_mutex(codec, 0x19004c, 0x0001e9c7); + chipio_write_no_mutex(codec, 0x190050, 0x0001ecc8); + chipio_write_no_mutex(codec, 0x190054, 0x0001edc9); + chipio_write_no_mutex(codec, 0x190058, 0x0001eaca); + chipio_write_no_mutex(codec, 0x19005c, 0x0001ebcb); + + chipio_write_no_mutex(codec, 0x19042c, 0x00000001); + + codec_dbg(codec, "Startup Data exited, mutex released.\n"); + mutex_unlock(&spec->chipio_mutex); +} + +/* + * Sound Blaster Z uses these after DSP is loaded. Weird SCP commands + * without a 0x20 source like normal. + */ +static void sbz_dsp_scp_startup(struct hda_codec *codec) +{ + unsigned int tmp; + + tmp = 0x00000003; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0A, tmp); + + tmp = 0x00000001; + dspio_set_uint_param_no_source(codec, 0x80, 0x0B, tmp); + + tmp = 0x00000004; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000005; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + + tmp = 0x00000000; + dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp); + +} + +static void sbz_dsp_initial_mic_setup(struct hda_codec *codec) +{ + unsigned int tmp; + + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000); + chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000); + + tmp = FLOAT_THREE; + dspio_set_uint_param(codec, 0x80, 0x00, tmp); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + chipio_write(codec, 0x18b098, 0x0000000c); + chipio_write(codec, 0x18b09C, 0x0000000c); +} + +/* * Setup default parameters for DSP */ static void ca0132_setup_defaults(struct hda_codec *codec) @@ -4332,16 +6415,159 @@ static void ca0132_setup_defaults(struct hda_codec *codec) } /* + * Setup default parameters for Recon3Di DSP. + */ + +static void r3di_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + r3di_dsp_scp_startup(codec); + + r3di_dsp_initial_mic_setup(codec); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED); + + /* Setup effect defaults */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + +} + +/* + * Setup default parameters for the Sound Blaster Z DSP. A lot more going on + * than the Chromebook setup. + */ +static void sbz_setup_defaults(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int tmp, stream_format; + int num_fx; + int idx, i; + + if (spec->dsp_state != DSP_DOWNLOADED) + return; + + sbz_dsp_scp_startup(codec); + + sbz_init_analog_mics(codec); + + sbz_connect_streams(codec); + + sbz_chipio_startup_data(codec); + + chipio_set_stream_control(codec, 0x03, 1); + chipio_set_stream_control(codec, 0x04, 1); + + /* + * Sets internal input loopback to off, used to have a switch to + * enable input loopback, but turned out to be way too buggy. + */ + tmp = FLOAT_ONE; + dspio_set_uint_param(codec, 0x37, 0x08, tmp); + dspio_set_uint_param(codec, 0x37, 0x10, tmp); + + /*remove DSP headroom*/ + tmp = FLOAT_ZERO; + dspio_set_uint_param(codec, 0x96, 0x3C, tmp); + + /* set WUH source */ + tmp = FLOAT_TWO; + dspio_set_uint_param(codec, 0x31, 0x00, tmp); + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + + /* Set speaker source? */ + dspio_set_uint_param(codec, 0x32, 0x00, tmp); + + sbz_dsp_initial_mic_setup(codec); + + + /* out, in effects + voicefx */ + num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1; + for (idx = 0; idx < num_fx; idx++) { + for (i = 0; i <= ca0132_effects[idx].params; i++) { + dspio_set_uint_param(codec, + ca0132_effects[idx].mid, + ca0132_effects[idx].reqs[i], + ca0132_effects[idx].def_vals[i]); + } + } + + /* + * Have to make a stream to bind the sound output to, otherwise + * you'll get dead audio. Before I did this, it would bind to an + * audio input, and would never work + */ + stream_format = snd_hdac_calc_stream_format(48000, 2, + SNDRV_PCM_FORMAT_S32_LE, 32, 0); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); + + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); + + snd_hda_codec_setup_stream(codec, spec->dacs[0], spec->dsp_stream_id, + 0, stream_format); + + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); +} + +/* * Initialization of flags in chip */ static void ca0132_init_flags(struct hda_codec *codec) { - chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_COMMON_MODE, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); - chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) { + chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1); + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1); + } else { + chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_COMMON_MODE, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, + CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0); + chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1); + } } /* @@ -4349,6 +6575,16 @@ static void ca0132_init_flags(struct hda_codec *codec) */ static void ca0132_init_params(struct hda_codec *codec) { + struct ca0132_spec *spec = codec->spec; + + if (spec->use_alt_functions) { + chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000); + chipio_set_conn_rate(codec, 0x0B, SR_48_000); + chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0); + chipio_set_control_param(codec, 0, 0); + chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0); + } + chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6); chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6); } @@ -4370,11 +6606,49 @@ static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k) static bool ca0132_download_dsp_images(struct hda_codec *codec) { bool dsp_loaded = false; + struct ca0132_spec *spec = codec->spec; const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - - if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) - return false; + /* + * Alternate firmwares for different variants. The Recon3Di apparently + * can use the default firmware, but I'll leave the option in case + * it needs it again. + */ + switch (spec->quirk) { + case QUIRK_SBZ: + if (request_firmware(&fw_entry, SBZ_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "SBZ alt firmware not detected. "); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Sound Blaster Z firmware selected."); + spec->alt_firmware_present = true; + } + break; + case QUIRK_R3DI: + if (request_firmware(&fw_entry, R3DI_EFX_FILE, + codec->card->dev) != 0) { + codec_dbg(codec, "Recon3Di alt firmware not detected."); + spec->alt_firmware_present = false; + } else { + codec_dbg(codec, "Recon3Di firmware selected."); + spec->alt_firmware_present = true; + } + break; + default: + spec->alt_firmware_present = false; + break; + } + /* + * Use default ctefx.bin if no alt firmware is detected, or if none + * exists for your particular codec. + */ + if (!spec->alt_firmware_present) { + codec_dbg(codec, "Default firmware selected."); + if (request_firmware(&fw_entry, EFX_FILE, + codec->card->dev) != 0) + return false; + } dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) { @@ -4402,13 +6676,17 @@ static void ca0132_download_dsp(struct hda_codec *codec) return; /* don't retry failures */ chipio_enable_clocks(codec); - spec->dsp_state = DSP_DOWNLOADING; - if (!ca0132_download_dsp_images(codec)) - spec->dsp_state = DSP_DOWNLOAD_FAILED; - else - spec->dsp_state = DSP_DOWNLOADED; + if (spec->dsp_state != DSP_DOWNLOADED) { + spec->dsp_state = DSP_DOWNLOADING; - if (spec->dsp_state == DSP_DOWNLOADED) + if (!ca0132_download_dsp_images(codec)) + spec->dsp_state = DSP_DOWNLOAD_FAILED; + else + spec->dsp_state = DSP_DOWNLOADED; + } + + /* For codecs using alt functions, this is already done earlier */ + if (spec->dsp_state == DSP_DOWNLOADED && (!spec->use_alt_functions)) ca0132_set_dsp_msr(codec, true); } @@ -4454,6 +6732,10 @@ static void ca0132_init_unsol(struct hda_codec *codec) amic_callback); snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP, ca0132_process_dsp_response); + /* Front headphone jack detection */ + if (spec->use_alt_functions) + snd_hda_jack_detect_enable_callback(codec, + spec->unsol_tag_front_hp, hp_callback); } /* @@ -4476,7 +6758,8 @@ static struct hda_verb ca0132_base_exit_verbs[] = { {} }; -/* Other verbs tables. Sends after DSP download. */ +/* Other verbs tables. Sends after DSP download. */ + static struct hda_verb ca0132_init_verbs0[] = { /* chip init verbs */ {0x15, 0x70D, 0xF0}, @@ -4506,8 +6789,27 @@ static struct hda_verb ca0132_init_verbs0[] = { {0x15, 0x546, 0xC9}, {0x15, 0x53B, 0xCE}, {0x15, 0x5E8, 0xC9}, - {0x15, 0x717, 0x0D}, - {0x15, 0x718, 0x20}, + {} +}; + +/* Extra init verbs for SBZ */ +static struct hda_verb sbz_init_verbs[] = { + {0x15, 0x70D, 0x20}, + {0x15, 0x70E, 0x19}, + {0x15, 0x707, 0x00}, + {0x15, 0x539, 0xCE}, + {0x15, 0x546, 0xC9}, + {0x15, 0x70D, 0xB7}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x10}, + {0x15, 0x70D, 0xAF}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x01}, + {0x15, 0x707, 0x05}, + {0x15, 0x70D, 0x73}, + {0x15, 0x70E, 0x09}, + {0x15, 0x707, 0x14}, + {0x15, 0x6FF, 0xC4}, {} }; @@ -4521,7 +6823,11 @@ static void ca0132_init_chip(struct hda_codec *codec) mutex_init(&spec->chipio_mutex); spec->cur_out_type = SPEAKER_OUT; - spec->cur_mic_type = DIGITAL_MIC; + if (!spec->use_alt_functions) + spec->cur_mic_type = DIGITAL_MIC; + else + spec->cur_mic_type = REAR_MIC; + spec->cur_mic_boost = 0; for (i = 0; i < VNODES_COUNT; i++) { @@ -4539,6 +6845,15 @@ static void ca0132_init_chip(struct hda_codec *codec) on = (unsigned int)ca0132_effects[i].reqs[0]; spec->effects_switch[i] = on ? 1 : 0; } + /* + * Sets defaults for the effect slider controls, only for alternative + * ca0132 codecs. Also sets x-bass crossover frequency to 80hz. + */ + if (spec->use_alt_controls) { + spec->xbass_xover_freq = 8; + for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++) + spec->fx_ctl_val[i] = effect_slider_defaults[i]; + } spec->voicefx_val = 0; spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1; @@ -4549,6 +6864,120 @@ static void ca0132_init_chip(struct hda_codec *codec) #endif } +/* + * Recon3Di exit specific commands. + */ +/* prevents popping noise on shutdown */ +static void r3di_gpio_shutdown(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00); +} + +/* + * Sound Blaster Z exit specific commands. + */ +static void sbz_region2_exit(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int i; + + for (i = 0; i < 4; i++) + writeb(0x0, spec->mem_base + 0x100); + for (i = 0; i < 8; i++) + writeb(0xb3, spec->mem_base + 0x304); + /* + * I believe these are GPIO, with the right most hex digit being the + * gpio pin, and the second digit being on or off. We see this more in + * the input/output select functions. + */ + writew(0x0000, spec->mem_base + 0x320); + writew(0x0001, spec->mem_base + 0x320); + writew(0x0104, spec->mem_base + 0x320); + writew(0x0005, spec->mem_base + 0x320); + writew(0x0007, spec->mem_base + 0x320); +} + +static void sbz_set_pin_ctl_default(struct hda_codec *codec) +{ + hda_nid_t pins[5] = {0x0B, 0x0C, 0x0E, 0x12, 0x13}; + unsigned int i; + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40); + + for (i = 0; i < 5; i++) + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); +} + +static void sbz_clear_unsolicited(struct hda_codec *codec) +{ + hda_nid_t pins[7] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13}; + unsigned int i; + + for (i = 0; i < 7; i++) { + snd_hda_codec_write(codec, pins[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, 0x00); + } +} + +/* On shutdown, sends commands in sets of three */ +static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir, + int mask, int data) +{ + if (dir >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DIRECTION, dir); + if (mask >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_MASK, mask); + + if (data >= 0) + snd_hda_codec_write(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, data); +} + +static void sbz_exit_chip(struct hda_codec *codec) +{ + chipio_set_stream_control(codec, 0x03, 0); + chipio_set_stream_control(codec, 0x04, 0); + + /* Mess with GPIO */ + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01); + + chipio_set_stream_control(codec, 0x14, 0); + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_conn_rate(codec, 0x41, SR_192_000); + chipio_set_conn_rate(codec, 0x91, SR_192_000); + + chipio_write(codec, 0x18a020, 0x00000083); + + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07); + sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06); + + chipio_set_stream_control(codec, 0x0C, 0); + + chipio_set_control_param(codec, 0x0D, 0x24); + + sbz_clear_unsolicited(codec); + sbz_set_pin_ctl_default(codec); + + snd_hda_codec_write(codec, 0x0B, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + + if (dspload_is_loaded(codec)) + dsp_reset(codec); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x00); + + sbz_region2_exit(codec); +} + static void ca0132_exit_chip(struct hda_codec *codec) { /* put any chip cleanup stuffs here. */ @@ -4557,28 +6986,264 @@ static void ca0132_exit_chip(struct hda_codec *codec) dsp_reset(codec); } +/* + * This fixes a problem that was hard to reproduce. Very rarely, I would + * boot up, and there would be no sound, but the DSP indicated it had loaded + * properly. I did a few memory dumps to see if anything was different, and + * there were a few areas of memory uninitialized with a1a2a3a4. This function + * checks if those areas are uninitialized, and if they are, it'll attempt to + * reload the card 3 times. Usually it fixes by the second. + */ +static void sbz_dsp_startup_check(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int dsp_data_check[4]; + unsigned int cur_address = 0x390; + unsigned int i; + unsigned int failure = 0; + unsigned int reload = 3; + + if (spec->startup_check_entered) + return; + + spec->startup_check_entered = true; + + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + + codec_dbg(codec, "Startup Check: %d ", failure); + if (failure) + codec_info(codec, "DSP not initialized properly. Attempting to fix."); + /* + * While the failure condition is true, and we haven't reached our + * three reload limit, continue trying to reload the driver and + * fix the issue. + */ + while (failure && (reload != 0)) { + codec_info(codec, "Reloading... Tries left: %d", reload); + sbz_exit_chip(codec); + spec->dsp_state = DSP_DOWNLOAD_INIT; + codec->patch_ops.init(codec); + failure = 0; + for (i = 0; i < 4; i++) { + chipio_read(codec, cur_address, &dsp_data_check[i]); + cur_address += 0x4; + } + for (i = 0; i < 4; i++) { + if (dsp_data_check[i] == 0xa1a2a3a4) + failure = 1; + } + reload--; + } + + if (!failure && reload < 3) + codec_info(codec, "DSP fixed."); + + if (!failure) + return; + + codec_info(codec, "DSP failed to initialize properly. Either try a full shutdown or a suspend to clear the internal memory."); +} + +/* + * This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add + * extra precision for decibel values. If you had the dB value in floating point + * you would take the value after the decimal point, multiply by 64, and divide + * by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to + * implement fixed point or floating point dB volumes. For now, I'll set them + * to 0 just incase a value has lingered from a boot into Windows. + */ +static void ca0132_alt_vol_setup(struct hda_codec *codec) +{ + snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00); + snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00); +} + +/* + * Extra commands that don't really fit anywhere else. + */ +static void sbz_pre_dsp_setup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00820680, spec->mem_base + 0x01C); + writel(0x00820680, spec->mem_base + 0x01C); + + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfc); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfd); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xfe); + snd_hda_codec_write(codec, 0x15, 0, 0xd00, 0xff); + + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44); +} + +/* + * Extra commands that don't really fit anywhere else. + */ +static void r3di_pre_dsp_setup(struct hda_codec *codec) +{ + chipio_write(codec, 0x18b0a4, 0x000000c2); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x1E); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x1C); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x5B); + + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x20); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0x19); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x00); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_8051_DATA_WRITE, 0x40); + + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04); +} + + +/* + * These are sent before the DSP is downloaded. Not sure + * what they do, or if they're necessary. Could possibly + * be removed. Figure they're better to leave in. + */ +static void sbz_region2_startup(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + writel(0x00000000, spec->mem_base + 0x400); + writel(0x00000000, spec->mem_base + 0x408); + writel(0x00000000, spec->mem_base + 0x40C); + writel(0x00880680, spec->mem_base + 0x01C); + writel(0x00000083, spec->mem_base + 0xC0C); + writel(0x00000030, spec->mem_base + 0xC00); + writel(0x00000000, spec->mem_base + 0xC04); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x00000003, spec->mem_base + 0xC0C); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x000000F1, spec->mem_base + 0xC08); + writel(0x00000001, spec->mem_base + 0xC08); + writel(0x000000C7, spec->mem_base + 0xC08); + writel(0x000000C1, spec->mem_base + 0xC08); + writel(0x00000080, spec->mem_base + 0xC04); +} + +/* + * Extra init functions for alternative ca0132 codecs. Done + * here so they don't clutter up the main ca0132_init function + * anymore than they have to. + */ +static void ca0132_alt_init(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + ca0132_alt_vol_setup(codec); + + switch (spec->quirk) { + case QUIRK_SBZ: + codec_dbg(codec, "SBZ alt_init"); + ca0132_gpio_init(codec); + sbz_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_sequence_write(codec, spec->sbz_init_verbs); + break; + case QUIRK_R3DI: + codec_dbg(codec, "R3DI alt_init"); + ca0132_gpio_init(codec); + ca0132_gpio_setup(codec); + r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING); + r3di_pre_dsp_setup(codec); + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4); + break; + } +} + static int ca0132_init(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i; + bool dsp_loaded; + + /* + * If the DSP is already downloaded, and init has been entered again, + * there's only two reasons for it. One, the codec has awaken from a + * suspended state, and in that case dspload_is_loaded will return + * false, and the init will be ran again. The other reason it gets + * re entered is on startup for some reason it triggers a suspend and + * resume state. In this case, it will check if the DSP is downloaded, + * and not run the init function again. For codecs using alt_functions, + * it will check if the DSP is loaded properly. + */ + if (spec->dsp_state == DSP_DOWNLOADED) { + dsp_loaded = dspload_is_loaded(codec); + if (!dsp_loaded) { + spec->dsp_reload = true; + spec->dsp_state = DSP_DOWNLOAD_INIT; + } else { + if (spec->quirk == QUIRK_SBZ) + sbz_dsp_startup_check(codec); + return 0; + } + } if (spec->dsp_state != DSP_DOWNLOAD_FAILED) spec->dsp_state = DSP_DOWNLOAD_INIT; spec->curr_chip_addx = INVALID_CHIP_ADDRESS; + if (spec->quirk == QUIRK_SBZ) + sbz_region2_startup(codec); + snd_hda_power_up_pm(codec); ca0132_init_unsol(codec); - ca0132_init_params(codec); ca0132_init_flags(codec); + snd_hda_sequence_write(codec, spec->base_init_verbs); + + if (spec->quirk != QUIRK_NONE) + ca0132_alt_init(codec); + ca0132_download_dsp(codec); + ca0132_refresh_widget_caps(codec); - ca0132_setup_defaults(codec); - ca0132_init_analog_mic2(codec); - ca0132_init_dmic(codec); + + if (spec->quirk == QUIRK_SBZ) + writew(0x0107, spec->mem_base + 0x320); + + switch (spec->quirk) { + case QUIRK_R3DI: + r3di_setup_defaults(codec); + break; + case QUIRK_NONE: + case QUIRK_ALIENWARE: + ca0132_setup_defaults(codec); + ca0132_init_analog_mic2(codec); + ca0132_init_dmic(codec); + break; + } for (i = 0; i < spec->num_outputs; i++) init_output(codec, spec->out_pins[i], spec->dacs[0]); @@ -4590,14 +7255,45 @@ static int ca0132_init(struct hda_codec *codec) init_input(codec, cfg->dig_in_pin, spec->dig_in); - snd_hda_sequence_write(codec, spec->chip_init_verbs); - snd_hda_sequence_write(codec, spec->spec_init_verbs); + if (!spec->use_alt_functions) { + snd_hda_sequence_write(codec, spec->chip_init_verbs); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D); + snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, + VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20); + } - ca0132_select_out(codec); - ca0132_select_mic(codec); + if (spec->quirk == QUIRK_SBZ) + ca0132_gpio_setup(codec); + + snd_hda_sequence_write(codec, spec->spec_init_verbs); + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_setup_defaults(codec); + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + case QUIRK_R3DI: + ca0132_alt_select_out(codec); + ca0132_alt_select_in(codec); + break; + default: + ca0132_select_out(codec); + ca0132_select_mic(codec); + break; + } snd_hda_jack_report_sync(codec); + /* + * Re set the PlayEnhancement switch on a resume event, because the + * controls will not be reloaded. + */ + if (spec->dsp_reload) { + spec->dsp_reload = false; + ca0132_pe_switch_set(codec); + } + snd_hda_power_down_pm(codec); return 0; @@ -4609,19 +7305,39 @@ static void ca0132_free(struct hda_codec *codec) cancel_delayed_work_sync(&spec->unsol_hp_work); snd_hda_power_up(codec); - snd_hda_sequence_write(codec, spec->base_exit_verbs); - ca0132_exit_chip(codec); + switch (spec->quirk) { + case QUIRK_SBZ: + sbz_exit_chip(codec); + break; + case QUIRK_R3DI: + r3di_gpio_shutdown(codec); + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + break; + default: + snd_hda_sequence_write(codec, spec->base_exit_verbs); + ca0132_exit_chip(codec); + break; + } snd_hda_power_down(codec); + if (spec->mem_base) + iounmap(spec->mem_base); kfree(spec->spec_init_verbs); kfree(codec->spec); } +static void ca0132_reboot_notify(struct hda_codec *codec) +{ + codec->patch_ops.free(codec); +} + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, + .reboot_notify = ca0132_reboot_notify, }; static void ca0132_config(struct hda_codec *codec) @@ -4635,9 +7351,14 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.dac_nids = spec->dacs; spec->multiout.num_dacs = 3; - spec->multiout.max_channels = 2; - if (spec->quirk == QUIRK_ALIENWARE) { + if (!spec->use_alt_functions) + spec->multiout.max_channels = 2; + else + spec->multiout.max_channels = 6; + + switch (spec->quirk) { + case QUIRK_ALIENWARE: codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); snd_hda_apply_pincfgs(codec, alienware_pincfgs); @@ -4657,7 +7378,71 @@ static void ca0132_config(struct hda_codec *codec) spec->input_pins[2] = 0x13; spec->shared_mic_nid = 0x7; spec->unsol_tag_amic1 = 0x11; - } else { + break; + case QUIRK_SBZ: + codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__); + snd_hda_apply_pincfgs(codec, sbz_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x7; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; + case QUIRK_R3DI: + codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__); + snd_hda_apply_pincfgs(codec, r3di_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0B; /* Line out */ + spec->out_pins[1] = 0x0F; /* Rear headphone out */ + spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/ + spec->out_pins[3] = 0x11; /* Rear surround */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + spec->unsol_tag_front_hp = spec->out_pins[2]; + + spec->adcs[0] = 0x07; /* Rear Mic / Line-in */ + spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */ + spec->adcs[2] = 0x0a; /* what u hear */ + + spec->num_inputs = 2; + spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */ + spec->input_pins[1] = 0x13; /* What U Hear */ + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + break; + default: spec->num_outputs = 2; spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ @@ -4684,6 +7469,7 @@ static void ca0132_config(struct hda_codec *codec) spec->dig_in = 0x09; cfg->dig_in_pin = 0x0e; cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; + break; } } @@ -4694,6 +7480,8 @@ static int ca0132_prepare_verbs(struct hda_codec *codec) struct ca0132_spec *spec = codec->spec; spec->chip_init_verbs = ca0132_init_verbs0; + if (spec->quirk == QUIRK_SBZ) + spec->sbz_init_verbs = sbz_init_verbs; spec->spec_init_verbs = kzalloc(sizeof(struct hda_verb) * NUM_SPEC_VERBS, GFP_KERNEL); if (!spec->spec_init_verbs) return -ENOMEM; @@ -4757,9 +7545,46 @@ static int patch_ca0132(struct hda_codec *codec) else spec->quirk = QUIRK_NONE; + /* Setup BAR Region 2 for Sound Blaster Z */ + if (spec->quirk == QUIRK_SBZ) { + spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20); + if (spec->mem_base == NULL) { + codec_warn(codec, "pci_iomap failed!"); + codec_info(codec, "perhaps this is not an SBZ?"); + spec->quirk = QUIRK_NONE; + } + } + spec->dsp_state = DSP_DOWNLOAD_INIT; spec->num_mixers = 1; - spec->mixers[0] = ca0132_mixer; + + /* Set which mixers each quirk uses. */ + switch (spec->quirk) { + case QUIRK_SBZ: + spec->mixers[0] = sbz_mixer; + snd_hda_codec_set_name(codec, "Sound Blaster Z"); + break; + case QUIRK_R3DI: + spec->mixers[0] = r3di_mixer; + snd_hda_codec_set_name(codec, "Recon3Di"); + break; + default: + spec->mixers[0] = ca0132_mixer; + break; + } + + /* Setup whether or not to use alt functions/controls */ + switch (spec->quirk) { + case QUIRK_SBZ: + case QUIRK_R3DI: + spec->use_alt_controls = true; + spec->use_alt_functions = true; + break; + default: + spec->use_alt_controls = false; + spec->use_alt_functions = false; + break; + } spec->base_init_verbs = ca0132_base_init_verbs; spec->base_exit_verbs = ca0132_base_exit_verbs; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5b4dbcec6de8..dbf9910c5269 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -588,6 +588,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct conexant_spec *spec = codec->spec; + struct snd_kcontrol_new *kctl; int i; if (action != HDA_FIXUP_ACT_PROBE) @@ -606,9 +607,7 @@ static void cxt_fixup_olpc_xo(struct hda_codec *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); + snd_array_for_each(&spec->gen.kctls, i, kctl) { if (!strcmp(kctl->name, "Mic Boost Volume")) { kctl->put = olpc_xo_mic_boost_put; break; @@ -965,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), 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), @@ -998,6 +998,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" }, { .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" }, { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, + { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7d7eb1354eee..8840daf9c6a3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -510,7 +510,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) snd_info_set_text_ops(entry, per_pin, print_eld_info); entry->c.text.write = write_eld_info; - entry->mode |= S_IWUSR; + entry->mode |= 0200; per_pin->proc_entry = entry; return 0; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01a6643fc7d4..d64dcb9a4c99 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2830,6 +2830,7 @@ static int find_ext_mic_pin(struct hda_codec *codec); static void alc286_shutup(struct hda_codec *codec) { + const struct hda_pincfg *pin; int i; int mic_pin = find_ext_mic_pin(codec); /* don't shut up pins when unloading the driver; otherwise it breaks @@ -2837,8 +2838,7 @@ static void alc286_shutup(struct hda_codec *codec) */ if (codec->bus->shutdown) return; - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + snd_array_for_each(&codec->init_pins, i, pin) { /* use read here for syncing after issuing each verb */ if (pin->nid != mic_pin) snd_hda_codec_read(codec, pin->nid, 0, @@ -3653,30 +3653,37 @@ static void alc269_fixup_hp_mute_led(struct hda_codec *codec, } } -static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc269_fixup_hp_mute_led_micx(struct hda_codec *codec, + const struct hda_fixup *fix, + int action, hda_nid_t pin) { struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x18; + spec->mute_led_nid = pin; spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; spec->gen.vmaster_mute_enum = 1; codec->power_filter = led_power_filter; } } +static void alc269_fixup_hp_mute_led_mic1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x18); +} + static void alc269_fixup_hp_mute_led_mic2(struct hda_codec *codec, const struct hda_fixup *fix, int action) { - struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) { - spec->mute_led_polarity = 0; - spec->mute_led_nid = 0x19; - spec->gen.vmaster_mute.hook = alc269_fixup_mic_mute_hook; - spec->gen.vmaster_mute_enum = 1; - codec->power_filter = led_power_filter; - } + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x19); +} + +static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc269_fixup_hp_mute_led_micx(codec, fix, action, 0x1b); } /* update LED status via GPIO */ @@ -5387,6 +5394,9 @@ static void alc274_fixup_bind_dacs(struct hda_codec *codec, /* for dell wmi mic mute led */ #include "dell_wmi_helper.c" +/* for alc295_fixup_hp_top_speakers */ +#include "hp_x360_helper.c" + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5413,6 +5423,7 @@ enum { ALC269_FIXUP_HP_MUTE_LED, ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC269_FIXUP_HP_MUTE_LED_MIC2, + ALC269_FIXUP_HP_MUTE_LED_MIC3, ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_HP_GPIO_MIC1_LED, ALC269_FIXUP_HP_LINE1_MIC1_LED, @@ -5506,6 +5517,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, + ALC295_FIXUP_HP_X360, }; static const struct hda_fixup alc269_fixups[] = { @@ -5672,6 +5684,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_mute_led_mic2, }, + [ALC269_FIXUP_HP_MUTE_LED_MIC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_hp_mute_led_mic3, + }, [ALC269_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_hp_gpio_led, @@ -6375,6 +6391,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC295_FIXUP_HP_X360] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_hp_top_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_HP_MUTE_LED_MIC3 + } }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -6494,6 +6516,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), + SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -6580,7 +6603,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3138, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), - SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), @@ -6752,6 +6774,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1b, 0x01111010}, {0x1e, 0x01451130}, {0x21, 0x02211020}), + SND_HDA_PIN_QUIRK(0x10ec0235, 0x17aa, "Lenovo", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + {0x12, 0x90a60140}, + {0x14, 0x90170110}, + {0x19, 0x02a11030}, + {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 5101f40f6fbd..93b8cfc6636f 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -662,7 +662,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 8dabd4d0211d..d7366ade5a25 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -926,7 +926,7 @@ static void wm_proc_init(struct snd_ice1712 *ice) struct snd_info_entry *entry; if (!snd_card_proc_new(ice->card, "wm_codec", &entry)) { snd_info_set_text_ops(entry, ice, wm_proc_regs_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = wm_proc_regs_write; } } diff --git a/sound/pci/lola/lola_proc.c b/sound/pci/lola/lola_proc.c index c241dc06dd92..904e3c4f4dfe 100644 --- a/sound/pci/lola/lola_proc.c +++ b/sound/pci/lola/lola_proc.c @@ -214,7 +214,7 @@ void lola_proc_debug_new(struct lola *chip) snd_info_set_text_ops(entry, chip, lola_proc_codec_read); if (!snd_card_proc_new(chip->card, "codec_rw", &entry)) { snd_info_set_text_ops(entry, chip, lola_proc_codec_rw_read); - entry->mode |= S_IWUSR; + entry->mode |= 0200; entry->c.text.write = lola_proc_codec_rw_write; } if (!snd_card_proc_new(chip->card, "regs", &entry)) diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 4ca12665ff73..81af21ac1439 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -1052,10 +1052,10 @@ static int add_controls(struct oxygen *chip, [CONTROL_CD_CAPTURE_SWITCH] = "CD Capture Switch", [CONTROL_AUX_CAPTURE_SWITCH] = "Aux Capture Switch", }; - unsigned int i, j; + unsigned int i; struct snd_kcontrol_new template; struct snd_kcontrol *ctl; - int err; + int j, err; for (i = 0; i < count; ++i) { template = controls[i]; @@ -1086,11 +1086,11 @@ static int add_controls(struct oxygen *chip, err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; - for (j = 0; j < CONTROL_COUNT; ++j) - if (!strcmp(ctl->id.name, known_ctl_names[j])) { - chip->controls[j] = ctl; - ctl->private_free = oxygen_any_ctl_free; - } + j = match_string(known_ctl_names, CONTROL_COUNT, ctl->id.name); + if (j >= 0) { + chip->controls[j] = ctl; + ctl->private_free = oxygen_any_ctl_free; + } } return 0; } diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index f9ae72f28ddc..e57da4036231 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1465,7 +1465,7 @@ static void pcxhr_proc_init(struct snd_pcxhr *chip) !snd_card_proc_new(chip->card, "gpio", &entry)) { snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); entry->c.text.write = pcxhr_proc_gpo_write; - entry->mode |= S_IWUSR; + entry->mode |= 0200; } if (!snd_card_proc_new(chip->card, "ltc", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc); |