diff options
Diffstat (limited to 'sound')
141 files changed, 5521 insertions, 1339 deletions
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166acf05..8eb58c709b14 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c index 36b2d7aca1bd..5e6aed64f451 100644 --- a/sound/core/pcm_iec958.c +++ b/sound/core/pcm_iec958.c @@ -9,30 +9,18 @@ #include <linux/types.h> #include <sound/asoundef.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/pcm_iec958.h> -/** - * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status - * @runtime: pcm runtime structure with ->rate filled in - * @cs: channel status buffer, at least four bytes - * @len: length of channel status buffer - * - * Create the consumer format channel status data in @cs of maximum size - * @len corresponding to the parameters of the PCM runtime @runtime. - * - * Drivers may wish to tweak the contents of the buffer after creation. - * - * Returns: length of buffer, or negative error code if something failed. - */ -int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, - size_t len) +static int create_iec958_consumer(uint rate, uint sample_width, + u8 *cs, size_t len) { unsigned int fs, ws; if (len < 4) return -EINVAL; - switch (runtime->rate) { + switch (rate) { case 32000: fs = IEC958_AES3_CON_FS_32000; break; @@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, } if (len > 4) { - switch (snd_pcm_format_width(runtime->format)) { + switch (sample_width) { case 16: ws = IEC958_AES4_CON_WORDLEN_20_16; break; @@ -71,6 +59,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, IEC958_AES4_CON_MAX_WORDLEN_24; break; case 24: + case 32: /* Assume 24-bit width for 32-bit samples. */ ws = IEC958_AES4_CON_WORDLEN_24_20 | IEC958_AES4_CON_MAX_WORDLEN_24; break; @@ -92,4 +81,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, return len; } + +/** + * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status + * @runtime: pcm runtime structure with ->rate filled in + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, + size_t len) +{ + return create_iec958_consumer(runtime->rate, + snd_pcm_format_width(runtime->format), + cs, len); +} EXPORT_SYMBOL(snd_pcm_create_iec958_consumer); + +/** + * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status + * @hw_params: the hw_params instance for extracting rate and sample format + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params, + u8 *cs, size_t len) +{ + return create_iec958_consumer(params_rate(params), params_width(params), + cs, len); +} +EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params); diff --git a/sound/core/timer.c b/sound/core/timer.c index aa1b15c155d1..6469bedda2f3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1019,8 +1019,8 @@ static int snd_timer_s_start(struct snd_timer * timer) njiff += timer->sticks - priv->correction; priv->correction = 0; } - priv->last_expires = priv->tlist.expires = njiff; - add_timer(&priv->tlist); + priv->last_expires = njiff; + mod_timer(&priv->tlist, njiff); return 0; } @@ -1502,17 +1502,13 @@ static int snd_timer_user_ginfo(struct file *file, return err; } -static int snd_timer_user_gparams(struct file *file, - struct snd_timer_gparams __user *_gparams) +static int timer_set_gparams(struct snd_timer_gparams *gparams) { - struct snd_timer_gparams gparams; struct snd_timer *t; int err; - if (copy_from_user(&gparams, _gparams, sizeof(gparams))) - return -EFAULT; mutex_lock(®ister_mutex); - t = snd_timer_find(&gparams.tid); + t = snd_timer_find(&gparams->tid); if (!t) { err = -ENODEV; goto _error; @@ -1525,12 +1521,22 @@ static int snd_timer_user_gparams(struct file *file, err = -ENOSYS; goto _error; } - err = t->hw.set_period(t, gparams.period_num, gparams.period_den); + err = t->hw.set_period(t, gparams->period_num, gparams->period_den); _error: mutex_unlock(®ister_mutex); return err; } +static int snd_timer_user_gparams(struct file *file, + struct snd_timer_gparams __user *_gparams) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams, _gparams, sizeof(gparams))) + return -EFAULT; + return timer_set_gparams(&gparams); +} + static int snd_timer_user_gstatus(struct file *file, struct snd_timer_gstatus __user *_gstatus) { diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index 2e908225d754..6a437eb66115 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -22,6 +22,19 @@ #include <linux/compat.h> +/* + * ILP32/LP64 has different size for 'long' type. Additionally, the size + * of storage alignment differs depending on architectures. Here, '__packed' + * qualifier is used so that the size of this structure is multiple of 4 and + * it fits to any architectures with 32 bit storage alignment. + */ +struct snd_timer_gparams32 { + struct snd_timer_id tid; + u32 period_num; + u32 period_den; + unsigned char reserved[32]; +} __packed; + struct snd_timer_info32 { u32 flags; s32 card; @@ -32,6 +45,19 @@ struct snd_timer_info32 { unsigned char reserved[64]; }; +static int snd_timer_user_gparams_compat(struct file *file, + struct snd_timer_gparams32 __user *user) +{ + struct snd_timer_gparams gparams; + + if (copy_from_user(&gparams.tid, &user->tid, sizeof(gparams.tid)) || + get_user(gparams.period_num, &user->period_num) || + get_user(gparams.period_den, &user->period_den)) + return -EFAULT; + + return timer_set_gparams(&gparams); +} + static int snd_timer_user_info_compat(struct file *file, struct snd_timer_info32 __user *_info) { @@ -99,6 +125,7 @@ static int snd_timer_user_status_compat(struct file *file, */ enum { + SNDRV_TIMER_IOCTL_GPARAMS32 = _IOW('T', 0x04, struct snd_timer_gparams32), SNDRV_TIMER_IOCTL_INFO32 = _IOR('T', 0x11, struct snd_timer_info32), SNDRV_TIMER_IOCTL_STATUS32 = _IOW('T', 0x14, struct snd_timer_status32), #ifdef CONFIG_X86_X32 @@ -114,7 +141,6 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PVERSION: case SNDRV_TIMER_IOCTL_TREAD: case SNDRV_TIMER_IOCTL_GINFO: - case SNDRV_TIMER_IOCTL_GPARAMS: case SNDRV_TIMER_IOCTL_GSTATUS: case SNDRV_TIMER_IOCTL_SELECT: case SNDRV_TIMER_IOCTL_PARAMS: @@ -128,6 +154,8 @@ static long snd_timer_user_ioctl_compat(struct file *file, unsigned int cmd, uns case SNDRV_TIMER_IOCTL_PAUSE_OLD: case SNDRV_TIMER_IOCTL_NEXT_DEVICE: return snd_timer_user_ioctl(file, cmd, (unsigned long)argp); + case SNDRV_TIMER_IOCTL_GPARAMS32: + return snd_timer_user_gparams_compat(file, argp); case SNDRV_TIMER_IOCTL_INFO32: return snd_timer_user_info_compat(file, argp); case SNDRV_TIMER_IOCTL_STATUS32: diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 845d5e5884a4..ec4db3a514fc 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -446,18 +446,12 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - struct reg_params tx_params, rx_params; - - snd_dice_transaction_clear_enable(dice); + unsigned int i; - if (get_register_params(dice, &tx_params, &rx_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, &tx_params); - stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); + for (i = 0; i < MAX_STREAMS; i++) { + destroy_stream(dice, AMDTP_IN_STREAM, i); + destroy_stream(dice, AMDTP_OUT_STREAM, i); } - - release_resources(dice); - - dice->substreams_counter = 0; } void snd_dice_stream_update_duplex(struct snd_dice *dice) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..3b7ae24900fd 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, INIT_LIST_HEAD(&ebus->hlink_list); ebus->idx = idx++; + mutex_init(&ebus->lock); + ebus->cmd_dma_state = true; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 548cc1e4114b..860f8cad6602 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP); hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID); + /* since link in On, update the ref */ + hlink->ref_count = 1; + list_add_tail(&hlink->list, &ebus->hlink_list); } @@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); + +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + + mutex_lock(&ebus->lock); + + /* + * if we move from 0 to 1, count will be 1 so power up this link + * as well, also check the dma status and trigger that + */ + if (++link->ref_count == 1) { + if (!ebus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(&ebus->bus); + ebus->cmd_dma_state = true; + } + + ret = snd_hdac_ext_bus_link_power_up(link); + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); + +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + struct hdac_ext_link *hlink; + bool link_up = false; + + mutex_lock(&ebus->lock); + + /* + * if we move from 1 to 0, count will be 0 + * so power down this link as well + */ + if (--link->ref_count == 0) { + ret = snd_hdac_ext_bus_link_power_down(link); + + /* + * now check if all links are off, if so turn off + * cmd dma as well + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) { + link_up = true; + break; + } + } + + if (!link_up) { + snd_hdac_bus_stop_cmd_io(&ebus->bus); + ebus->cmd_dma_state = false; + } + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index 023cc4cad5c1..626f3bb24c55 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -104,12 +104,11 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_init_all); */ void snd_hdac_stream_free_all(struct hdac_ext_bus *ebus) { - struct hdac_stream *s; + struct hdac_stream *s, *_s; struct hdac_ext_stream *stream; struct hdac_bus *bus = ebus_to_hbus(ebus); - while (!list_empty(&bus->stream_list)) { - s = list_first_entry(&bus->stream_list, struct hdac_stream, list); + list_for_each_entry_safe(s, _s, &bus->stream_list, list) { stream = stream_to_hdac_ext_stream(s); snd_hdac_ext_stream_decouple(ebus, stream, false); list_del(&s->list); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index d1a4d6973330..03c9872c31cf 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -299,13 +299,11 @@ EXPORT_SYMBOL_GPL(_snd_hdac_read_parm); int snd_hdac_read_parm_uncached(struct hdac_device *codec, hda_nid_t nid, int parm) { - int val; + unsigned int cmd, val; - if (codec->regmap) - regcache_cache_bypass(codec->regmap, true); - val = snd_hdac_read_parm(codec, nid, parm); - if (codec->regmap) - regcache_cache_bypass(codec->regmap, false); + cmd = snd_hdac_regmap_encode_verb(nid, AC_VERB_PARAMETERS) | parm; + if (snd_hdac_regmap_read_raw_uncached(codec, cmd, &val) < 0) + return -1; return val; } EXPORT_SYMBOL_GPL(snd_hdac_read_parm_uncached); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index fb96aead8257..607bbeaebddf 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,7 @@ #include <sound/core.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_register.h> static struct i915_audio_component *hdac_acomp; @@ -97,26 +98,65 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable) } EXPORT_SYMBOL_GPL(snd_hdac_display_power); +#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ + ((pci)->device == 0x0c0c) || \ + ((pci)->device == 0x0d0c) || \ + ((pci)->device == 0x160c)) + /** - * snd_hdac_get_display_clk - Get CDCLK in kHz + * snd_hdac_i915_set_bclk - Reprogram BCLK for HSW/BDW * @bus: HDA core bus * - * This function is supposed to be used only by a HD-audio controller - * driver that needs the interaction with i915 graphics. + * Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK + * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) + * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: + * BCLK = CDCLK * M / N + * The values will be lost when the display power well is disabled and need to + * be restored to avoid abnormal playback speed. * - * This function queries CDCLK value in kHz from the graphics driver and - * returns the value. A negative code is returned in error. + * Call this function at initializing and changing power well, as well as + * at ELD notifier for the hotplug. */ -int snd_hdac_get_display_clk(struct hdac_bus *bus) +void snd_hdac_i915_set_bclk(struct hdac_bus *bus) { struct i915_audio_component *acomp = bus->audio_component; + struct pci_dev *pci = to_pci_dev(bus->dev); + int cdclk_freq; + unsigned int bclk_m, bclk_n; + + if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq) + return; /* only for i915 binding */ + if (!CONTROLLER_IN_GPU(pci)) + return; /* only HSW/BDW */ + + cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev); + switch (cdclk_freq) { + case 337500: + bclk_m = 16; + bclk_n = 225; + break; + + case 450000: + default: /* default CDCLK 450MHz */ + bclk_m = 4; + bclk_n = 75; + break; + + case 540000: + bclk_m = 4; + bclk_n = 90; + break; + + case 675000: + bclk_m = 8; + bclk_n = 225; + break; + } - if (!acomp || !acomp->ops) - return -ENODEV; - - return acomp->ops->get_cdclk_freq(acomp->dev); + snd_hdac_chip_writew(bus, HSW_EM4, bclk_m); + snd_hdac_chip_writew(bus, HSW_EM5, bclk_n); } -EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); +EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /* There is a fixed mapping between audio pin node and display port * on current Intel platforms: @@ -267,6 +307,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops } EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier); +/* check whether intel graphics is present */ +static bool i915_gfx_present(void) +{ + static struct pci_device_id ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), + .class = PCI_BASE_CLASS_DISPLAY << 16, + .class_mask = 0xff << 16 }, + {} + }; + return pci_dev_present(ids); +} + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -286,6 +338,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct i915_audio_component *acomp; int ret; + if (!i915_gfx_present()) + return -ENODEV; + acomp = kzalloc(sizeof(*acomp), GFP_KERNEL); if (!acomp) return -ENOMEM; diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index bdbcd6b75ff6..87041ddd29cb 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -453,14 +453,30 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, EXPORT_SYMBOL_GPL(snd_hdac_regmap_write_raw); static int reg_raw_read(struct hdac_device *codec, unsigned int reg, - unsigned int *val) + unsigned int *val, bool uncached) { - if (!codec->regmap) + if (uncached || !codec->regmap) return hda_reg_read(codec, reg, val); else return regmap_read(codec->regmap, reg, val); } +static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, + unsigned int reg, unsigned int *val, + bool uncached) +{ + int err; + + err = reg_raw_read(codec, reg, val, uncached); + if (err == -EAGAIN) { + err = snd_hdac_power_up_pm(codec); + if (!err) + err = reg_raw_read(codec, reg, val, uncached); + snd_hdac_power_down_pm(codec); + } + return err; +} + /** * snd_hdac_regmap_read_raw - read a pseudo register with power mgmt * @codec: the codec object @@ -472,19 +488,19 @@ static int reg_raw_read(struct hdac_device *codec, unsigned int reg, int snd_hdac_regmap_read_raw(struct hdac_device *codec, unsigned int reg, unsigned int *val) { - int err; - - err = reg_raw_read(codec, reg, val); - if (err == -EAGAIN) { - err = snd_hdac_power_up_pm(codec); - if (!err) - err = reg_raw_read(codec, reg, val); - snd_hdac_power_down_pm(codec); - } - return err; + return __snd_hdac_regmap_read_raw(codec, reg, val, false); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_read_raw); +/* Works like snd_hdac_regmap_read_raw(), but this doesn't read from the + * cache but always via hda verbs. + */ +int snd_hdac_regmap_read_raw_uncached(struct hdac_device *codec, + unsigned int reg, unsigned int *val) +{ + return __snd_hdac_regmap_read_raw(codec, reg, val, true); +} + /** * snd_hdac_regmap_update_raw - update a pseudo register with power mgmt * @codec: the codec object diff --git a/sound/hda/local.h b/sound/hda/local.h index d692f417ddc0..0d5bb159d538 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps) return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; } +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = (chans + 1) * 2; + + return chans; +} + extern const struct attribute_group *hdac_dev_attr_groups[]; int hda_widget_sysfs_init(struct hdac_device *codec); void hda_widget_sysfs_exit(struct hdac_device *codec); diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 7b248cdf06e2..fdcfa29e2205 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -591,7 +591,7 @@ static int sscape_upload_microcode(struct snd_card *card, int version) } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %zu KBs\n", init_fw->size >> 10); release_firmware(init_fw); diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89f088a..dfaf1a93fb8a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -826,7 +826,7 @@ static hda_nid_t path_power_update(struct hda_codec *codec, bool allow_powerdown) { hda_nid_t nid, changed = 0; - int i, state; + int i, state, power; for (i = 0; i < path->depth; i++) { nid = path->path[i]; @@ -838,7 +838,9 @@ static hda_nid_t path_power_update(struct hda_codec *codec, state = AC_PWRST_D0; else state = AC_PWRST_D3; - if (!snd_hda_check_power_state(codec, nid, state)) { + power = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + if (power != (state | (state << 4))) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, state); changed = nid; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2624cfe98884..9a0d1445ca5c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -857,50 +857,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) #define azx_del_card_list(chip) /* NOP */ #endif /* CONFIG_PM */ -/* Intel HSW/BDW display HDA controller is in GPU. Both its power and link BCLK - * depends on GPU. Two Extended Mode registers EM4 (M value) and EM5 (N Value) - * are used to convert CDClk (Core Display Clock) to 24MHz BCLK: - * BCLK = CDCLK * M / N - * The values will be lost when the display power well is disabled and need to - * be restored to avoid abnormal playback speed. - */ -static void haswell_set_bclk(struct hda_intel *hda) -{ - struct azx *chip = &hda->chip; - int cdclk_freq; - unsigned int bclk_m, bclk_n; - - if (!hda->need_i915_power) - return; - - cdclk_freq = snd_hdac_get_display_clk(azx_bus(chip)); - switch (cdclk_freq) { - case 337500: - bclk_m = 16; - bclk_n = 225; - break; - - case 450000: - default: /* default CDCLK 450MHz */ - bclk_m = 4; - bclk_n = 75; - break; - - case 540000: - bclk_m = 4; - bclk_n = 90; - break; - - case 675000: - bclk_m = 8; - bclk_n = 225; - break; - } - - azx_writew(chip, HSW_EM4, bclk_m); - azx_writew(chip, HSW_EM5, bclk_n); -} - #if defined(CONFIG_PM_SLEEP) || defined(SUPPORT_VGA_SWITCHEROO) /* * power management @@ -958,7 +914,7 @@ static int azx_resume(struct device *dev) if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL && hda->need_i915_power) { snd_hdac_display_power(azx_bus(chip), true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(azx_bus(chip)); } if (chip->msi) if (pci_enable_msi(pci) < 0) @@ -1058,7 +1014,7 @@ static int azx_runtime_resume(struct device *dev) bus = azx_bus(chip); if (hda->need_i915_power) { snd_hdac_display_power(bus, true); - haswell_set_bclk(hda); + snd_hdac_i915_set_bclk(bus); } else { /* toggle codec wakeup bit for STATESTS read */ snd_hdac_set_codec_wakeup(bus, true); @@ -1796,12 +1752,8 @@ static int azx_first_init(struct azx *chip) /* initialize chip */ azx_init_pci(chip); - if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { - struct hda_intel *hda; - - hda = container_of(chip, struct hda_intel, chip); - haswell_set_bclk(hda); - } + if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) + snd_hdac_i915_set_bclk(bus); hda_intel_init_chip(chip, (probe_only[dev] & 2) == 0); @@ -2232,6 +2184,9 @@ static const struct pci_device_id azx_ids[] = { /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, + /* Broxton-T */ + { PCI_DEVICE(0x8086, 0x1a98), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, @@ -2361,6 +2316,10 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, { PCI_DEVICE(0x1002, 0xaae8), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaae0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, + { PCI_DEVICE(0x1002, 0xaaf0), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, /* VIA VT8251/VT8237A */ { PCI_DEVICE(0x1106, 0x3288), .driver_data = AZX_DRIVER_VIA }, /* VIA GFX VT7122/VX900 */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 64e0d1d81ca5..9739fce9e032 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -141,14 +141,6 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_configure(codec); if (err < 0) goto error; - /* rebuild PCMs */ - err = snd_hda_codec_build_pcms(codec); - if (err < 0) - goto error; - /* rebuild mixers */ - err = snd_hda_codec_build_controls(codec); - if (err < 0) - goto error; err = snd_card_register(codec->card); error: snd_hda_power_down(codec); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index a47e8ae0eb30..80bbadc83721 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -361,6 +361,7 @@ static int cs_parse_auto_config(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; int err; + int i; err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); if (err < 0) @@ -370,6 +371,19 @@ static int cs_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + /* keep the ADCs powered up when it's dynamically switchable */ + if (spec->gen.dyn_adc_switch) { + unsigned int done = 0; + for (i = 0; i < spec->gen.input_mux.num_items; i++) { + int idx = spec->gen.dyn_adc_idx[i]; + if (done & (1 << idx)) + continue; + snd_hda_gen_fix_pin_power(codec, + spec->gen.adc_nids[idx]); + done |= 1 << idx; + } + } + return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d01834..a010d704e0e2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1396,7 +1396,6 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; - struct hdmi_eld *pin_eld = &per_pin->sink_eld; hda_nid_t pin_nid = per_pin->pin_nid; /* * Always execute a GetPinSense verb here, even when called from @@ -1413,15 +1412,15 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, present = snd_hda_pin_sense(codec, pin_nid); mutex_lock(&per_pin->lock); - pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - if (pin_eld->monitor_present) + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + if (eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); else eld->eld_valid = false; codec_dbg(codec, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer, @@ -1441,7 +1440,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, else update_eld(codec, per_pin, eld); - ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; + ret = !repoll || !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) @@ -1859,6 +1858,8 @@ static void hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + if (!per_pin) + return; mutex_lock(&per_pin->lock); per_pin->chmap_set = true; memcpy(per_pin->chmap, chmap, ARRAY_SIZE(per_pin->chmap)); @@ -2231,6 +2232,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) if (atomic_read(&(codec)->core.in_pm)) return; + snd_hdac_i915_set_bclk(&codec->bus->core); check_presence_and_report(codec, pin_nid); } @@ -3399,6 +3401,9 @@ static int patch_atihdmi(struct hda_codec *codec) spec->ops.pin_hbr_setup = atihdmi_pin_hbr_setup; spec->ops.setup_stream = atihdmi_setup_stream; + spec->chmap.ops.pin_get_slot_channel = atihdmi_pin_get_slot_channel; + spec->chmap.ops.pin_set_slot_channel = atihdmi_pin_set_slot_channel; + if (!has_amd_full_remap_support(codec)) { /* override to ATI/AMD-specific versions with pairwise mapping */ spec->chmap.ops.chmap_cea_alloc_validate_get_type = @@ -3406,10 +3411,6 @@ static int patch_atihdmi(struct hda_codec *codec) spec->chmap.ops.cea_alloc_to_tlv_chmap = atihdmi_paired_cea_alloc_to_tlv_chmap; spec->chmap.ops.chmap_validate = atihdmi_paired_chmap_validate; - spec->chmap.ops.pin_get_slot_channel = - atihdmi_pin_get_slot_channel; - spec->chmap.ops.pin_set_slot_channel = - atihdmi_pin_set_slot_channel; } /* ATI/AMD converters do not advertise all of their capabilities */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f5ca0b9ce27..4918ffa5ba68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4759,6 +4759,8 @@ enum { ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC280_FIXUP_HP_HEADSET_MIC, + ALC221_FIXUP_HP_FRONT_MIC, + ALC292_FIXUP_TPT460, }; static const struct hda_fixup alc269_fixups[] = { @@ -5401,6 +5403,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC, }, + [ALC221_FIXUP_HP_FRONT_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a19020 }, /* Front Mic */ + { } + }, + }, + [ALC292_FIXUP_TPT460] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_tpt440_dock, + .chained = true, + .chain_id = ALC293_FIXUP_LENOVO_SPK_NOISE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5434,6 +5449,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), + SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -5506,6 +5522,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 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(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5554,7 +5571,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -5567,6 +5584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), @@ -5649,6 +5667,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {.id = ALC292_FIXUP_TPT440_DOCK, .name = "tpt440-dock"}, {.id = ALC292_FIXUP_TPT440, .name = "tpt440"}, + {.id = ALC292_FIXUP_TPT460, .name = "tpt460"}, {} }; #define ALC225_STANDARD_PINS \ @@ -6406,6 +6425,8 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6646,6 +6667,21 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC662_FIXUP_ASUS_Nx50] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_auto_mute_via_amp, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_1A + }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6668,10 +6704,14 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), + SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), + SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index c5194f5b150a..d7e71f309299 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1341,5 +1341,6 @@ irqreturn_t pcxhr_threaded_irq(int irq, void *dev_id) } pcxhr_msg_thread(mgr); + mutex_unlock(&mgr->lock); return IRQ_HANDLED; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 276897033639..1267e1af0fae 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 5741c0aa6c03..b5d1caa04d8e 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, stype = substream->stream; pcd = to_dmadata(substream); - DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " - "runtime->min_align %d\n", + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", (unsigned long)runtime->dma_area, (unsigned long)runtime->dma_addr, runtime->dma_bytes, runtime->min_align); diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 1c1f2210387b..6ba20498202e 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: data_length = 16; break; + case SNDRV_PCM_FORMAT_S24_LE: + data_length = 24; + break; case SNDRV_PCM_FORMAT_S32_LE: data_length = 32; break; @@ -273,13 +276,20 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* otherwise calculate a fitting block ratio */ bclk_ratio = 2 * data_length; - /* set target clock rate*/ - clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + /* Clock should only be set up here if CPU is clock master */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + break; + default: + break; + } /* Setup the frame format */ format = BCM2835_I2S_CHEN; - if (data_length > 24) + if (data_length >= 24) format |= BCM2835_I2S_CHWEX; format |= BCM2835_I2S_CHWID((data_length-8)&0xf); @@ -570,6 +580,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .capture = { @@ -577,6 +588,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .ops = &bcm2835_i2s_dai_ops, @@ -678,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + /* + * Set the PACK flag to enable S16_LE support (2 S16_LE values + * packed into 32-bit transfers). + */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + /* BCLK ratio - use default */ dev->bclk_ratio = 0; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..03242a4dc5a8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -88,12 +89,14 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_NAU8825 if I2C + select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM179X_I2C if I2C select SND_SOC_PCM179X_SPI if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER + select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C @@ -124,12 +127,14 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C + select SND_SOC_TAS5720 if I2C select SND_SOC_TFA9879 if I2C select SND_SOC_TLV320AIC23_I2C if I2C select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4 if I2C + select SND_SOC_TLV320AIC32X4_I2C if I2C + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -477,6 +482,11 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" @@ -522,6 +532,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate @@ -575,6 +588,9 @@ config SND_SOC_PCM3168A_SPI select SND_SOC_PCM3168A select REGMAP_SPI +config SND_SOC_PCM5102A + tristate + config SND_SOC_PCM512x tristate @@ -629,6 +645,7 @@ config SND_SOC_RT5514 config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" + depends on I2C config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" @@ -737,8 +754,15 @@ config SND_SOC_TAS5086 depends on I2C config SND_SOC_TAS571X - tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers" + tristate "Texas Instruments TAS5711/TAS5717/TAS5719/TAS5721 power amplifiers" + depends on I2C + +config SND_SOC_TAS5720 + tristate "Texas Instruments TAS5720 Mono Audio amplifier" depends on I2C + help + Enable support for Texas Instruments TAS5720L/M high-efficiency mono + Class-D audio power amplifiers. config SND_SOC_TFA9879 tristate "NXP Semiconductors TFA9879 amplifier" @@ -769,6 +793,16 @@ config SND_SOC_TLV320AIC31XX config SND_SOC_TLV320AIC32X4 tristate +config SND_SOC_TLV320AIC32X4_I2C + tristate + depends on I2C + select SND_SOC_TLV320AIC32X4 + +config SND_SOC_TLV320AIC32X4_SPI + tristate + depends on SPI_MASTER + select SND_SOC_TLV320AIC32X4 + config SND_SOC_TLV320AIC3X tristate "Texas Instruments TLV320AIC3x CODECs" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..2620ee001071 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o @@ -81,6 +82,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o +snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm179x-codec-objs := pcm179x.o snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o @@ -89,6 +91,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o +snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o @@ -129,6 +132,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas5720-objs := tas5720.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -136,6 +140,8 @@ snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic32x4-i2c-objs := tlv320aic32x4-i2c.o +snd-soc-tlv320aic32x4-spi-objs := tlv320aic32x4-spi.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-ts3a227e-objs := ts3a227e.o @@ -290,6 +296,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o @@ -298,6 +305,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o +obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o @@ -335,6 +343,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o @@ -342,6 +351,8 @@ obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_I2C) += snd-soc-tlv320aic32x4-i2c.o +obj-$(CONFIG_SND_SOC_TLV320AIC32X4_SPI) += snd-soc-tlv320aic32x4-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TS3A227E) += snd-soc-ts3a227e.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c22812a..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { @@ -608,9 +611,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 92d22a018d68..664a8c044ffb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); @@ -249,6 +251,18 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +int arizona_free_spk(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT_WARN, arizona); + arizona_free_irq(arizona, ARIZONA_IRQ_SPK_OVERHEAT, arizona); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_free_spk); + static const struct snd_soc_dapm_route arizona_mono_routes[] = { { "OUT1R", NULL, "OUT1L" }, { "OUT2R", NULL, "OUT2L" }, @@ -1122,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int mask = 0x3 << w->shift; unsigned int val; switch (event) { @@ -1136,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, return 0; } - snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val); + snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val); return 0; } @@ -2035,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll, init_ratio, Fref, refdiv); while (div <= ARIZONA_FLL_MAX_REFDIV) { - for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + /* start from init_ratio because this may already give a + * fractional N.K + */ + for (ratio = init_ratio; ratio > 0; ratio--) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); + return ratio; + } + } + + for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / (fll->vco_mult * ratio) < Fref) { @@ -2061,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (target % (ratio * Fref)) { - cfg->refdiv = refdiv; - cfg->fratio = ratio - 1; - arizona_fll_dbg(fll, - "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", - Fref, refdiv, div, ratio); - return ratio; - } - } - div *= 2; Fref /= 2; refdiv++; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 1ea8e4ecf8d4..ce0531b8c632 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -307,6 +307,8 @@ extern int arizona_init_spk(struct snd_soc_codec *codec); extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_mono(struct snd_soc_codec *codec); +extern int arizona_free_spk(struct snd_soc_codec *codec); + extern int arizona_init_dai(struct arizona_priv *priv, int dai); int arizona_set_output_mode(struct snd_soc_codec *codec, int output, diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..287d13740be4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,7 +274,9 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: @@ -282,13 +284,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BOOST_MGR_FIXED: pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +300,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +316,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: @@ -326,6 +334,7 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_RECOV_3_6V: pdata->batt_recov = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-recovery DT value %d\n", val); diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cd5f769bb61..eec1ff853b98 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -56,7 +56,7 @@ struct cs42l56_private { u8 iface; u8 iface_fmt; u8 iface_inv; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 576087bda330..5ec5a682d186 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "IN2L PGA", NULL, "IN2L" }, { "IN2R PGA", NULL, "IN2R" }, + { "Audio Trace DSP", NULL, "DSP2" }, + { "Audio Trace DSP", NULL, "SYSCLK" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), @@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = { .formats = CS47L24_FORMATS, }, }, + { + .name = "cs47l24-cpu-trace", + .capture = { + .stream_name = "Audio Trace CPU", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l24-dsp-trace", + .capture = { + .stream_name = "Audio Trace DSP", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + }, }; static int cs47l24_open(struct snd_compr_stream *stream) @@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream) if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { n_adsp = 2; + } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) { + n_adsp = 1; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", @@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) { struct cs47l24_priv *priv = data; struct arizona *arizona = priv->core.arizona; - int ret; + int serviced = 0; + int i, ret; + + for (i = 1; i <= 2; ++i) { + ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]); + if (ret != -ENODEV) + serviced++; + } - ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]); - if (ret == -ENODEV) { + if (!serviced) { dev_err(arizona->dev, "Spurious compressed data IRQ\n"); return IRQ_NONE; } @@ -1108,6 +1140,9 @@ static int cs47l24_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + + arizona_free_spk(codec); + return 0; } @@ -1157,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = { static struct snd_soc_platform_driver cs47l24_compr_platform = { .compr_ops = &cs47l24_compr_ops, }; + static int cs47l24_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); @@ -1225,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); return ret; } + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); - if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); snd_soc_unregister_platform(&pdev->dev); @@ -1238,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev) static int cs47l24_remove(struct platform_device *pdev) { + struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&cs47l24->core.adsp[1]); + wm_adsp2_remove(&cs47l24->core.adsp[2]); + return 0; } diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 7278f93460c1..e5527bc570ae 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -726,6 +726,68 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = { /* + * DAPM Events + */ + +static int da7213_dai_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable DAI clks for master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, + DA7213_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL); + if (!(pll_ctrl & DA7213_PLL_SRM_EN)) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7213_PLL_STATUS); + if (pll_status & DA7219_PLL_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* * DAPM widgets */ @@ -736,7 +798,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = { /* Use a supply here as this controls both input & output DAIs */ SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT, - DA7213_NO_INVERT, NULL, 0), + DA7213_NO_INVERT, da7213_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* * Input @@ -1143,11 +1206,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Set master/slave mode */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE; da7213->master = true; break; case SND_SOC_DAIFMT_CBS_CFS: - dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE; da7213->master = false; break; default: @@ -1281,28 +1342,28 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_ctrl = 0; /* Workout input divider based on MCLK rate */ - if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { + if (da7213->mclk_rate == 32768) { /* 32KHz PLL Mode */ - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; freq_ref = 3750000; pll_ctrl |= DA7213_PLL_32K_MODE; } else { /* 5 - 54MHz MCLK */ if (da7213->mclk_rate < 5000000) { goto pll_err; - } else if (da7213->mclk_rate <= 10000000) { - indiv_bits = DA7213_PLL_INDIV_5_10_MHZ; - indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7213->mclk_rate <= 20000000) { - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7213->mclk_rate <= 40000000) { - indiv_bits = DA7213_PLL_INDIV_20_40_MHZ; - indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7213->mclk_rate <= 9000000) { + indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ; + indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL; + } else if (da7213->mclk_rate <= 18000000) { + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7213->mclk_rate <= 36000000) { + indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7213->mclk_rate <= 54000000) { - indiv_bits = DA7213_PLL_INDIV_40_54_MHZ; - indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL; } else { goto pll_err; } @@ -1547,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec) /* Default to using SRM for slave mode */ da7213->srm_en = true; + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA7213_AUX_L_CTRL, DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 030fd691b076..fbb7a356a501 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -142,6 +142,9 @@ * Bit fields */ +/* DA7213_PLL_STATUS = 0x03 */ +#define DA7219_PLL_SRM_LOCK (0x1 << 1) + /* DA7213_SR = 0x22 */ #define DA7213_SR_8000 (0x1 << 0) #define DA7213_SR_11025 (0x2 << 0) @@ -160,10 +163,10 @@ #define DA7213_VMID_EN (0x1 << 7) /* DA7213_PLL_CTRL = 0x27 */ -#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2) -#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2) -#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2) -#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2) +#define DA7213_PLL_INDIV_5_TO_9_MHZ (0x0 << 2) +#define DA7213_PLL_INDIV_9_TO_18_MHZ (0x1 << 2) +#define DA7213_PLL_INDIV_18_TO_36_MHZ (0x2 << 2) +#define DA7213_PLL_INDIV_36_TO_54_MHZ (0x3 << 2) #define DA7213_PLL_INDIV_MASK (0x3 << 2) #define DA7213_PLL_MCLK_SQR_EN (0x1 << 4) #define DA7213_PLL_32K_MODE (0x1 << 5) @@ -178,8 +181,6 @@ #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) -#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7) -#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ @@ -412,6 +413,9 @@ #define DA7213_DMIC_CLK_RATE_SHIFT 2 #define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2) +/* DA7213_PC_COUNT = 0x94 */ +#define DA7213_PC_FREERUN_MASK (0x1 << 0) + /* DA7213_DIG_CTRL = 0x99 */ #define DA7213_DAC_L_INV_SHIFT 3 #define DA7213_DAC_R_INV_SHIFT 7 @@ -495,15 +499,16 @@ #define DA7213_ALC_AVG_ITERATIONS 5 /* PLL related */ -#define DA7213_SYSCLK_MCLK 0 -#define DA7213_SYSCLK_PLL 1 -#define DA7213_PLL_FREQ_OUT_90316800 90316800 -#define DA7213_PLL_FREQ_OUT_98304000 98304000 -#define DA7213_PLL_FREQ_OUT_94310400 94310400 -#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7213_SYSCLK_MCLK 0 +#define DA7213_SYSCLK_PLL 1 +#define DA7213_PLL_FREQ_OUT_90316800 90316800 +#define DA7213_PLL_FREQ_OUT_98304000 98304000 +#define DA7213_PLL_FREQ_OUT_94310400 94310400 +#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL 2 +#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL 16 +#define DA7213_SRM_CHECK_RETRIES 8 enum da7213_clk_src { DA7213_CLKSRC_MCLK = 0, diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 93575f251866..99ce23e113bf 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ if (da7218->mclk_rate == 32768) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; } else if (da7218->mclk_rate < 2000000) { dev_err(codec->dev, "PLL input clock %d below valid range\n", da7218->mclk_rate); return -EINVAL; - } else if (da7218->mclk_rate <= 5000000) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 10000000) { - indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 20000000) { - indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; - indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7218->mclk_rate <= 40000000) { - indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; - indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 4500000) { + indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7218->mclk_rate <= 9000000) { + indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7218->mclk_rate <= 18000000) { + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7218->mclk_rate <= 36000000) { + indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7218->mclk_rate <= 54000000) { - indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; - indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7218->mclk_rate); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h index c2c59049a2ad..477cd37723cf 100644 --- a/sound/soc/codecs/da7218.h +++ b/sound/soc/codecs/da7218.h @@ -876,15 +876,11 @@ /* DA7218_PLL_CTRL = 0x91 */ #define DA7218_PLL_INDIV_SHIFT 0 #define DA7218_PLL_INDIV_MASK (0x7 << 0) -#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) -#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) -#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) -#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) -#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) -#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 -#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0) #define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 #define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) #define DA7218_PLL_MODE_SHIFT 6 @@ -1336,6 +1332,13 @@ #define DA7218_PLL_FREQ_OUT_90316 90316800 #define DA7218_PLL_FREQ_OUT_98304 98304000 +/* PLL Frequency Dividers */ +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16 + /* ALC Calibration */ #define DA7218_ALC_CALIB_DELAY_MIN 2500 #define DA7218_ALC_CALIB_DELAY_MAX 5000 diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 81c0708b85c1..5c93899f1f0e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -11,6 +11,7 @@ * option) any later version. */ +#include <linux/acpi.h> #include <linux/clk.h> #include <linux/i2c.h> #include <linux/of_device.h> @@ -1025,7 +1026,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) return 0; - if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + if ((freq < 2000000) || (freq > 54000000)) { dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; @@ -1079,21 +1080,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, dev_err(codec->dev, "PLL input clock %d below valid range\n", da7219->mclk_rate); return -EINVAL; - } else if (da7219->mclk_rate <= 5000000) { - indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; - indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; - } else if (da7219->mclk_rate <= 10000000) { - indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; - indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7219->mclk_rate <= 20000000) { - indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; - indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7219->mclk_rate <= 40000000) { - indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; - indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 4500000) { + indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 9000000) { + indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7219->mclk_rate <= 18000000) { + indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7219->mclk_rate <= 36000000) { + indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7219->mclk_rate <= 54000000) { - indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; - indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7219->mclk_rate); @@ -1426,6 +1427,12 @@ static const struct of_device_id da7219_of_match[] = { }; MODULE_DEVICE_TABLE(of, da7219_of_match); +static const struct acpi_device_id da7219_acpi_match[] = { + { .id = "DLGS7219", }, + { } +}; +MODULE_DEVICE_TABLE(acpi, da7219_acpi_match); + static enum da7219_micbias_voltage da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) { @@ -1955,6 +1962,7 @@ static struct i2c_driver da7219_i2c_driver = { .driver = { .name = "da7219", .of_match_table = of_match_ptr(da7219_of_match), + .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe = da7219_i2c_probe, .remove = da7219_i2c_remove, diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 5a787e738084..ff2a2f02ce40 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -194,11 +194,11 @@ /* DA7219_PLL_CTRL = 0x20 */ #define DA7219_PLL_INDIV_SHIFT 2 #define DA7219_PLL_INDIV_MASK (0x7 << 2) -#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) -#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) -#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) -#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) -#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_9_TO_18_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_18_TO_36_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_36_TO_54_MHZ (0x4 << 2) #define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 #define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) #define DA7219_PLL_MODE_SHIFT 6 @@ -761,11 +761,11 @@ #define DA7219_PLL_FREQ_OUT_98304 98304000 /* PLL Frequency Dividers */ -#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 -#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16 /* SRM */ #define DA7219_SRM_CHECK_RETRIES 8 diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index afa6c5db9dcc..2086d7107622 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -26,18 +26,30 @@ #include <sound/tlv.h> #include "es8328.h" -#define ES8328_SYSCLK_RATE_1X 11289600 -#define ES8328_SYSCLK_RATE_2X 22579200 +static const unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; -/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ -static struct { - int rate; - u8 ratio; -} mclk_ratios[] = { - { 8000, 9 }, - {11025, 7 }, - {22050, 4 }, - {44100, 2 }, +static const int ratios_12288[] = { + 10, 7, 6, 4, 3, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static const unsigned int rates_11289[] = { + 8018, 11025, 22050, 44100, 88200, +}; + +static const int ratios_11289[] = { + 9, 7, 4, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_11289 = { + .count = ARRAY_SIZE(rates_11289), + .list = rates_11289, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = { "HPVDD", }; -#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ +#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_11025) -#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_8000) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) struct es8328_priv { struct regmap *regmap; struct clk *clk; int playback_fs; bool deemph; + int mclkdiv2; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; + const int *mclk_ratios; struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; }; @@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute) mute ? ES8328_DACCONTROL3_DACMUTE : 0); } +static int es8328_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->sysclk_constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + es8328->sysclk_constraints); + + return 0; +} + static int es8328_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; int i; int reg; - u8 ratio; + int wl; + int ratio; + + if (!es8328->sysclk_constraints) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = ES8328_DACCONTROL2; else reg = ES8328_ADCCONTROL5; - clk_rate = clk_get_rate(es8328->clk); + for (i = 0; i < es8328->sysclk_constraints->count; i++) + if (es8328->sysclk_constraints->list[i] == params_rate(params)) + break; - if ((clk_rate != ES8328_SYSCLK_RATE_1X) && - (clk_rate != ES8328_SYSCLK_RATE_2X)) { - dev_err(codec->dev, - "%s: clock is running at %d Hz, not %d or %d Hz\n", - __func__, clk_rate, - ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + if (i == es8328->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with current clock\n", + params_rate(params)); return -EINVAL; } - /* find master mode MCLK to sampling frequency ratio */ - ratio = mclk_ratios[0].rate; - for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) - if (params_rate(params) <= mclk_ratios[i].rate) - ratio = mclk_ratios[i].ratio; + ratio = es8328->mclk_ratios[i]; + snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2, + es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0); + + switch (params_width(params)) { + case 16: + wl = 3; + break; + case 18: + wl = 2; + break; + case 20: + wl = 1; + break; + case 24: + wl = 0; + break; + case 32: + wl = 4; + break; + default: + return -EINVAL; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACWL_MASK, + wl << ES8328_DACCONTROL1_DACWL_SHIFT); + es8328->playback_fs = params_rate(params); es8328_set_deemph(codec); - } + } else + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCWL_MASK, + wl << ES8328_ADCCONTROL4_ADCWL_SHIFT); return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); } +static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int mclkdiv2 = 0; + + switch (freq) { + case 0: + es8328->sysclk_constraints = NULL; + es8328->mclk_ratios = NULL; + break; + case 22579200: + mclkdiv2 = 1; + /* fallthru */ + case 11289600: + es8328->sysclk_constraints = &constraints_11289; + es8328->mclk_ratios = ratios_11289; + break; + case 24576000: + mclkdiv2 = 1; + /* fallthru */ + case 12288000: + es8328->sysclk_constraints = &constraints_12288; + es8328->mclk_ratios = ratios_12288; + break; + default: + return -EINVAL; + } + + es8328->mclkdiv2 = mclkdiv2; + return 0; +} + static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; - u8 mode = ES8328_DACCONTROL1_DACWL_16; + u8 dac_mode = 0; + u8 adc_mode = 0; /* set master/slave audio interface */ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) @@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S; break; case SND_SOC_DAIFMT_RIGHT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST; break; case SND_SOC_DAIFMT_LEFT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST; break; default: return -EINVAL; @@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) return -EINVAL; - snd_soc_write(codec, ES8328_DACCONTROL1, mode); - snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode); + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode); /* Master serial port mode, with BCLK generated automatically */ - clk_rate = clk_get_rate(es8328->clk); - if (clk_rate == ES8328_SYSCLK_RATE_1X) - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MSC); - else - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MCLKDIV2 | - ES8328_MASTERMODE_MSC); + snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC); return 0; } @@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec, } static const struct snd_soc_dai_ops es8328_dai_ops = { + .startup = es8328_startup, .hw_params = es8328_hw_params, .digital_mute = es8328_mute, + .set_sysclk = es8328_set_sysclk, .set_fmt = es8328_set_dai_fmt, }; @@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = { .formats = ES8328_FORMATS, }, .ops = &es8328_dai_ops, + .symmetric_rates = 1, }; static int es8328_suspend(struct snd_soc_codec *codec) @@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = { .val_bits = 8, .max_register = ES8328_REG_MAX, .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, }; EXPORT_SYMBOL_GPL(es8328_regmap_config); diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 156c748c89c7..1a736e72a929 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) -#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0) #define ES8328_CONTROL1_ENREF (1 << 2) #define ES8328_CONTROL1_SEQEN (1 << 3) #define ES8328_CONTROL1_SAMEFS (1 << 4) @@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL1 0x09 #define ES8328_ADCCONTROL2 0x0a #define ES8328_ADCCONTROL3 0x0b + #define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0) +#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2 +#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5) + #define ES8328_ADCCONTROL5 0x0d #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) @@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL14 0x16 #define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1) #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) -#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) -#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) -#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) -#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) -#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACWL_SHIFT 3 +#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 26f9459cb3bc..181cd3bf0b92 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -29,6 +29,7 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_i915.h> #include <sound/pcm_drm_eld.h> +#include <sound/hda_chmap.h> #include "../../hda/local.h" #include "hdac_hdmi.h" @@ -60,11 +61,17 @@ struct hdac_hdmi_cvt { struct hdac_hdmi_cvt_params params; }; +/* Currently only spk_alloc, more to be added */ +struct hdac_hdmi_parsed_eld { + u8 spk_alloc; +}; + struct hdac_hdmi_eld { bool monitor_present; bool eld_valid; int eld_size; char eld_buffer[ELD_MAX_SIZE]; + struct hdac_hdmi_parsed_eld info; }; struct hdac_hdmi_pin { @@ -76,6 +83,10 @@ struct hdac_hdmi_pin { struct hdac_ext_device *edev; int repoll_count; struct delayed_work work; + struct mutex lock; + bool chmap_set; + unsigned char chmap[8]; /* ALSA API channel-map */ + int channels; /* current number of channels */ }; struct hdac_hdmi_pcm { @@ -100,8 +111,22 @@ struct hdac_hdmi_priv { int num_pin; int num_cvt; struct mutex pin_mutex; + struct hdac_chmap chmap; }; +static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, + int pcm_idx) +{ + struct hdac_hdmi_pcm *pcm; + + list_for_each_entry(pcm, &hdmi->pcm_list, head) { + if (pcm->pcm_id == pcm_idx) + return pcm; + } + + return NULL; +} + static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) { struct hdac_device *hdac = dev_to_hdac_dev(dev); @@ -278,26 +303,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, int i; const u8 *eld_buf; u8 conn_type; - int channels = 2; + int channels, ca; list_for_each_entry(pin, &hdmi->pin_list, head) { if (pin->nid == pin_nid) break; } + ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc, + pin->channels, pin->chmap_set, true, pin->chmap); + + channels = snd_hdac_get_active_channels(ca); + hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels); + + snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, + pin->channels, pin->chmap, pin->chmap_set); + eld_buf = pin->eld.eld_buffer; conn_type = drm_eld_get_conn_type(eld_buf); - /* setup channel count */ - snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, channels - 1); - switch (conn_type) { case DRM_ELD_CONN_TYPE_HDMI: hdmi_audio_infoframe_init(&frame); - /* Default stereo for now */ frame.channels = channels; + frame.channel_allocation = ca; ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer)); if (ret < 0) @@ -311,7 +341,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, dp_ai.len = 0x1b; dp_ai.ver = 0x11 << 2; dp_ai.CC02_CT47 = channels - 1; - dp_ai.CA = 0; + dp_ai.CA = ca; dip = (u8 *)&dp_ai; break; @@ -370,17 +400,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream, struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_priv *hdmi = hdac->private_data; struct hdac_hdmi_dai_pin_map *dai_map; + struct hdac_hdmi_pin *pin; struct hdac_ext_dma_params *dd; int ret; dai_map = &hdmi->dai_map[dai->id]; + pin = dai_map->pin; dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream); dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n", dd->stream_tag, dd->format); + mutex_lock(&pin->lock); + pin->channels = substream->runtime->channels; + ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid, dai_map->pin->nid); + mutex_unlock(&pin->lock); if (ret < 0) return ret; @@ -640,6 +676,12 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + mutex_lock(&dai_map->pin->lock); + dai_map->pin->chmap_set = false; + memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap)); + dai_map->pin->channels = 0; + mutex_unlock(&dai_map->pin->lock); + dai_map->pin = NULL; } } @@ -647,10 +689,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, static int hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt) { + unsigned int chans; + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; int err; - /* Only stereo supported as of now */ - cvt->params.channels_min = cvt->params.channels_max = 2; + chans = get_wcaps(hdac, cvt->nid); + chans = get_wcaps_channels(chans); + + cvt->params.channels_min = 2; + + cvt->params.channels_max = chans; + if (chans > hdmi->chmap.channels_max) + hdmi->chmap.channels_max = chans; err = snd_hdac_query_supported_pcm(hdac, cvt->nid, &cvt->params.rates, @@ -1008,6 +1059,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) return hdac_hdmi_query_cvt_params(&edev->hdac, cvt); } +static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev, + struct hdac_hdmi_pin *pin) +{ + pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER]; +} + static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; @@ -1065,6 +1122,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) snd_jack_report(pcm->jack, SND_JACK_AVOUT); } + hdac_hdmi_parse_eld(edev, pin); print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET, pin->eld.eld_buffer, pin->eld.eld_size); @@ -1123,6 +1181,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) hdmi->num_pin++; pin->edev = edev; + mutex_init(&pin->lock); INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld); return 0; @@ -1342,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = { .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; +static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, + int device) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->pcm && (rtd->pcm->device == device)) + return rtd->pcm; + } + + return NULL; +} + int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) { char jack_name[NAME_SIZE]; @@ -1351,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pcm *pcm; + struct snd_pcm *snd_pcm; + int err; /* * this is a new PCM device, create new pcm and @@ -1362,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) pcm->pcm_id = device; pcm->cvt = hdmi->dai_map[dai->id].cvt; + snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device); + if (snd_pcm) { + err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); + if (err < 0) { + dev_err(&edev->hdac.dev, + "chmap control add failed with err: %d for pcm: %d\n", + err, device); + kfree(pcm); + return err; + } + } + list_add_tail(&pcm->head, &hdmi->pcm_list); sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device); @@ -1378,10 +1464,18 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_pin *pin; + struct hdac_ext_link *hlink = NULL; int ret; edev->scodec = codec; + /* + * hold the ref while we probe, also no need to drop the ref on + * exit, we call pm_runtime_suspend() so that will do for us + */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; @@ -1420,32 +1514,39 @@ static int hdmi_codec_remove(struct snd_soc_codec *codec) } #ifdef CONFIG_PM -static int hdmi_codec_resume(struct snd_soc_codec *codec) +static int hdmi_codec_prepare(struct device *dev) { - struct hdac_ext_device *edev = snd_soc_codec_get_drvdata(codec); + struct hdac_ext_device *edev = to_hda_ext_device(dev); + struct hdac_device *hdac = &edev->hdac; + + pm_runtime_get_sync(&edev->hdac.dev); + + /* + * Power down afg. + * codec_read is preferred over codec_write to set the power state. + * This way verb is send to set the power state and response + * is received. So setting power state is ensured without using loop + * to read the state. + */ + snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + + return 0; +} + +static void hdmi_codec_complete(struct device *dev) +{ + struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pin *pin; struct hdac_device *hdac = &edev->hdac; - struct hdac_bus *bus = hdac->bus; - int err; - unsigned long timeout; - - hdac_hdmi_skl_enable_all_pins(&edev->hdac); - hdac_hdmi_skl_enable_dp12(&edev->hdac); /* Power up afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0)) { + snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); - snd_hdac_codec_write(hdac, hdac->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); - - /* Wait till power state is set to D0 */ - timeout = jiffies + msecs_to_jiffies(1000); - while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0) - && time_before(jiffies, timeout)) { - msleep(50); - } - } + hdac_hdmi_skl_enable_all_pins(&edev->hdac); + hdac_hdmi_skl_enable_dp12(&edev->hdac); /* * As the ELD notify callback request is not entertained while the @@ -1455,44 +1556,96 @@ static int hdmi_codec_resume(struct snd_soc_codec *codec) list_for_each_entry(pin, &hdmi->pin_list, head) hdac_hdmi_present_sense(pin, 1); - /* - * Codec power is turned ON during controller resume. - * Turn it OFF here - */ - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, - "Cannot turn OFF display power on i915, err: %d\n", - err); - return err; - } - - return 0; + pm_runtime_put_sync(&edev->hdac.dev); } #else -#define hdmi_codec_resume NULL +#define hdmi_codec_prepare NULL +#define hdmi_codec_complete NULL #endif static struct snd_soc_codec_driver hdmi_hda_codec = { .probe = hdmi_codec_probe, .remove = hdmi_codec_remove, - .resume = hdmi_codec_resume, .idle_bias_off = true, }; +static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + /* chmap is already set to 0 in caller */ + if (!pin) + return; + + memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap)); +} + +static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap, int prepared) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + mutex_lock(&pin->lock); + pin->chmap_set = true; + memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap)); + if (prepared) + hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid); + mutex_unlock(&pin->lock); +} + +static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + return pin ? true:false; +} + +static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + if (!pin || !pin->eld.eld_valid) + return 0; + + return pin->eld.info.spk_alloc; +} + static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) { struct hdac_device *codec = &edev->hdac; struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; + struct hdac_ext_link *hlink = NULL; int num_dais = 0; int ret = 0; + /* hold the ref while we probe */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; edev->private_data = hdmi_priv; + snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap); + hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; + hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; + hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; + hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; dev_set_drvdata(&codec->dev, edev); @@ -1521,8 +1674,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) } /* ASoC specific initialization */ - return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, - hdmi_dais, num_dais); + ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, + hdmi_dais, num_dais); + + snd_hdac_ext_bus_link_put(edev->ebus, hlink); + + return ret; } static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) @@ -1561,7 +1718,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; - unsigned long timeout; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1570,26 +1728,24 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) if (!bus) return 0; - /* Power down afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3)) { - snd_hdac_codec_write(hdac, hdac->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - - /* Wait till power state is set to D3 */ - timeout = jiffies + msecs_to_jiffies(1000); - while (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D3) - && time_before(jiffies, timeout)) { - - msleep(50); - } - } - + /* + * Power down afg. + * codec_read is preferred over codec_write to set the power state. + * This way verb is send to set the power state and response + * is received. So setting power state is ensured without using loop + * to read the state. + */ + snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); err = snd_hdac_display_power(bus, false); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); return err; } + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_put(ebus, hlink); + return 0; } @@ -1598,6 +1754,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1606,6 +1764,9 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_get(ebus, hlink); + err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); @@ -1616,9 +1777,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev) hdac_hdmi_skl_enable_dp12(&edev->hdac); /* Power up afg */ - if (!snd_hdac_check_power_state(hdac, hdac->afg, AC_PWRST_D0)) - snd_hdac_codec_write(hdac, hdac->afg, 0, - AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); return 0; } @@ -1629,6 +1789,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev) static const struct dev_pm_ops hdac_hdmi_pm = { SET_RUNTIME_PM_OPS(hdac_hdmi_runtime_suspend, hdac_hdmi_runtime_resume, NULL) + .prepare = hdmi_codec_prepare, + .complete = hdmi_codec_complete, }; static const struct hda_device_id hdmi_list[] = { diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c new file mode 100644 index 000000000000..8e36e883e453 --- /dev/null +++ b/sound/soc/codecs/hdmi-codec.c @@ -0,0 +1,432 @@ +/* + * ALSA SoC codec for HDMI encoder drivers + * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Jyri Sarha <jsarha@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ +#include <linux/module.h> +#include <linux/string.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pcm_drm_eld.h> +#include <sound/hdmi-codec.h> +#include <sound/pcm_iec958.h> + +#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */ + +struct hdmi_codec_priv { + struct hdmi_codec_pdata hcd; + struct snd_soc_dai_driver *daidrv; + struct hdmi_codec_daifmt daifmt[2]; + struct mutex current_stream_lock; + struct snd_pcm_substream *current_stream; + struct snd_pcm_hw_constraint_list ratec; + uint8_t eld[MAX_ELD_BYTES]; +}; + +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +enum { + DAI_ID_I2S = 0, + DAI_ID_SPDIF, +}; + +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = sizeof(hcp->eld); + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld)); + + return 0; +} + +static const struct snd_kcontrol_new hdmi_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, + }, +}; + +static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + mutex_lock(&hcp->current_stream_lock); + if (!hcp->current_stream) { + hcp->current_stream = substream; + } else if (hcp->current_stream != substream) { + dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + ret = -EINVAL; + } + mutex_unlock(&hcp->current_stream_lock); + + return ret; +} + +static int hdmi_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + if (hcp->hcd.ops->audio_startup) { + ret = hcp->hcd.ops->audio_startup(dai->dev->parent); + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); + return ret; + } + } + + if (hcp->hcd.ops->get_eld) { + ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld, + sizeof(hcp->eld)); + + if (!ret) { + ret = snd_pcm_hw_constraint_eld(substream->runtime, + hcp->eld); + if (ret) + return ret; + } + } + return 0; +} + +static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + WARN_ON(hcp->current_stream != substream); + + hcp->hcd.ops->audio_shutdown(dai->dev->parent); + + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); +} + +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_params hp = { + .iec = { + .status = { 0 }, + .subcode = { 0 }, + .pad = 0, + .dig_subframe = { 0 }, + } + }; + int ret; + + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, + params_width(params), params_rate(params), + params_channels(params)); + + if (params_width(params) > 24) + params->msbits = 24; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, + sizeof(hp.iec.status)); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + hdmi_audio_infoframe_init(&hp.cea); + hp.cea.channels = params_channels(params); + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; + + hp.sample_width = params_width(params); + hp.sample_rate = params_rate(params); + hp.channels = params_channels(params); + + return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id], + &hp); +} + +static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_daifmt cf = { 0 }; + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (dai->id == DAI_ID_SPDIF) { + cf.fmt = HDMI_SPDIF; + } else { + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + } + + hcp->daifmt[dai->id] = cf; + + return ret; +} + +static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (hcp->hcd.ops->digital_mute) + return hcp->hcd.ops->digital_mute(dai->dev->parent, mute); + + return 0; +} + +static const struct snd_soc_dai_ops hdmi_dai_ops = { + .startup = hdmi_codec_startup, + .shutdown = hdmi_codec_shutdown, + .hw_params = hdmi_codec_hw_params, + .set_fmt = hdmi_codec_set_fmt, + .digital_mute = hdmi_codec_digital_mute, +}; + + +#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +/* + * This list is only for formats allowed on the I2S bus. So there is + * some formats listed that are not supported by HDMI interface. For + * instance allowing the 32-bit formats enables 24-precision with CPU + * DAIs that do not support 24-bit formats. If the extra formats cause + * problems, we should add the video side driver an option to disable + * them. + */ +#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", + .id = DAI_ID_I2S, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = HDMI_RATES, + .formats = I2S_FORMATS, + .sig_bits = 24, + }, + .ops = &hdmi_dai_ops, +}; + +static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", + .id = DAI_ID_SPDIF, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = HDMI_RATES, + .formats = SPDIF_FORMATS, + }, + .ops = &hdmi_dai_ops, +}; + +static struct snd_soc_codec_driver hdmi_codec = { + .controls = hdmi_controls, + .num_controls = ARRAY_SIZE(hdmi_controls), + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + +static int hdmi_codec_probe(struct platform_device *pdev) +{ + struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; + struct device *dev = &pdev->dev; + struct hdmi_codec_priv *hcp; + int dai_count, i = 0; + int ret; + + dev_dbg(dev, "%s()\n", __func__); + + if (!hcd) { + dev_err(dev, "%s: No plalform data\n", __func__); + return -EINVAL; + } + + dai_count = hcd->i2s + hcd->spdif; + if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params || + !hcd->ops->audio_shutdown) { + dev_err(dev, "%s: Invalid parameters\n", __func__); + return -EINVAL; + } + + hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL); + if (!hcp) + return -ENOMEM; + + hcp->hcd = *hcd; + mutex_init(&hcp->current_stream_lock); + + hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + GFP_KERNEL); + if (!hcp->daidrv) + return -ENOMEM; + + if (hcd->i2s) { + hcp->daidrv[i] = hdmi_i2s_dai; + hcp->daidrv[i].playback.channels_max = + hcd->max_i2s_channels; + i++; + } + + if (hcd->spdif) + hcp->daidrv[i] = hdmi_spdif_dai; + + ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, + dai_count); + if (ret) { + dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n", + __func__, ret); + return ret; + } + + dev_set_drvdata(dev, hcp); + return 0; +} + +static int hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_codec_driver = { + .driver = { + .name = HDMI_CODEC_DRV_NAME, + }, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, +}; + +module_platform_driver(hdmi_codec_driver); + +MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>"); +MODULE_DESCRIPTION("HDMI Audio Codec Driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME); diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 000000000000..cf0a39bb631a --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,441 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN, + 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0, + max98371_dht_max_gain), + SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0, + max98371_dht_min_gain), + SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0, + max98371_dht_rot_gain), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, ®); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere@gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 000000000000..9f6330964d98 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c8729984c2b..683769f0f246 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -343,9 +343,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -607,6 +610,16 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -728,7 +741,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to read irq status\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -1141,33 +1157,74 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } - - ret = regcache_sync(nau8825->regmap); - if (ret) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } } - break; case SND_SOC_BIAS_OFF: if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); - - regcache_mark_dirty(nau8825->regmap); break; } return 0; } +#ifdef CONFIG_PM +static int nau8825_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + regcache_cache_only(nau8825->regmap, true); + regcache_mark_dirty(nau8825->regmap); + + return 0; +} + +static int nau8825_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* The chip may lose power and reset in S3. regcache_sync restores + * register values including configurations for sysclk, irq, and + * jack/button detection. + */ + regcache_cache_only(nau8825->regmap, false); + regcache_sync(nau8825->regmap); + + /* Check the jack plug status directly. If the headset is unplugged + * during S3 when the chip has no power, there will be no jack + * detection irq even after the nau8825_restart_jack_detection below, + * because the chip just thinks no headset has ever been plugged in. + */ + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + + enable_irq(nau8825->irq); + + /* Run jack detection to check the type (OMTP or CTIA) of the headset + * if there is one. This handles the case where a different type of + * headset is plugged in during S3. This triggers an IRQ iff a headset + * is already plugged in. + */ + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} +#else +#define nau8825_suspend NULL +#define nau8825_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_suspend, + .resume = nau8825_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1277,16 +1334,6 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); @@ -1354,36 +1401,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1410,7 +1427,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c new file mode 100644 index 000000000000..ed515677409b --- /dev/null +++ b/sound/soc/codecs/pcm5102a.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM5102A codec + * + * Author: Florian Meier <florian.meier@koalo.de> + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> + +static struct snd_soc_dai_driver pcm5102a_dai = { + .name = "pcm5102a-hifi", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; + +static int pcm5102a_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a, + &pcm5102a_dai, 1); +} + +static int pcm5102a_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id pcm5102a_of_match[] = { + { .compatible = "ti,pcm5102a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm5102a_of_match); + +static struct platform_driver pcm5102a_codec_driver = { + .probe = pcm5102a_probe, + .remove = pcm5102a_remove, + .driver = { + .name = "pcm5102a-codec", + .owner = THIS_MODULE, + .of_match_table = pcm5102a_of_match, + }, +}; + +module_platform_driver(pcm5102a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM5102A codec driver"); +MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 68cf8d5a174f..f80cfe4d2ef2 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1127,6 +1128,17 @@ static const struct acpi_device_id rt298_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); +static const struct dmi_system_id force_combo_jack_table[] = { + { + .ident = "Intel Broxton P", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"), + DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P") + } + }, + { } +}; + static int rt298_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1179,11 +1191,16 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } + if (dmi_check_system(force_combo_jack_table)) { + rt298->pdata.cbj_en = true; + rt298->pdata.gpio2_en = false; + } + /* VREF Charging */ regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80); regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e8b5ba04417a..09e8988bbb2d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -359,7 +359,7 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, /* Interface data select */ static const char * const rt5640_data_select[] = { - "Normal", "left copy to right", "right copy to left", "Swap"}; + "Normal", "Swap", "left copy to right", "right copy to left"}; static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 1761c3a98b76..58b664b06c16 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -443,39 +443,39 @@ #define RT5640_IF1_DAC_SEL_MASK (0x3 << 14) #define RT5640_IF1_DAC_SEL_SFT 14 #define RT5640_IF1_DAC_SEL_NOR (0x0 << 14) -#define RT5640_IF1_DAC_SEL_L2R (0x1 << 14) -#define RT5640_IF1_DAC_SEL_R2L (0x2 << 14) -#define RT5640_IF1_DAC_SEL_SWAP (0x3 << 14) +#define RT5640_IF1_DAC_SEL_SWAP (0x1 << 14) +#define RT5640_IF1_DAC_SEL_L2R (0x2 << 14) +#define RT5640_IF1_DAC_SEL_R2L (0x3 << 14) #define RT5640_IF1_ADC_SEL_MASK (0x3 << 12) #define RT5640_IF1_ADC_SEL_SFT 12 #define RT5640_IF1_ADC_SEL_NOR (0x0 << 12) -#define RT5640_IF1_ADC_SEL_L2R (0x1 << 12) -#define RT5640_IF1_ADC_SEL_R2L (0x2 << 12) -#define RT5640_IF1_ADC_SEL_SWAP (0x3 << 12) +#define RT5640_IF1_ADC_SEL_SWAP (0x1 << 12) +#define RT5640_IF1_ADC_SEL_L2R (0x2 << 12) +#define RT5640_IF1_ADC_SEL_R2L (0x3 << 12) #define RT5640_IF2_DAC_SEL_MASK (0x3 << 10) #define RT5640_IF2_DAC_SEL_SFT 10 #define RT5640_IF2_DAC_SEL_NOR (0x0 << 10) -#define RT5640_IF2_DAC_SEL_L2R (0x1 << 10) -#define RT5640_IF2_DAC_SEL_R2L (0x2 << 10) -#define RT5640_IF2_DAC_SEL_SWAP (0x3 << 10) +#define RT5640_IF2_DAC_SEL_SWAP (0x1 << 10) +#define RT5640_IF2_DAC_SEL_L2R (0x2 << 10) +#define RT5640_IF2_DAC_SEL_R2L (0x3 << 10) #define RT5640_IF2_ADC_SEL_MASK (0x3 << 8) #define RT5640_IF2_ADC_SEL_SFT 8 #define RT5640_IF2_ADC_SEL_NOR (0x0 << 8) -#define RT5640_IF2_ADC_SEL_L2R (0x1 << 8) -#define RT5640_IF2_ADC_SEL_R2L (0x2 << 8) -#define RT5640_IF2_ADC_SEL_SWAP (0x3 << 8) +#define RT5640_IF2_ADC_SEL_SWAP (0x1 << 8) +#define RT5640_IF2_ADC_SEL_L2R (0x2 << 8) +#define RT5640_IF2_ADC_SEL_R2L (0x3 << 8) #define RT5640_IF3_DAC_SEL_MASK (0x3 << 6) #define RT5640_IF3_DAC_SEL_SFT 6 #define RT5640_IF3_DAC_SEL_NOR (0x0 << 6) -#define RT5640_IF3_DAC_SEL_L2R (0x1 << 6) -#define RT5640_IF3_DAC_SEL_R2L (0x2 << 6) -#define RT5640_IF3_DAC_SEL_SWAP (0x3 << 6) +#define RT5640_IF3_DAC_SEL_SWAP (0x1 << 6) +#define RT5640_IF3_DAC_SEL_L2R (0x2 << 6) +#define RT5640_IF3_DAC_SEL_R2L (0x3 << 6) #define RT5640_IF3_ADC_SEL_MASK (0x3 << 4) #define RT5640_IF3_ADC_SEL_SFT 4 #define RT5640_IF3_ADC_SEL_NOR (0x0 << 4) -#define RT5640_IF3_ADC_SEL_L2R (0x1 << 4) -#define RT5640_IF3_ADC_SEL_R2L (0x2 << 4) -#define RT5640_IF3_ADC_SEL_SWAP (0x3 << 4) +#define RT5640_IF3_ADC_SEL_SWAP (0x1 << 4) +#define RT5640_IF3_ADC_SEL_L2R (0x2 << 4) +#define RT5640_IF3_ADC_SEL_R2L (0x3 << 4) /* REC Left Mixer Control 1 (0x3b) */ #define RT5640_G_HP_L_RM_L_MASK (0x7 << 13) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7af5e7380d61..3c6594da6c9c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (btn_type == 0)/* button release */ report = rt5645->jack_type; else { - if (rt5645->pdata.jd_invert) { - mod_timer(&rt5645->btn_check_timer, - msecs_to_jiffies(100)); - } + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); } break; @@ -3557,6 +3555,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Setzer", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"), + }, + }, { } }; @@ -3810,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); } + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad41a34..b8d19b77bde9 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -4,6 +4,9 @@ * Copyright (C) 2015 Google, Inc. * Copyright (c) 2013 Daniel Mack <zonque@gmail.com> * + * TAS5721 support: + * Copyright (C) 2016 Petr Kulhavy, Barix AG <petr@barix.com> + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -57,6 +60,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS571X_INPUT_MUX_REG: + case TAS571X_CH4_SRC_SELECT_REG: + case TAS571X_PWM_MUX_REG: + return 4; default: return 1; } @@ -167,6 +174,23 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } +static int tas571x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 sysctl2; + int ret; + + sysctl2 = mute ? TAS571X_SYS_CTRL_2_SDN_MASK : 0; + + ret = snd_soc_update_bits(codec, + TAS571X_SYS_CTRL_2_REG, + TAS571X_SYS_CTRL_2_SDN_MASK, + sysctl2); + usleep_range(1000, 2000); + + return ret; +} + static int tas571x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -214,6 +238,7 @@ static int tas571x_set_bias_level(struct snd_soc_codec *codec, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, + .digital_mute = tas571x_mute, }; static const char *const tas5711_supply_names[] = { @@ -241,6 +266,26 @@ static const struct snd_kcontrol_new tas5711_controls[] = { 1, 1), }; +static const struct regmap_range tas571x_readonly_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_DEV_ID_REG), +}; + +static const struct regmap_range tas571x_volatile_regs_range[] = { + regmap_reg_range(TAS571X_CLK_CTRL_REG, TAS571X_ERR_STATUS_REG), + regmap_reg_range(TAS571X_OSC_TRIM_REG, TAS571X_OSC_TRIM_REG), +}; + +static const struct regmap_access_table tas571x_write_regs = { + .no_ranges = tas571x_readonly_regs_range, + .n_no_ranges = ARRAY_SIZE(tas571x_readonly_regs_range), +}; + +static const struct regmap_access_table tas571x_volatile_regs = { + .yes_ranges = tas571x_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(tas571x_volatile_regs_range), + +}; + static const struct reg_default tas5711_reg_defaults[] = { { 0x04, 0x05 }, { 0x05, 0x40 }, @@ -260,6 +305,8 @@ static const struct regmap_config tas5711_regmap_config = { .reg_defaults = tas5711_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5711_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; static const struct tas571x_chip tas5711_chip = { @@ -314,6 +361,8 @@ static const struct regmap_config tas5717_regmap_config = { .reg_defaults = tas5717_reg_defaults, .num_reg_defaults = ARRAY_SIZE(tas5717_reg_defaults), .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, }; /* This entry is reused for tas5719 as the software interface is identical. */ @@ -326,6 +375,77 @@ static const struct tas571x_chip tas5717_chip = { .vol_reg_size = 2, }; +static const char *const tas5721_supply_names[] = { + "AVDD", + "DVDD", + "DRVDD", + "PVDD", +}; + +static const struct snd_kcontrol_new tas5721_controls[] = { + SOC_SINGLE_TLV("Master Volume", + TAS571X_MVOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", + TAS571X_CH1_VOL_REG, + TAS571X_CH2_VOL_REG, + 0, 0xff, 1, tas5711_volume_tlv), + SOC_DOUBLE("Speaker Switch", + TAS571X_SOFT_MUTE_REG, + TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, + 1, 1), +}; + +static const struct reg_default tas5721_reg_defaults[] = { + {TAS571X_CLK_CTRL_REG, 0x6c}, + {TAS571X_DEV_ID_REG, 0x00}, + {TAS571X_ERR_STATUS_REG, 0x00}, + {TAS571X_SYS_CTRL_1_REG, 0xa0}, + {TAS571X_SDI_REG, 0x05}, + {TAS571X_SYS_CTRL_2_REG, 0x40}, + {TAS571X_SOFT_MUTE_REG, 0x00}, + {TAS571X_MVOL_REG, 0xff}, + {TAS571X_CH1_VOL_REG, 0x30}, + {TAS571X_CH2_VOL_REG, 0x30}, + {TAS571X_CH3_VOL_REG, 0x30}, + {TAS571X_VOL_CFG_REG, 0x91}, + {TAS571X_MODULATION_LIMIT_REG, 0x02}, + {TAS571X_IC_DELAY_CH1_REG, 0xac}, + {TAS571X_IC_DELAY_CH2_REG, 0x54}, + {TAS571X_IC_DELAY_CH3_REG, 0xac}, + {TAS571X_IC_DELAY_CH4_REG, 0x54}, + {TAS571X_PWM_CH_SDN_GROUP_REG, 0x30}, + {TAS571X_START_STOP_PERIOD_REG, 0x0f}, + {TAS571X_OSC_TRIM_REG, 0x82}, + {TAS571X_BKND_ERR_REG, 0x02}, + {TAS571X_INPUT_MUX_REG, 0x17772}, + {TAS571X_CH4_SRC_SELECT_REG, 0x4303}, + {TAS571X_PWM_MUX_REG, 0x1021345}, +}; + +static const struct regmap_config tas5721_regmap_config = { + .reg_bits = 8, + .val_bits = 32, + .max_register = 0xff, + .reg_read = tas571x_reg_read, + .reg_write = tas571x_reg_write, + .reg_defaults = tas5721_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5721_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .wr_table = &tas571x_write_regs, + .volatile_table = &tas571x_volatile_regs, +}; + + +static const struct tas571x_chip tas5721_chip = { + .supply_names = tas5721_supply_names, + .num_supply_names = ARRAY_SIZE(tas5721_supply_names), + .controls = tas5711_controls, + .num_controls = ARRAY_SIZE(tas5711_controls), + .regmap_config = &tas5721_regmap_config, + .vol_reg_size = 1, +}; + static const struct snd_soc_dapm_widget tas571x_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), @@ -386,11 +506,10 @@ static int tas571x_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, priv); of_id = of_match_device(tas571x_of_match, dev); - if (!of_id) { - dev_err(dev, "Unknown device type\n"); - return -EINVAL; - } - priv->chip = of_id->data; + if (of_id) + priv->chip = of_id->data; + else + priv->chip = (void *) id->driver_data; priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk) && PTR_ERR(priv->mclk) != -ENOENT) { @@ -445,10 +564,6 @@ static int tas571x_i2c_probe(struct i2c_client *client, if (ret) return ret; - ret = regmap_update_bits(priv->regmap, TAS571X_SYS_CTRL_2_REG, - TAS571X_SYS_CTRL_2_SDN_MASK, 0); - if (ret) - return ret; memcpy(&priv->codec_driver, &tas571x_codec, sizeof(priv->codec_driver)); priv->codec_driver.controls = priv->chip->controls; @@ -486,14 +601,16 @@ static const struct of_device_id tas571x_of_match[] = { { .compatible = "ti,tas5711", .data = &tas5711_chip, }, { .compatible = "ti,tas5717", .data = &tas5717_chip, }, { .compatible = "ti,tas5719", .data = &tas5717_chip, }, + { .compatible = "ti,tas5721", .data = &tas5721_chip, }, { } }; MODULE_DEVICE_TABLE(of, tas571x_of_match); static const struct i2c_device_id tas571x_i2c_id[] = { - { "tas5711", 0 }, - { "tas5717", 0 }, - { "tas5719", 0 }, + { "tas5711", (kernel_ulong_t) &tas5711_chip }, + { "tas5717", (kernel_ulong_t) &tas5717_chip }, + { "tas5719", (kernel_ulong_t) &tas5717_chip }, + { "tas5721", (kernel_ulong_t) &tas5721_chip }, { } }; MODULE_DEVICE_TABLE(i2c, tas571x_i2c_id); diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471232cd..cf800c364f0f 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -13,6 +13,10 @@ #define _TAS571X_H /* device registers */ +#define TAS571X_CLK_CTRL_REG 0x00 +#define TAS571X_DEV_ID_REG 0x01 +#define TAS571X_ERR_STATUS_REG 0x02 +#define TAS571X_SYS_CTRL_1_REG 0x03 #define TAS571X_SDI_REG 0x04 #define TAS571X_SDI_FMT_MASK 0x0f @@ -27,7 +31,25 @@ #define TAS571X_MVOL_REG 0x07 #define TAS571X_CH1_VOL_REG 0x08 #define TAS571X_CH2_VOL_REG 0x09 +#define TAS571X_CH3_VOL_REG 0x0a +#define TAS571X_VOL_CFG_REG 0x0e +#define TAS571X_MODULATION_LIMIT_REG 0x10 +#define TAS571X_IC_DELAY_CH1_REG 0x11 +#define TAS571X_IC_DELAY_CH2_REG 0x12 +#define TAS571X_IC_DELAY_CH3_REG 0x13 +#define TAS571X_IC_DELAY_CH4_REG 0x14 +#define TAS571X_PWM_CH_SDN_GROUP_REG 0x19 /* N/A on TAS5717, TAS5719 */ +#define TAS571X_PWM_CH1_SDN_MASK (1<<0) +#define TAS571X_PWM_CH2_SDN_SHIFT (1<<1) +#define TAS571X_PWM_CH3_SDN_SHIFT (1<<2) +#define TAS571X_PWM_CH4_SDN_SHIFT (1<<3) + +#define TAS571X_START_STOP_PERIOD_REG 0x1a #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS571X_BKND_ERR_REG 0x1c +#define TAS571X_INPUT_MUX_REG 0x20 +#define TAS571X_CH4_SRC_SELECT_REG 0x21 +#define TAS571X_PWM_MUX_REG 0x25 #endif /* _TAS571X_H */ diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c new file mode 100644 index 000000000000..f54fb46b77c2 --- /dev/null +++ b/sound/soc/codecs/tas5720.c @@ -0,0 +1,620 @@ +/* + * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg <dannenberg@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/regulator/consumer.h> +#include <linux/delay.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tas5720.h" + +/* Define how often to check (and clear) the fault status register (in ms) */ +#define TAS5720_FAULT_CHECK_INTERVAL 200 + +static const char * const tas5720_supply_names[] = { + "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ + "pvdd", /* Class-D amp and analog power supply (connected). */ +}; + +#define TAS5720_NUM_SUPPLIES ARRAY_SIZE(tas5720_supply_names) + +struct tas5720_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas5720_client; + struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; + struct delayed_work fault_check_work; + unsigned int last_fault; +}; + +static int tas5720_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int rate = params_rate(params); + bool ssz_ds; + int ret; + + switch (rate) { + case 44100: + case 48000: + ssz_ds = false; + break; + case 88200: + case 96000: + ssz_ds = true; + break; + default: + dev_err(codec->dev, "unsupported sample rate: %u\n", rate); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SSZ_DS, ssz_ds); + if (ret < 0) { + dev_err(codec->dev, "error setting sample rate: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + int ret; + + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | + SND_SOC_DAIFMT_INV_MASK)) { + case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): + /* 1st data bit occur one BCLK cycle after the frame sync */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF): + /* + * Note that although the TAS5720 does not have a dedicated DSP + * mode it doesn't care about the LRCLK duty cycle during TDM + * operation. Therefore we can use the device's I2S mode with + * its delaying of the 1st data bit to receive DSP_A formatted + * data. See device datasheet for additional details. + */ + serial_format = TAS5720_SAIF_I2S; + break; + case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF): + /* + * Similar to DSP_A, we can use the fact that the TAS5720 does + * not care about the LRCLK duty cycle during TDM to receive + * DSP_B formatted data in LEFTJ mode (no delaying of the 1st + * data bit). + */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF): + /* No delay after the frame sync */ + serial_format = TAS5720_SAIF_LEFTJ; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_SAIF_FORMAT_MASK, + serial_format); + if (ret < 0) { + dev_err(codec->dev, "error setting SAIF format: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int first_slot; + int ret; + + if (!tx_mask) { + dev_err(codec->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + /* + * Determine the first slot that is being requested. We will only + * use the first slot that is found since the TAS5720 is a mono + * amplifier. + */ + first_slot = __ffs(tx_mask); + + if (first_slot > 7) { + dev_err(codec->dev, "slot selection out of bounds (%u)\n", + first_slot); + return -EINVAL; + } + + /* Enable manual TDM slot selection (instead of I2C ID based) */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL1_REG, + TAS5720_TDM_CFG_SRC, TAS5720_TDM_CFG_SRC); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* Configure the TDM slot to process audio from */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_TDM_SLOT_SEL_MASK, first_slot); + if (ret < 0) + goto error_snd_soc_update_bits; + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring TDM mode: %d\n", ret); + return ret; +} + +static int tas5720_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int ret; + + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, mute ? TAS5720_MUTE : 0); + if (ret < 0) { + dev_err(codec->dev, "error (un-)muting device: %d\n", ret); + return ret; + } + + return 0; +} + +static void tas5720_fault_check_work(struct work_struct *work) +{ + struct tas5720_data *tas5720 = container_of(work, struct tas5720_data, + fault_check_work.work); + struct device *dev = tas5720->codec->dev; + unsigned int curr_fault; + int ret; + + ret = regmap_read(tas5720->regmap, TAS5720_FAULT_REG, &curr_fault); + if (ret < 0) { + dev_err(dev, "failed to read FAULT register: %d\n", ret); + goto out; + } + + /* Check/handle all errors except SAIF clock errors */ + curr_fault &= TAS5720_OCE | TAS5720_DCE | TAS5720_OTE; + + /* + * Only flag errors once for a given occurrence. This is needed as + * the TAS5720 will take time clearing the fault condition internally + * during which we don't want to bombard the system with the same + * error message over and over. + */ + if ((curr_fault & TAS5720_OCE) && !(tas5720->last_fault & TAS5720_OCE)) + dev_crit(dev, "experienced an over current hardware fault\n"); + + if ((curr_fault & TAS5720_DCE) && !(tas5720->last_fault & TAS5720_DCE)) + dev_crit(dev, "experienced a DC detection fault\n"); + + if ((curr_fault & TAS5720_OTE) && !(tas5720->last_fault & TAS5720_OTE)) + dev_crit(dev, "experienced an over temperature fault\n"); + + /* Store current fault value so we can detect any changes next time */ + tas5720->last_fault = curr_fault; + + if (!curr_fault) + goto out; + + /* + * Periodically toggle SDZ (shutdown bit) H->L->H to clear any latching + * faults as long as a fault condition persists. Always going through + * the full sequence no matter the first return value to minimizes + * chances for the device to end up in shutdown mode. + */ + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + + ret = regmap_write_bits(tas5720->regmap, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) + dev_err(dev, "failed to write POWER_CTRL register: %d\n", ret); + +out: + /* Schedule the next fault check at the specified interval */ + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); +} + +static int tas5720_codec_probe(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + unsigned int device_id; + int ret; + + tas5720->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret != 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); + if (ret < 0) { + dev_err(codec->dev, "failed to read device ID register: %d\n", + ret); + goto probe_fail; + } + + if (device_id != TAS5720_DEVICE_ID) { + dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", + TAS5720_DEVICE_ID, device_id); + ret = -ENODEV; + goto probe_fail; + } + + /* Set device to mute */ + ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, + TAS5720_MUTE, TAS5720_MUTE); + if (ret < 0) + goto error_snd_soc_update_bits; + + /* + * Enter shutdown mode - our default when not playing audio - to + * minimize current consumption. On the TAS5720 there is no real down + * side doing so as all device registers are preserved and the wakeup + * of the codec is rather quick which we do using a dapm widget. + */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) + goto error_snd_soc_update_bits; + + INIT_DELAYED_WORK(&tas5720->fault_check_work, tas5720_fault_check_work); + + return 0; + +error_snd_soc_update_bits: + dev_err(codec->dev, "error configuring device registers: %d\n", ret); + +probe_fail: + regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + return ret; +} + +static int tas5720_codec_remove(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + cancel_delayed_work_sync(&tas5720->fault_check_work); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +}; + +static int tas5720_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (event & SND_SOC_DAPM_POST_PMU) { + /* Take TAS5720 out of shutdown mode */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, TAS5720_SDZ); + if (ret < 0) { + dev_err(codec->dev, "error waking codec: %d\n", ret); + return ret; + } + + /* + * Observe codec shutdown-to-active time. The datasheet only + * lists a nominal value however just use-it as-is without + * additional padding to minimize the delay introduced in + * starting to play audio (actually there is other setup done + * by the ASoC framework that will provide additional delays, + * so we should always be safe). + */ + msleep(25); + + /* Turn on TAS5720 periodic fault checking/handling */ + tas5720->last_fault = 0; + schedule_delayed_work(&tas5720->fault_check_work, + msecs_to_jiffies(TAS5720_FAULT_CHECK_INTERVAL)); + } else if (event & SND_SOC_DAPM_PRE_PMD) { + /* Disable TAS5720 periodic fault checking/handling */ + cancel_delayed_work_sync(&tas5720->fault_check_work); + + /* Place TAS5720 in shutdown mode to minimize current draw */ + ret = snd_soc_update_bits(codec, TAS5720_POWER_CTRL_REG, + TAS5720_SDZ, 0); + if (ret < 0) { + dev_err(codec->dev, "error shutting down codec: %d\n", + ret); + return ret; + } + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas5720_suspend(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + regcache_cache_only(tas5720->regmap, true); + regcache_mark_dirty(tas5720->regmap); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) + dev_err(codec->dev, "failed to disable supplies: %d\n", ret); + + return ret; +} + +static int tas5720_resume(struct snd_soc_codec *codec) +{ + struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas5720->supplies), + tas5720->supplies); + if (ret < 0) { + dev_err(codec->dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(tas5720->regmap, false); + + ret = regcache_sync(tas5720->regmap); + if (ret < 0) { + dev_err(codec->dev, "failed to sync regcache: %d\n", ret); + return ret; + } + + return 0; +} +#else +#define tas5720_suspend NULL +#define tas5720_resume NULL +#endif + +static bool tas5720_is_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5720_DEVICE_ID_REG: + case TAS5720_FAULT_REG: + return true; + default: + return false; + } +} + +static const struct regmap_config tas5720_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS5720_MAX_REG, + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5720_is_volatile_reg, +}; + +/* + * DAC analog gain. There are four discrete values to select from, ranging + * from 19.2 dB to 26.3dB. + */ +static const DECLARE_TLV_DB_RANGE(dac_analog_tlv, + 0x0, 0x0, TLV_DB_SCALE_ITEM(1920, 0, 0), + 0x1, 0x1, TLV_DB_SCALE_ITEM(2070, 0, 0), + 0x2, 0x2, TLV_DB_SCALE_ITEM(2350, 0, 0), + 0x3, 0x3, TLV_DB_SCALE_ITEM(2630, 0, 0), +); + +/* + * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that + * setting the gain below -100 dB (register value <0x7) is effectively a MUTE + * as per device datasheet. + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0); + +static const struct snd_kcontrol_new tas5720_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS5720_VOLUME_CTRL_REG, 0, 0xff, 0, dac_tlv), + SOC_SINGLE_TLV("Speaker Driver Analog Gain", TAS5720_ANALOG_CTRL_REG, + TAS5720_ANALOG_GAIN_SHIFT, 3, 0, dac_analog_tlv), +}; + +static const struct snd_soc_dapm_widget tas5720_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas5720_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT") +}; + +static const struct snd_soc_dapm_route tas5720_audio_map[] = { + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5720 = { + .probe = tas5720_codec_probe, + .remove = tas5720_codec_remove, + .suspend = tas5720_suspend, + .resume = tas5720_resume, + + .controls = tas5720_snd_controls, + .num_controls = ARRAY_SIZE(tas5720_snd_controls), + .dapm_widgets = tas5720_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas5720_dapm_widgets), + .dapm_routes = tas5720_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas5720_audio_map), +}; + +/* PCM rates supported by the TAS5720 driver */ +#define TAS5720_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +/* Formats supported by TAS5720 driver */ +#define TAS5720_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops tas5720_speaker_dai_ops = { + .hw_params = tas5720_hw_params, + .set_fmt = tas5720_set_dai_fmt, + .set_tdm_slot = tas5720_set_dai_tdm_slot, + .digital_mute = tas5720_mute, +}; + +/* + * TAS5720 DAI structure + * + * Note that were are advertising .playback.channels_max = 2 despite this being + * a mono amplifier. The reason for that is that some serial ports such as TI's + * McASP module have a minimum number of channels (2) that they can output. + * Advertising more channels than we have will allow us to interface with such + * a serial port without really any negative side effects as the TAS5720 will + * simply ignore any extra channel(s) asides from the one channel that is + * configured to be played back. + */ +static struct snd_soc_dai_driver tas5720_dai[] = { + { + .name = "tas5720-amplifier", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TAS5720_RATES, + .formats = TAS5720_FORMATS, + }, + .ops = &tas5720_speaker_dai_ops, + }, +}; + +static int tas5720_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas5720_data *data; + int ret; + int i; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->tas5720_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas5720_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + dev_set_drvdata(dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas5720, + tas5720_dai, ARRAY_SIZE(tas5720_dai)); + if (ret < 0) { + dev_err(dev, "failed to register codec: %d\n", ret); + return ret; + } + + return 0; +} + +static int tas5720_remove(struct i2c_client *client) +{ + struct device *dev = &client->dev; + + snd_soc_unregister_codec(dev); + + return 0; +} + +static const struct i2c_device_id tas5720_id[] = { + { "tas5720", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5720_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas5720_of_match[] = { + { .compatible = "ti,tas5720", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas5720_of_match); +#endif + +static struct i2c_driver tas5720_i2c_driver = { + .driver = { + .name = "tas5720", + .of_match_table = of_match_ptr(tas5720_of_match), + }, + .probe = tas5720_probe, + .remove = tas5720_remove, + .id_table = tas5720_id, +}; + +module_i2c_driver(tas5720_i2c_driver); + +MODULE_AUTHOR("Andreas Dannenberg <dannenberg@ti.com>"); +MODULE_DESCRIPTION("TAS5720 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h new file mode 100644 index 000000000000..3d077c779b12 --- /dev/null +++ b/sound/soc/codecs/tas5720.h @@ -0,0 +1,90 @@ +/* + * tas5720.h - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier + * + * Copyright (C)2015-2016 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Andreas Dannenberg <dannenberg@ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __TAS5720_H__ +#define __TAS5720_H__ + +/* Register Address Map */ +#define TAS5720_DEVICE_ID_REG 0x00 +#define TAS5720_POWER_CTRL_REG 0x01 +#define TAS5720_DIGITAL_CTRL1_REG 0x02 +#define TAS5720_DIGITAL_CTRL2_REG 0x03 +#define TAS5720_VOLUME_CTRL_REG 0x04 +#define TAS5720_ANALOG_CTRL_REG 0x06 +#define TAS5720_FAULT_REG 0x08 +#define TAS5720_DIGITAL_CLIP2_REG 0x10 +#define TAS5720_DIGITAL_CLIP1_REG 0x11 +#define TAS5720_MAX_REG TAS5720_DIGITAL_CLIP1_REG + +/* TAS5720_DEVICE_ID_REG */ +#define TAS5720_DEVICE_ID 0x01 + +/* TAS5720_POWER_CTRL_REG */ +#define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) +#define TAS5720_SLEEP BIT(1) +#define TAS5720_SDZ BIT(0) + +/* TAS5720_DIGITAL_CTRL1_REG */ +#define TAS5720_HPF_BYPASS BIT(7) +#define TAS5720_TDM_CFG_SRC BIT(6) +#define TAS5720_SSZ_DS BIT(3) +#define TAS5720_SAIF_RIGHTJ_24BIT (0x0) +#define TAS5720_SAIF_RIGHTJ_20BIT (0x1) +#define TAS5720_SAIF_RIGHTJ_18BIT (0x2) +#define TAS5720_SAIF_RIGHTJ_16BIT (0x3) +#define TAS5720_SAIF_I2S (0x4) +#define TAS5720_SAIF_LEFTJ (0x5) +#define TAS5720_SAIF_FORMAT_MASK GENMASK(2, 0) + +/* TAS5720_DIGITAL_CTRL2_REG */ +#define TAS5720_MUTE BIT(4) +#define TAS5720_TDM_SLOT_SEL_MASK GENMASK(2, 0) + +/* TAS5720_ANALOG_CTRL_REG */ +#define TAS5720_PWM_RATE_6_3_FSYNC (0x0 << 4) +#define TAS5720_PWM_RATE_8_4_FSYNC (0x1 << 4) +#define TAS5720_PWM_RATE_10_5_FSYNC (0x2 << 4) +#define TAS5720_PWM_RATE_12_6_FSYNC (0x3 << 4) +#define TAS5720_PWM_RATE_14_7_FSYNC (0x4 << 4) +#define TAS5720_PWM_RATE_16_8_FSYNC (0x5 << 4) +#define TAS5720_PWM_RATE_20_10_FSYNC (0x6 << 4) +#define TAS5720_PWM_RATE_24_12_FSYNC (0x7 << 4) +#define TAS5720_PWM_RATE_MASK GENMASK(6, 4) +#define TAS5720_ANALOG_GAIN_19_2DBV (0x0 << 2) +#define TAS5720_ANALOG_GAIN_20_7DBV (0x1 << 2) +#define TAS5720_ANALOG_GAIN_23_5DBV (0x2 << 2) +#define TAS5720_ANALOG_GAIN_26_3DBV (0x3 << 2) +#define TAS5720_ANALOG_GAIN_MASK GENMASK(3, 2) +#define TAS5720_ANALOG_GAIN_SHIFT (0x2) + +/* TAS5720_FAULT_REG */ +#define TAS5720_OC_THRESH_100PCT (0x0 << 4) +#define TAS5720_OC_THRESH_75PCT (0x1 << 4) +#define TAS5720_OC_THRESH_50PCT (0x2 << 4) +#define TAS5720_OC_THRESH_25PCT (0x3 << 4) +#define TAS5720_OC_THRESH_MASK GENMASK(5, 4) +#define TAS5720_CLKE BIT(3) +#define TAS5720_OCE BIT(2) +#define TAS5720_DCE BIT(1) +#define TAS5720_OTE BIT(0) +#define TAS5720_FAULT_MASK GENMASK(3, 0) + +/* TAS5720_DIGITAL_CLIP1_REG */ +#define TAS5720_CLIP1_MASK GENMASK(7, 2) +#define TAS5720_CLIP1_SHIFT (0x2) + +#endif /* __TAS5720_H__ */ diff --git a/sound/soc/codecs/tlv320aic32x4-i2c.c b/sound/soc/codecs/tlv320aic32x4-i2c.c new file mode 100644 index 000000000000..59606cf3008f --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-i2c.c @@ -0,0 +1,74 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-i2c.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond <nh6z@nh6z.net> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "tlv320aic32x4.h" + +static int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 8; + config.val_bits = 8; + + regmap = devm_regmap_init_i2c(i2c, &config); + return aic32x4_probe(&i2c->dev, regmap); +} + +static int aic32x4_i2c_remove(struct i2c_client *i2c) +{ + return aic32x4_remove(&i2c->dev); +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_i2c_probe, + .remove = aic32x4_i2c_remove, + .id_table = aic32x4_i2c_id, +}; + +module_i2c_driver(aic32x4_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver I2C"); +MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4-spi.c b/sound/soc/codecs/tlv320aic32x4-spi.c new file mode 100644 index 000000000000..724fcdd491b2 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4-spi.c @@ -0,0 +1,76 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4-spi.c + * + * Copyright 2011 NW Digital Radio + * + * Author: Jeremy McDermond <nh6z@nh6z.net> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/spi/spi.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "tlv320aic32x4.h" + +static int aic32x4_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + struct regmap_config config; + + config = aic32x4_regmap_config; + config.reg_bits = 7; + config.pad_bits = 1; + config.val_bits = 8; + config.read_flag_mask = 0x01; + + regmap = devm_regmap_init_spi(spi, &config); + return aic32x4_probe(&spi->dev, regmap); +} + +static int aic32x4_spi_remove(struct spi_device *spi) +{ + return aic32x4_remove(&spi->dev); +} + +static const struct spi_device_id aic32x4_spi_id[] = { + { "tlv320aic32x4", 0 }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(spi, aic32x4_spi_id); + +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + +static struct spi_driver aic32x4_spi_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, + }, + .probe = aic32x4_spi_probe, + .remove = aic32x4_spi_remove, + .id_table = aic32x4_spi_id, +}; + +module_spi_driver(aic32x4_spi_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC32x4 codec driver SPI"); +MODULE_AUTHOR("Jeremy McDermond <nh6z@nh6z.net>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index f2d3191961e1..85d4978d0384 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -30,7 +30,6 @@ #include <linux/pm.h> #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <linux/i2c.h> #include <linux/cdev.h> #include <linux/slab.h> #include <linux/clk.h> @@ -160,7 +159,10 @@ static const struct aic32x4_rate_divs aic32x4_divs[] = { /* 48k rate */ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, - {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}, + + /* 96k rate */ + {AIC32X4_FREQ_25000000, 96000, 2, 7, 8643, 64, 4, 4, 64, 4, 4, 1}, }; static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { @@ -181,16 +183,71 @@ static const struct snd_kcontrol_new lor_output_mixer_controls[] = { SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), }; -static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +static const char * const resistor_text[] = { + "Off", "10 kOhm", "20 kOhm", "40 kOhm", }; -static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), - SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), - SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +/* Left mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text); + +static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum), +}; +static const struct snd_kcontrol_new cml_to_lmixer_controls[] = { + SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum), +}; +static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum), +}; +static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), +}; + +/* Right mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text); +static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum), +}; +static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = { + SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum), +}; +static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum), +}; +static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -214,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, - &left_input_mixer_controls[0], - ARRAY_SIZE(left_input_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, - &right_input_mixer_controls[0], - ARRAY_SIZE(right_input_mixer_controls)), - SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_rmixer_controls), + SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cmr_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_rmixer_controls), + + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_lmixer_controls), + SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cml_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_lmixer_controls), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), @@ -261,19 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, - /* Left input */ - {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, - {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, - {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, - - {"Left ADC", NULL, "Left Input Mixer"}, - /* Right Input */ - {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, - {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, - {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, - - {"Right ADC", NULL, "Right Input Mixer"}, + {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"}, + {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"}, + {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"}, + + {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"}, + {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"}, + + {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"}, + {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"}, + {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"}, + + {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"}, + {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"}, + + {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"}, + {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"}, + + /* Left Input */ + {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"}, + {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"}, + + {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"}, + {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"}, + {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"}, + + {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"}, + {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"}, + {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"}, + + {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"}, + {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"}, + + {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"}, + {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"}, }; static const struct regmap_range_cfg aic32x4_regmap_pages[] = { @@ -287,14 +427,12 @@ static const struct regmap_range_cfg aic32x4_regmap_pages[] = { }, }; -static const struct regmap_config aic32x4_regmap = { - .reg_bits = 8, - .val_bits = 8, - +const struct regmap_config aic32x4_regmap_config = { .max_register = AIC32X4_RMICPGAVOL, .ranges = aic32x4_regmap_pages, .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), }; +EXPORT_SYMBOL(aic32x4_regmap_config); static inline int aic32x4_get_divs(int mclk, int rate) { @@ -567,7 +705,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_96000 #define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -596,7 +734,7 @@ static struct snd_soc_dai_driver aic32x4_dai = { .symmetric_rates = 1, }; -static int aic32x4_probe(struct snd_soc_codec *codec) +static int aic32x4_codec_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; @@ -655,7 +793,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .probe = aic32x4_probe, + .probe = aic32x4_codec_probe, .set_bias_level = aic32x4_set_bias_level, .suspend_bias_off = true, @@ -777,24 +915,22 @@ error_ldo: return ret; } -static int aic32x4_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +int aic32x4_probe(struct device *dev, struct regmap *regmap) { - struct aic32x4_pdata *pdata = i2c->dev.platform_data; struct aic32x4_priv *aic32x4; - struct device_node *np = i2c->dev.of_node; + struct aic32x4_pdata *pdata = dev->platform_data; + struct device_node *np = dev->of_node; int ret; - aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + aic32x4 = devm_kzalloc(dev, sizeof(struct aic32x4_priv), GFP_KERNEL); if (aic32x4 == NULL) return -ENOMEM; - aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); - if (IS_ERR(aic32x4->regmap)) - return PTR_ERR(aic32x4->regmap); - - i2c_set_clientdata(i2c, aic32x4); + dev_set_drvdata(dev, aic32x4); if (pdata) { aic32x4->power_cfg = pdata->power_cfg; @@ -804,7 +940,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, } else if (np) { ret = aic32x4_parse_dt(aic32x4, np); if (ret) { - dev_err(&i2c->dev, "Failed to parse DT node\n"); + dev_err(dev, "Failed to parse DT node\n"); return ret; } } else { @@ -814,71 +950,48 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk"); + aic32x4->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(aic32x4->mclk)) { - dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + dev_err(dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); return PTR_ERR(aic32x4->mclk); } if (gpio_is_valid(aic32x4->rstn_gpio)) { - ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + ret = devm_gpio_request_one(dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) return ret; } - ret = aic32x4_setup_regulators(&i2c->dev, aic32x4); + ret = aic32x4_setup_regulators(dev, aic32x4); if (ret) { - dev_err(&i2c->dev, "Failed to setup regulators\n"); + dev_err(dev, "Failed to setup regulators\n"); return ret; } - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); if (ret) { - dev_err(&i2c->dev, "Failed to register codec\n"); + dev_err(dev, "Failed to register codec\n"); aic32x4_disable_regulators(aic32x4); return ret; } - i2c_set_clientdata(i2c, aic32x4); - return 0; } +EXPORT_SYMBOL(aic32x4_probe); -static int aic32x4_i2c_remove(struct i2c_client *client) +int aic32x4_remove(struct device *dev) { - struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client); + struct aic32x4_priv *aic32x4 = dev_get_drvdata(dev); aic32x4_disable_regulators(aic32x4); - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(dev); + return 0; } - -static const struct i2c_device_id aic32x4_i2c_id[] = { - { "tlv320aic32x4", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); - -static const struct of_device_id aic32x4_of_id[] = { - { .compatible = "ti,tlv320aic32x4", }, - { /* senitel */ } -}; -MODULE_DEVICE_TABLE(of, aic32x4_of_id); - -static struct i2c_driver aic32x4_i2c_driver = { - .driver = { - .name = "tlv320aic32x4", - .of_match_table = aic32x4_of_id, - }, - .probe = aic32x4_i2c_probe, - .remove = aic32x4_i2c_remove, - .id_table = aic32x4_i2c_id, -}; - -module_i2c_driver(aic32x4_i2c_driver); +EXPORT_SYMBOL(aic32x4_remove); MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index 995f033a855d..a197dd51addc 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -10,6 +10,13 @@ #ifndef _TLV320AIC32X4_H #define _TLV320AIC32X4_H +struct device; +struct regmap_config; + +extern const struct regmap_config aic32x4_regmap_config; +int aic32x4_probe(struct device *dev, struct regmap *regmap); +int aic32x4_remove(struct device *dev); + /* tlv320aic32x4 register space (in decimal to match datasheet) */ #define AIC32X4_PAGE1 128 diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..1f7081043566 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, @@ -983,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i if (mute) { /* Power down drivers and DACs */ hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); } twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a8b3e3f701f9..da60e3fe5ee7 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1955,11 +1955,16 @@ err_adsp2_codec_probe: static int wm5102_codec_remove(struct snd_soc_codec *codec) { struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; wm_adsp2_codec_remove(&priv->core.adsp[0], codec); priv->core.arizona->dapm = NULL; + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + + arizona_free_spk(codec); + return 0; } @@ -2093,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev) static int wm5102_remove(struct platform_device *pdev) { + struct wm5102_priv *wm5102 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&wm5102->core.adsp[0]); + return 0; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 83ba70fe16e6..b5820e4d5471 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2298,6 +2298,8 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + arizona_free_spk(codec); + return 0; } @@ -2435,10 +2437,16 @@ static int wm5110_probe(struct platform_device *pdev) static int wm5110_remove(struct platform_device *pdev) { + struct wm5110_priv *wm5110 = platform_get_drvdata(pdev); + int i; + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp2_remove(&wm5110->core.adsp[i]); + return 0; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 88223608a33f..720a14e0687d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2471,7 +2471,7 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) break; default: dev_warn(codec->dev, "Unknown DSPCLK divisor read back\n"); - dspclk = wm8962->sysclk; + dspclk = wm8962->sysclk_rate; } dev_dbg(codec->dev, "DSPCLK is %dHz, BCLK %d\n", dspclk, wm8962->bclk); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 52d766efe14f..6b0785b5a5c5 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1072,6 +1072,8 @@ static int wm8997_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 012396074a8a..449f66636205 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1324,6 +1324,8 @@ static int wm8998_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_spk(codec); + return 0; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb15e7f0..a07bd7c2c587 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -271,8 +273,11 @@ struct wm_adsp_buffer { __be32 words_written[2]; /* total words written (64 bit) */ }; +struct wm_adsp_compr; + struct wm_adsp_compr_buf { struct wm_adsp *dsp; + struct wm_adsp_compr *compr; struct wm_adsp_buffer_region *regions; u32 host_buf_ptr; @@ -435,6 +440,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +717,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +776,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +795,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +860,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +885,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +968,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -944,6 +1040,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -1032,11 +1135,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { @@ -1564,6 +1662,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp, return alg_region; } +static void wm_adsp_free_alg_regions(struct wm_adsp *dsp) +{ + struct wm_adsp_alg_region *alg_region; + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } +} + static int wm_adsp1_setup_algs(struct wm_adsp *dsp) { struct wmfw_adsp1_id_hdr adsp1_id; @@ -1994,7 +2105,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; unsigned int val; @@ -2074,13 +2184,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + + wm_adsp_free_alg_regions(dsp); break; default: @@ -2222,7 +2327,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; @@ -2240,9 +2344,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + mutex_lock(&dsp->pwr_lock); + if (wm_adsp_fw[dsp->fw].num_caps != 0) ret = wm_adsp_buffer_init(dsp); + mutex_unlock(&dsp->pwr_lock); + break; case SND_SOC_DAPM_PRE_PMD: @@ -2269,13 +2377,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + wm_adsp_free_alg_regions(dsp); if (wm_adsp_fw[dsp->fw].num_caps != 0) wm_adsp_buffer_free(dsp); @@ -2340,6 +2442,54 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + compr->buf->compr = compr; + + return 0; +} + +static void wm_adsp_compr_detach(struct wm_adsp_compr *compr) +{ + if (!compr) + return; + + /* Wake the poll so it can see buffer is no longer attached */ + if (compr->stream) + snd_compr_fragment_elapsed(compr->stream); + + if (wm_adsp_compr_attached(compr)) { + compr->buf->compr = NULL; + compr->buf = NULL; + } +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2393,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream) mutex_lock(&dsp->pwr_lock); + wm_adsp_compr_detach(compr); dsp->compr = NULL; kfree(compr->raw_buf); @@ -2689,6 +2840,8 @@ err_buffer: static int wm_adsp_buffer_free(struct wm_adsp *dsp) { if (dsp->buffer) { + wm_adsp_compr_detach(dsp->buffer->compr); + kfree(dsp->buffer->regions); kfree(dsp->buffer); @@ -2698,25 +2851,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } -static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) -{ - return compr->buf != NULL; -} - -static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) -{ - /* - * Note this will be more complex once each DSP can support multiple - * streams - */ - if (!compr->dsp->buffer) - return -EINVAL; - - compr->buf = compr->dsp->buffer; - - return 0; -} - int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) { struct wm_adsp_compr *compr = stream->runtime->private_data; @@ -2805,21 +2939,41 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) avail += wm_adsp_buffer_size(buf); adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail); + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; return 0; } +static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf) +{ + int ret; + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); + if (ret < 0) { + adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret); + return ret; + } + if (buf->error != 0) { + adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error); + return -EIO; + } + + return 0; +} + int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) { - struct wm_adsp_compr_buf *buf = dsp->buffer; - struct wm_adsp_compr *compr = dsp->compr; + struct wm_adsp_compr_buf *buf; + struct wm_adsp_compr *compr; int ret = 0; mutex_lock(&dsp->pwr_lock); + buf = dsp->buffer; + compr = dsp->compr; + if (!buf) { ret = -ENODEV; goto out; @@ -2827,16 +2981,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) adsp_dbg(dsp, "Handling buffer IRQ\n"); - ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); - if (ret < 0) { - adsp_err(dsp, "Failed to check buffer error: %d\n", ret); - goto out; - } - if (buf->error != 0) { - adsp_err(dsp, "Buffer error occurred: %d\n", buf->error); - ret = -EIO; - goto out; - } + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) + goto out_notify; /* Wake poll to report error */ ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); @@ -2851,6 +2998,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) goto out; } +out_notify: if (compr && compr->stream) snd_compr_fragment_elapsed(compr->stream); @@ -2879,14 +3027,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { struct wm_adsp_compr *compr = stream->runtime->private_data; - struct wm_adsp_compr_buf *buf = compr->buf; struct wm_adsp *dsp = compr->dsp; + struct wm_adsp_compr_buf *buf; int ret = 0; adsp_dbg(dsp, "Pointer request\n"); mutex_lock(&dsp->pwr_lock); + buf = compr->buf; + if (!compr->buf) { ret = -ENXIO; goto out; @@ -2909,6 +3059,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, * DSP to inform us once a whole fragment is available. */ if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) + goto out; + ret = wm_adsp_buffer_reenable_irq(buf); if (ret < 0) { adsp_err(dsp, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57e600f..feb61e2c4bb4 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50ca291cc225..6b732d8e5896 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -16,7 +16,11 @@ config SND_EDMA_SOC - DRA7xx family config SND_DAVINCI_SOC_I2S - tristate + tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support" + depends on SND_EDMA_SOC + help + Say Y or M here if you want to have support for McBSP IP found in + Texas Instruments DaVinci DA850 SoCs. config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ec98548a5fc9..384961651904 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -4,9 +4,15 @@ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> * + * DT support (c) 2016 Petr Kulhavy, Barix AG <petr@barix.com> + * based on davinci-mcasp.c DT support + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. + * + * TODO: + * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers */ #include <linux/init.h> @@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { + struct snd_dmaengine_dai_dma_data *dma_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *res; void __iomem *io_base; int *dma; int ret; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!mem) { + dev_warn(&pdev->dev, + "\"mpu\" mem resource not found, using index 0\n"); + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + } + io_base = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(io_base)) return PTR_ERR(io_base); @@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return -ENODEV; - clk_enable(dev->clk); - dev->base = io_base; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); + /* setup DMA, first TX, then RX */ + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); - - /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "tx"; + } else { + dev_err(&pdev->dev, "Missing DMA tx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; + + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "rx"; + } else { + dev_err(&pdev->dev, "Missing DMA rx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return -ENODEV; + clk_enable(dev->clk); dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id davinci_i2s_match[] = { + { .compatible = "ti,da850-mcbsp" }, + {}, +}; +MODULE_DEVICE_TABLE(of, davinci_i2s_match); + static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", + .of_match_table = of_match_ptr(davinci_i2s_match), }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e1324989bd6b..0f66fda2c772 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: @@ -540,21 +540,19 @@ out: return ret; } -static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, +static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, int div, bool explicit) { - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - pm_runtime_get_sync(mcasp->dev); switch (div_id) { - case 0: /* MCLK divider */ + case MCASP_CLKDIV_AUXCLK: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; - case 1: /* BCLK divider */ + case MCASP_CLKDIV_BCLK: /* BCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, @@ -563,7 +561,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* + case MCASP_CLKDIV_BCLK_FS_RATIO: + /* * BCLK/LRCLK ratio descries how many bit-clock cycles * fit into one frame. The clock ratio is given for a * full period of data (for I2S format both left and @@ -591,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + + return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1); } static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -999,27 +1000,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, } static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, - unsigned int bclk_freq, - int *error_ppm) + unsigned int bclk_freq, bool set) { - int div = mcasp->sysclk_freq / bclk_freq; - int rem = mcasp->sysclk_freq % bclk_freq; + int error_ppm; + unsigned int sysclk_freq = mcasp->sysclk_freq; + u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG); + int div = sysclk_freq / bclk_freq; + int rem = sysclk_freq % bclk_freq; + int aux_div = 1; + + if (div > (ACLKXDIV_MASK + 1)) { + if (reg & AHCLKXE) { + aux_div = div / (ACLKXDIV_MASK + 1); + if (div % (ACLKXDIV_MASK + 1)) + aux_div++; + + sysclk_freq /= aux_div; + div = sysclk_freq / bclk_freq; + rem = sysclk_freq % bclk_freq; + } else if (set) { + dev_warn(mcasp->dev, "Too fast reference clock (%u)\n", + sysclk_freq); + } + } if (rem != 0) { if (div == 0 || - ((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + ((sysclk_freq / div) - bclk_freq) > + (bclk_freq - (sysclk_freq / (div+1)))) { div++; rem = rem - bclk_freq; } } - if (error_ppm) - *error_ppm = - (div*1000000 + (int)div64_long(1000000LL*rem, - (int)bclk_freq)) - /div - 1000000; + error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) / div - 1000000; + + if (set) { + if (error_ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + error_ppm); + + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0); + if (reg & AHCLKXE) + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK, + aux_div, 0); + } - return div; + return error_ppm; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -1044,18 +1071,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); - int ppm, div; if (mcasp->slot_width) sbits = mcasp->slot_width; - div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots, - &ppm); - if (ppm) - dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", - ppm); - - __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); + davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1166,7 +1186,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, davinci_mcasp_dai_rates[i]; int ppm; - davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { if (range.empty) { range.min = davinci_mcasp_dai_rates[i]; @@ -1205,8 +1226,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, if (rd->mcasp->slot_width) sbits = rd->mcasp->slot_width; - davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate, - &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, + sbits * slots * rate, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { snd_mask_set(&nfmt, i); count++; @@ -1230,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a3be108a8c17..1e8787fb3fb7 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -306,4 +306,9 @@ #define NUMEVT(x) (((x) & 0xFF) << 8) #define NUMDMA_MASK (0xFF) +/* clock divider IDs */ +#define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */ +#define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */ +#define MCASP_CLKDIV_BCLK_FS_RATIO 2 /* to set BCLK FS ration */ + #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bff258d7bcea..0db69b7e9617 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -100,6 +100,7 @@ struct dw_i2s_dev { struct device *dev; u32 ccr; u32 xfer_resolution; + u32 fifo_th; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; @@ -147,17 +148,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { + struct i2s_clk_config_data *config = &dev->config; u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); } i2s_write_reg(dev->i2s_base, ITER, 1); } else { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); } @@ -231,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); } else { i2s_write_reg(dev->i2s_base, RCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); @@ -498,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, */ u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1); u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2); + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); u32 idx; if (dev->capability & DWC_I2S_RECORD && @@ -536,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, dev->capability |= DW_I2S_SLAVE; } + dev->fifo_th = fifo_depth / 2; return 0; } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df771e3b..2147994ab46f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> #include "fsl_sai.h" #include "imx-pcm.h" @@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + int index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + return PTR_ERR(gpr); + } + + index = of_alias_get_id(np, "sai"); + if (index < 0) + return index; + + regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index), + MCLK_DIR(index)); + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; @@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de1035cda..632ecc0e3956 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; @@ -261,6 +262,7 @@ struct fsl_ssi_private { struct fsl_ssi_dbg dbg_stats; const struct fsl_ssi_soc_data *soc; + struct device *dev; }; /* @@ -400,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private, } /* + * Clear RX or TX FIFO to remove samples from the previous + * stream session which may be still present in the FIFO and + * may introduce bad samples and/or channel slipping. + * + * Note: The SOR is not documented in recent IMX datasheet, but + * is described in IMX51 reference manual at section 56.3.3.15. + */ +static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private, + bool is_rx) +{ + if (is_rx) { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR); + } else { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR); + } +} + +/* * Calculate the bits that have to be disabled for the current stream that is * getting disabled. This keeps the bits enabled that are necessary for the * second stream to work if 'stream_active' is true. @@ -474,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, * (online configuration) */ if (enable) { - regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); + fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE); + regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr); + regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); } else { u32 sier; u32 srcr; @@ -506,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, config_done: /* Enabling of subunits is done after configuration */ - if (enable) + if (enable) { + if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) { + /* + * Be sure the Tx FIFO is filled when TE is set. + * Otherwise, there are some chances to start the + * playback with some void samples inserted first, + * generating a channel slip. + * + * First, SSIEN must be set, to let the FIFO be filled. + * + * Notes: + * - Limit this fix to the DMA case until FIQ cases can + * be tested. + * - Limit the length of the busy loop to not lock the + * system too long, even if 1-2 loops are sufficient + * in general. + */ + int i; + int max_loop = 100; + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN); + for (i = 0; i < max_loop; i++) { + u32 sfcsr; + regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr); + if (CCSR_SSI_SFCSR_TFCNT0(sfcsr)) + break; + } + if (i == max_loop) { + dev_err(ssi_private->dev, + "Timeout waiting TX FIFO filling\n"); + } + } regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr); + } } @@ -670,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (IS_ERR(ssi_private->baudclk)) return -EINVAL; + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (freq * 5 > clk_get_rate(ssi_private->clk)) { + dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n"); + return -EINVAL; + } + baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream)); /* It should be already enough to divide clock by setting pm alone */ @@ -686,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - /* - * Hardware limitation: The bclk rate must be - * never greater than 1/5 IPG clock rate - */ - if (clkrate * 5 > clk_get_rate(ssi_private->clk)) - continue; - clkrate /= factor; afreq = clkrate / (i + 1); @@ -1158,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, @@ -1402,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->soc = of_id->data; + ssi_private->dev = &pdev->dev; sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index e63cd5ecfd8f..dac6688540dc 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret, runtime->dma_area, &runtime->dma_addr, runtime->dma_bytes); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2389ab47e25f..466492b7d4f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -643,6 +643,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c2300457..91c15abb625e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BXT_RT298_MACH + tristate "ASoC Audio driver for Broxton with RT298 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT298 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Broxton platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C @@ -162,8 +177,8 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE + select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY - select SND_HDA_I915 select SND_SOC_INTEL_SST config SND_SOC_INTEL_SKL_RT286_MACH diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index b97e6adcf1b2..98720a93de8a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol if (e->w && e->w->power) ret = sst_send_slot_map(drv); - else + else if (!e->w) dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", kcontrol->id.name); return ret; diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3310c0f9c356..a8506774f510 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3f8a1e10bed0..7486a0022fde 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c new file mode 100644 index 000000000000..f4787515c0ed --- /dev/null +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -0,0 +1,353 @@ +/* + * Intel Broxton-P I2S Machine Driver + * + * Copyright (C) 2014-2016, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Skylake I2S Machine driver + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt298.h" + +static struct snd_soc_jack broxton_headset; +/* Headset jack detection DAPM pins */ + +enum { + BXT_DPCM_AUDIO_PB = 0, + BXT_DPCM_AUDIO_CP, + BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_HDMI1_PB, + BXT_DPCM_AUDIO_HDMI2_PB, + BXT_DPCM_AUDIO_HDMI3_PB, +}; + +static struct snd_soc_jack_pin broxton_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broxton_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget broxton_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), +}; + +static const struct snd_soc_dapm_route broxton_rt298_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMic", NULL, "SoC DMIC"}, + + {"HDMI1", NULL, "hif5 Output"}, + {"HDMI2", NULL, "hif6 Output"}, + {"HDMI3", NULL, "hif7 Output"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp5 Tx"}, + { "ssp5 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp5 Rx" }, + { "ssp5 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, + +}; + +static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &broxton_headset, + broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins)); + + if (ret) + return ret; + + rt298_mic_detect(codec, &broxton_headset); + return 0; +} + +static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->codec_dai; + + return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id); +} + +static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP5 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int broxton_rt298_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broxton_rt298_ops = { + .hw_params = broxton_rt298_hw_params, +}; + +/* broxton digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broxton_rt298_dais[] = { + /* Front End DAI links */ + [BXT_DPCM_AUDIO_PB] + { + .name = "Bxt Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [BXT_DPCM_AUDIO_CP] + { + .name = "Bxt Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [BXT_DPCM_AUDIO_REF_CP] + { + .name = "Bxt Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI1_PB] + { + .name = "Bxt HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI2_PB] + { + .name = "Bxt HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI3_PB] + { + .name = "Bxt HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP5 - Codec */ + .name = "SSP5-Codec", + .id = 0, + .cpu_dai_name = "SSP5 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt298-aif1", + .init = broxton_rt298_codec_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broxton_ssp5_fixup, + .ops = &broxton_rt298_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* broxton audio machine driver for SPT + RT298S */ +static struct snd_soc_card broxton_rt298 = { + .name = "broxton-rt298", + .owner = THIS_MODULE, + .dai_link = broxton_rt298_dais, + .num_links = ARRAY_SIZE(broxton_rt298_dais), + .controls = broxton_controls, + .num_controls = ARRAY_SIZE(broxton_controls), + .dapm_widgets = broxton_widgets, + .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), + .dapm_routes = broxton_rt298_map, + .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), + .fully_routed = true, +}; + +static int broxton_audio_probe(struct platform_device *pdev) +{ + broxton_rt298.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298); +} + +static struct platform_driver broxton_audio = { + .probe = broxton_audio_probe, + .driver = { + .name = "bxt_alc298s_i2s", + }, +}; +module_platform_driver(broxton_audio) + +/* Module information */ +MODULE_AUTHOR("Ramesh Babu <Ramesh.Babu@intel.com>"); +MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Broxton"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bxt_alc298s_i2s"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e753f0b..88efb62439ba 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1c95ccc886c4..35f591eab3c9 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index e609f089593a..6260df6bd49c 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2a6f80843bc9..0618a7f1025b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2e5347f8f96c..df9d254baa18 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..863f1d5e2a2c 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b79a18d..d2808652b974 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau8825_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -391,7 +434,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, @@ -456,7 +498,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -472,7 +514,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -489,7 +531,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -501,7 +543,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -512,7 +554,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -523,7 +565,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -534,6 +576,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825max", @@ -547,11 +604,21 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau8825_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99ae9b0..e19aa99c4f72 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -34,6 +34,15 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau88125_private { + struct list_head hdmi_pcm_list; +}; enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -440,7 +482,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, @@ -505,7 +546,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -523,7 +564,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -540,7 +581,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -552,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -563,7 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -574,7 +615,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -585,6 +626,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", @@ -600,11 +656,21 @@ static struct snd_soc_card skylake_audio_card = { .codec_conf = ssm4567_codec_conf, .num_configs = ARRAY_SIZE(ssm4567_codec_conf), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau88125_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a8e75..426b48233fdb 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_rt286_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id); + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static unsigned int rates[] = { @@ -317,7 +338,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, }, @@ -375,7 +395,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -393,7 +413,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "dmic01", - .be_id = 1, + .id = 1, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -405,7 +425,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp1", - .be_id = 2, + .id = 2, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -416,7 +436,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp2", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -427,7 +447,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp3", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -438,6 +458,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + RT286S */ static struct snd_soc_card skylake_rt286 = { .name = "skylake-rt286", @@ -451,11 +486,21 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_rt286_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_rt286.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_rt286, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); } diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 4dcfb7e5ed70..8398cb227ba9 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include <linux/kconfig.h> +#include <linux/stddef.h> #include <linux/acpi.h> /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index ac60f1301e21..91565229d074 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -1345,7 +1345,7 @@ int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; /* wait for pause to complete before we reset the stream */ - while (stream->running && tries--) + while (stream->running && --tries) msleep(1); if (!tries) { dev_err(hsw->dev, "error: reset stream %d still running\n", diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c7e09b..994256b39b9c 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index 914b6dab9bea..c28f5d0e1d99 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o + skl-sst.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c new file mode 100644 index 000000000000..965ce40ce752 --- /dev/null +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -0,0 +1,328 @@ +/* + * bxt-sst.c - DSP library functions for BXT platform + * + * Copyright (C) 2015-16 Intel Corp + * Author:Rafal Redzimski <rafal.f.redzimski@intel.com> + * Jeeja KP <jeeja.kp@intel.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/firmware.h> +#include <linux/device.h> + +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-sst-ipc.h" + +#define BXT_BASEFW_TIMEOUT 3000 +#define BXT_INIT_TIMEOUT 500 +#define BXT_IPC_PURGE_FW 0x01004000 + +#define BXT_ROM_INIT 0x5 +#define BXT_ADSP_SRAM0_BASE 0x80000 + +/* Firmware status window */ +#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE +#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4) + +#define BXT_ADSP_SRAM1_BASE 0xA0000 + +static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) +{ + return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); +} + +static int sst_bxt_prepare_fw(struct sst_dsp *ctx, + const void *fwdata, u32 fwsize) +{ + int stream_tag, ret, i; + u32 reg; + + stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab); + if (stream_tag < 0) { + dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n", + stream_tag); + return stream_tag; + } + + ctx->dsp_ops.stream_tag = stream_tag; + memcpy(ctx->dmab.area, fwdata, fwsize); + + /* Purge FW request */ + sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY | + BXT_IPC_PURGE_FW | (stream_tag - 1)); + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret); + ret = -EIO; + goto base_fw_load_failed; + } + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE); + + if (reg & SKL_ADSP_REG_HIPCIE_DONE) { + sst_dsp_shim_update_bits_forced(ctx, + SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + break; + } + mdelay(1); + } + if (!i) { + dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + } + + /* enable Interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + if (SKL_FW_INIT == + (sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) & + SKL_FW_STS_MASK)) { + + dev_info(ctx->dev, "ROM loaded, continue FW loading\n"); + break; + } + mdelay(1); + } + if (!i) { + dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg); + ret = -EIO; + goto base_fw_load_failed; + } + + return ret; + +base_fw_load_failed: + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); + skl_dsp_disable_core(ctx); + return ret; +} + +static int sst_transfer_fw_host_dma(struct sst_dsp *ctx) +{ + int ret; + + ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag); + ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK, + BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot"); + + ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag); + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag); + + return ret; +} + +static int bxt_load_base_firmware(struct sst_dsp *ctx) +{ + const struct firmware *fw = NULL; + struct skl_sst *skl = ctx->thread_context; + int ret; + + ret = request_firmware(&fw, ctx->fw_name, ctx->dev); + if (ret < 0) { + dev_err(ctx->dev, "Request firmware failed %d\n", ret); + goto sst_load_base_firmware_failed; + } + + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + /* Retry Enabling core and ROM load. Retry seemed to help */ + if (ret < 0) { + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + if (ret < 0) { + dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret); + goto sst_load_base_firmware_failed; + } + } + + ret = sst_transfer_fw_host_dma(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Transfer firmware failed %d\n", ret); + dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + + skl_dsp_disable_core(ctx); + } else { + dev_dbg(ctx->dev, "Firmware download successful\n"); + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n"); + skl_dsp_disable_core(ctx); + ret = -EIO; + } else { + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + ret = 0; + } + } + +sst_load_base_firmware_failed: + release_firmware(fw); + return ret; +} + +static int bxt_set_dsp_D0(struct sst_dsp *ctx) +{ + struct skl_sst *skl = ctx->thread_context; + int ret; + + skl->boot_complete = false; + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret); + return ret; + } + + /* enable interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "ipc: error DSP boot timeout\n"); + dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + return -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + return 0; +} + +static int bxt_set_dsp_D3(struct sst_dsp *ctx) +{ + struct skl_ipc_dxstate_info dx; + struct skl_sst *skl = ctx->thread_context; + int ret = 0; + + if (!is_skl_dsp_running(ctx)) + return ret; + + dx.core_mask = SKL_DSP_CORE0_MASK; + dx.dx_mask = SKL_IPC_D3_MASK; + + ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, + SKL_BASE_FW_MODULE_ID, &dx); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret); + return ret; + } + + ret = skl_dsp_disable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret); + ret = -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RESET); + return 0; +} + +static struct skl_dsp_fw_ops bxt_fw_ops = { + .set_state_D0 = bxt_set_dsp_D0, + .set_state_D3 = bxt_set_dsp_D3, + .load_fw = bxt_load_base_firmware, + .get_fw_errcode = bxt_get_errorcode, +}; + +static struct sst_ops skl_ops = { + .irq_handler = skl_dsp_sst_interrupt, + .write = sst_shim32_write, + .read = sst_shim32_read, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .free = skl_dsp_free, +}; + +static struct sst_dsp_device skl_dev = { + .thread = skl_dsp_irq_thread_handler, + .ops = &skl_ops, +}; + +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp) +{ + struct skl_sst *skl; + struct sst_dsp *sst; + int ret; + + skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL); + if (skl == NULL) + return -ENOMEM; + + skl->dev = dev; + skl_dev.thread_context = skl; + + skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq); + if (!skl->dsp) { + dev_err(skl->dev, "skl_dsp_ctx_init failed\n"); + return -ENODEV; + } + + sst = skl->dsp; + sst->fw_name = fw_name; + sst->dsp_ops = dsp_ops; + sst->fw_ops = bxt_fw_ops; + sst->addr.lpe = mmio_base; + sst->addr.shim = mmio_base; + + sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), + SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + + ret = skl_ipc_init(dev, skl); + if (ret) + return ret; + + skl->boot_complete = false; + init_waitqueue_head(&skl->boot_wait); + + ret = sst->fw_ops.load_fw(sst); + if (ret < 0) { + dev_err(dev, "Load base fw failed: %x", ret); + return ret; + } + + if (dsp) + *dsp = skl; + + return 0; +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); + + +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) +{ + skl_ipc_free(&ctx->ipc); + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + + if (ctx->dsp->addr.lpe) + iounmap(ctx->dsp->addr.lpe); + + ctx->dsp->ops->free(ctx->dsp); +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Broxton IPC driver"); diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 79c5089b85d6..226db84ba20f 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask); } +static int skl_dsp_setup_spib(struct device *dev, unsigned int size, + int stream_tag, int enable) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_stream *stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + struct hdac_ext_stream *estream; + + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + /* enable/disable SPIB for this hdac stream */ + snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + + /* set the spib value */ + snd_hdac_ext_stream_set_spib(ebus, estream, size); + + return 0; +} + +static int skl_dsp_prepare(struct device *dev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_stream *estream; + struct hdac_stream *stream; + struct snd_pcm_substream substream; + int ret; + + if (!bus) + return -ENODEV; + + memset(&substream, 0, sizeof(substream)); + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + + estream = snd_hdac_ext_stream_assign(ebus, &substream, + HDAC_EXT_STREAM_TYPE_HOST); + if (!estream) + return -ENODEV; + + stream = hdac_stream(estream); + + /* assign decouple host dma channel */ + ret = snd_hdac_dsp_prepare(stream, format, size, dmab); + if (ret < 0) + return ret; + + skl_dsp_setup_spib(dev, size, stream->stream_tag, true); + + return stream->stream_tag; +} + +static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + snd_hdac_dsp_trigger(stream, start); + + return 0; +} + +static int skl_dsp_cleanup(struct device *dev, + struct snd_dma_buffer *dmab, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_ext_stream *estream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + skl_dsp_setup_spib(dev, 0, stream_tag, false); + snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST); + + snd_hdac_dsp_cleanup(stream, dmab); + + return 0; +} + static struct skl_dsp_loader_ops skl_get_loader_ops(void) { struct skl_dsp_loader_ops loader_ops; @@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void) return loader_ops; }; +static struct skl_dsp_loader_ops bxt_get_loader_ops(void) +{ + struct skl_dsp_loader_ops loader_ops; + + memset(&loader_ops, 0, sizeof(loader_ops)); + + loader_ops.alloc_dma_buf = skl_alloc_dma_buf; + loader_ops.free_dma_buf = skl_free_dma_buf; + loader_ops.prepare = skl_dsp_prepare; + loader_ops.trigger = skl_dsp_trigger; + loader_ops.cleanup = skl_dsp_cleanup; + + return loader_ops; +}; + static const struct skl_dsp_ops dsp_ops[] = { { .id = 0x9d70, @@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = { .init = skl_sst_dsp_init, .cleanup = skl_sst_dsp_cleanup }, + { + .id = 0x5a98, + .loader_ops = bxt_get_loader_ops, + .init = bxt_sst_dsp_init, + .cleanup = bxt_sst_dsp_cleanup + }, }; static int skl_get_dsp_ops(int pci_id) @@ -744,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx, return ret; } mconfig->m_state = SKL_MODULE_INIT_DONE; - + kfree(param_data); return ret; } diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 14d1916ea9f8..7d73648e5f9a 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void *skl_nhlt_init(struct device *dev) +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; struct nhlt_resource_desc *nhlt_ptr = NULL; + struct nhlt_acpi_table *nhlt_table = NULL; if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) { dev_err(dev, "Requested NHLT device not found\n"); @@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev) obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - - return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + nhlt_table = (struct nhlt_acpi_table *) + memremap(nhlt_ptr->min_addr, nhlt_ptr->length, MEMREMAP_WB); + ACPI_FREE(obj); + return nhlt_table; } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void *addr) +void skl_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap(addr); + memunmap((void *) nhlt); } static struct nhlt_specific_cfg *skl_get_specific_cfg( @@ -120,7 +123,7 @@ struct nhlt_specific_cfg struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_acpi_table *nhlt = skl->nhlt; u16 bps = (s_fmt == 16) ? 16 : 32; u8 j; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index dab0900eef26..7c81b31748ff 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { .rate_min = 8000, .rate_max = 48000, .channels_min = 1, - .channels_max = HDA_QUAD, + .channels_max = 8, .buffer_bytes_max = AZX_MAX_BUF_SIZE, .period_bytes_min = 128, .period_bytes_max = AZX_MAX_BUF_SIZE / 2, @@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream, struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - if ((dai->playback_active > 1) || (dai->capture_active > 1)) + if (dai->playback_widget->power || dai->capture_widget->power) return 0; mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream); @@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl_module_cfg *mconfig; struct hdac_ext_bus *ebus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct snd_soc_dapm_widget *w; int ret; mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); if (!mconfig) return -EIO; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - skl_pcm_prepare(substream, dai); - /* - * enable DMA Resume enable bit for the stream, set the dpib - * & lpib position to resune before starting the DMA - */ - snd_hdac_ext_stream_drsm_enable(ebus, true, - hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib); - snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + if (!w->ignore_suspend) { + skl_pcm_prepare(substream, dai); + /* + * enable DMA Resume enable bit for the stream, set the + * dpib & lpib position to resume before starting the + * DMA + */ + snd_hdac_ext_stream_drsm_enable(ebus, true, + hdac_stream(stream)->index); + snd_hdac_ext_stream_set_dpibr(ebus, stream, + stream->dpib); + snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + } case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return ret; ret = skl_decoupled_trigger(substream, cmd); - if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) { + if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ stream->dpib = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + @@ -523,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, if (!link) return -EINVAL; - snd_hdac_ext_bus_link_power_up(link); snd_hdac_ext_link_stream_reset(link_dev); snd_hdac_ext_link_stream_setup(link_dev, format_val); @@ -682,7 +691,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI1 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -697,7 +706,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI2 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -712,7 +721,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI3 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -760,12 +769,84 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { + .name = "SSP2 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp2 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp2 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP3 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp3 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp3 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP4 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp4 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp4 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP5 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp5 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp5 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ .name = "iDisp1 Pin", .ops = &skl_link_dai_ops, .playback = { .stream_name = "iDisp1 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE, @@ -777,7 +858,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp2 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | @@ -790,7 +871,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp3 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index a5267e8a96e0..13c19855ee1a 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -336,6 +336,9 @@ void skl_dsp_free(struct sst_dsp *dsp) skl_ipc_int_disable(dsp); free_irq(dsp->irq, dsp); + skl_ipc_op_int_disable(dsp); + skl_ipc_int_disable(dsp); + skl_dsp_disable_core(dsp); } EXPORT_SYMBOL_GPL(skl_dsp_free); diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index b6e310d49dd6..deabe7308d3b 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -118,16 +118,25 @@ struct skl_dsp_fw_ops { int (*set_state_D0)(struct sst_dsp *ctx); int (*set_state_D3)(struct sst_dsp *ctx); unsigned int (*get_fw_errcode)(struct sst_dsp *ctx); - int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name); + int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name); int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id); }; struct skl_dsp_loader_ops { + int stream_tag; + int (*alloc_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab, size_t size); int (*free_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab); + int (*prepare)(struct device *dev, unsigned int format, + unsigned int byte_size, + struct snd_dma_buffer *bufp); + int (*trigger)(struct device *dev, bool start, int stream_tag); + + int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab, + int stream_tag); }; struct skl_load_module_info { @@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx); int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp); +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp); void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); #endif /*__SKL_SST_DSP_H__*/ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 348a734f8e24..13ec8d53b526 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -20,6 +20,7 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> +#include <linux/uuid.h> #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" @@ -304,14 +305,16 @@ static int skl_transfer_module(struct sst_dsp *ctx, return ret; } -static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) +static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) { struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ + uuid_le *uuid_mod; - snprintf(mod_name, sizeof(mod_name), "%s%s%s", - "intel/dsp_fw_", guid, ".bin"); + uuid_mod = (uuid_le *)guid; + snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", + "intel/dsp_fw_", uuid_mod, ".bin"); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { @@ -451,6 +454,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) skl_clear_module_table(ctx->dsp); skl_ipc_free(&ctx->ipc); ctx->dsp->ops->free(ctx->dsp); + if (ctx->boot_complete) { + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + skl_cldma_int_disable(ctx->dsp); + } } EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e77b8aa..3e036b0349b9 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx, dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg); } +static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs) +{ + int slot_map = 0xFFFFFFFF; + int start_slot = 0; + int i; + + for (i = 0; i < chs; i++) { + /* + * For 2 channels with starting slot as 0, slot map will + * look like 0xFFFFFF10. + */ + slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i))); + start_slot++; + } + fmt->ch_map = slot_map; +} + static void skl_tplg_update_params(struct skl_module_fmt *fmt, struct skl_pipe_params *params, int fixup) { if (fixup & SKL_RATE_FIXUP_MASK) fmt->s_freq = params->s_freq; - if (fixup & SKL_CH_FIXUP_MASK) + if (fixup & SKL_CH_FIXUP_MASK) { fmt->channels = params->ch; + skl_tplg_update_chmap(fmt, fmt->channels); + } if (fixup & SKL_FMT_FIXUP_MASK) { fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt); @@ -239,6 +258,7 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, { int multiplier = 1; struct skl_module_fmt *in_fmt, *out_fmt; + int in_rate, out_rate; /* Since fixups is applied to pin 0 only, ibs, obs needs @@ -249,15 +269,24 @@ static void skl_tplg_update_buffer_size(struct skl_sst *ctx, if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) multiplier = 5; - mcfg->ibs = (in_fmt->s_freq / 1000) * - (mcfg->in_fmt->channels) * - (mcfg->in_fmt->bit_depth >> 3) * - multiplier; - - mcfg->obs = (mcfg->out_fmt->s_freq / 1000) * - (mcfg->out_fmt->channels) * - (mcfg->out_fmt->bit_depth >> 3) * - multiplier; + + if (in_fmt->s_freq % 1000) + in_rate = (in_fmt->s_freq / 1000) + 1; + else + in_rate = (in_fmt->s_freq / 1000); + + mcfg->ibs = in_rate * (mcfg->in_fmt->channels) * + (mcfg->in_fmt->bit_depth >> 3) * + multiplier; + + if (mcfg->out_fmt->s_freq % 1000) + out_rate = (mcfg->out_fmt->s_freq / 1000) + 1; + else + out_rate = (mcfg->out_fmt->s_freq / 1000); + + mcfg->obs = out_rate * (mcfg->out_fmt->channels) * + (mcfg->out_fmt->bit_depth >> 3) * + multiplier; } static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, @@ -485,11 +514,15 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) if (!skl_is_pipe_mcps_avail(skl, mconfig)) return -ENOMEM; + skl_tplg_alloc_pipe_mcps(skl, mconfig); + if (mconfig->is_loadable && ctx->dsp->fw_ops.load_mod) { ret = ctx->dsp->fw_ops.load_mod(ctx->dsp, mconfig->id.module_id, mconfig->guid); if (ret < 0) return ret; + + mconfig->m_state = SKL_MODULE_LOADED; } /* update blob if blob is null for be with default value */ @@ -509,7 +542,6 @@ skl_tplg_init_pipe_modules(struct skl *skl, struct skl_pipe *pipe) ret = skl_tplg_set_module_params(w, ctx); if (ret < 0) return ret; - skl_tplg_alloc_pipe_mcps(skl, mconfig); } return 0; @@ -524,7 +556,8 @@ static int skl_tplg_unload_pipe_modules(struct skl_sst *ctx, list_for_each_entry(w_module, &pipe->w_list, node) { mconfig = w_module->w->priv; - if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod) + if (mconfig->is_loadable && ctx->dsp->fw_ops.unload_mod && + mconfig->m_state > SKL_MODULE_UNINIT) return ctx->dsp->fw_ops.unload_mod(ctx->dsp, mconfig->id.module_id); } @@ -558,6 +591,9 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, if (!skl_is_pipe_mem_avail(skl, mconfig)) return -ENOMEM; + skl_tplg_alloc_pipe_mem(skl, mconfig); + skl_tplg_alloc_pipe_mcps(skl, mconfig); + /* * Create a list of modules for pipe. * This list contains modules from source to sink @@ -601,9 +637,6 @@ static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, src_module = dst_module; } - skl_tplg_alloc_pipe_mem(skl, mconfig); - skl_tplg_alloc_pipe_mcps(skl, mconfig); - return 0; } @@ -1550,6 +1583,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, return -ENOMEM; w->priv = mconfig; + memcpy(&mconfig->guid, &dfw_config->uuid, 16); + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; @@ -1579,10 +1614,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->time_slot = dfw_config->time_slot; mconfig->formats_config.caps_size = dfw_config->caps.caps_size; - if (dfw_config->is_loadable) - memcpy(mconfig->guid, dfw_config->uuid, - ARRAY_SIZE(dfw_config->uuid)); - mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) * sizeof(*mconfig->m_in_pin), GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index de3c401284d9..e4b399cd7868 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -274,14 +274,14 @@ struct skl_pipe { enum skl_module_state { SKL_MODULE_UNINIT = 0, - SKL_MODULE_INIT_DONE = 1, - SKL_MODULE_LOADED = 2, - SKL_MODULE_UNLOADED = 3, - SKL_MODULE_BIND_DONE = 4 + SKL_MODULE_LOADED = 1, + SKL_MODULE_INIT_DONE = 2, + SKL_MODULE_BIND_DONE = 3, + SKL_MODULE_UNLOADED = 4, }; struct skl_module_cfg { - char guid[SKL_UUID_STR_SZ]; + u8 guid[16]; struct skl_module_inst_id id; u8 domain; bool homogenous_inputs; diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h index 1db88a63ac17..a32e5e9cc530 100644 --- a/sound/soc/intel/skylake/skl-tplg-interface.h +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -181,7 +181,7 @@ struct skl_dfw_pipe { } __packed; struct skl_dfw_module { - char uuid[SKL_UUID_STR_SZ]; + u8 uuid[16]; u16 module_id; u16 instance_id; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ab5e25aaeee3..06d8c263c68f 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -222,20 +222,36 @@ static int skl_suspend(struct device *dev) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); + int ret = 0; /* * Do not suspend if streams which are marked ignore suspend are * running, we need to save the state for these and continue */ if (skl->supend_active) { + /* turn off the links and stop the CORB/RIRB DMA if it is On */ snd_hdac_ext_bus_link_power_down_all(ebus); + + if (ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); + enable_irq_wake(bus->irq); pci_save_state(pci); pci_disable_device(pci); - return 0; } else { - return _skl_suspend(ebus); + ret = _skl_suspend(ebus); + if (ret < 0) + return ret; + } + + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { + ret = snd_hdac_display_power(bus, false); + if (ret < 0) + dev_err(bus->dev, + "Cannot turn OFF display power on i915\n"); } + + return ret; } static int skl_resume(struct device *dev) @@ -244,6 +260,7 @@ static int skl_resume(struct device *dev) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; int ret; /* Turned OFF in HDMI codec driver after codec reconfiguration */ @@ -265,8 +282,29 @@ static int skl_resume(struct device *dev) ret = pci_enable_device(pci); snd_hdac_ext_bus_link_power_up_all(ebus); disable_irq_wake(bus->irq); + /* + * turn On the links which are On before active suspend + * and start the CORB/RIRB DMA if On before + * active suspend. + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) + snd_hdac_ext_bus_link_power_up(hlink); + } + + if (ebus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(&ebus->bus); } else { ret = _skl_resume(ebus); + + /* turn off the links which are off before suspend */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (!hlink->ref_count) + snd_hdac_ext_bus_link_power_down(hlink); + } + + if (!ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); } return ret; @@ -316,17 +354,20 @@ static int skl_free(struct hdac_ext_bus *ebus) if (bus->irq >= 0) free_irq(bus->irq, (void *)bus); - if (bus->remap_addr) - iounmap(bus->remap_addr); - snd_hdac_bus_free_stream_pages(bus); snd_hdac_stream_free_all(ebus); snd_hdac_link_free_all(ebus); + + if (bus->remap_addr) + iounmap(bus->remap_addr); + pci_release_regions(skl->pci); pci_disable_device(skl->pci); snd_hdac_ext_bus_exit(ebus); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) + snd_hdac_i915_exit(&ebus->bus); return 0; } @@ -599,6 +640,7 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; + struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -629,7 +671,7 @@ static int skl_probe(struct pci_dev *pci, err = skl_machine_device_register(skl, (void *)pci_id->driver_data); if (err < 0) - goto out_free; + goto out_nhlt_free; err = skl_init_dsp(skl); if (err < 0) { @@ -665,6 +707,12 @@ static int skl_probe(struct pci_dev *pci, } } + /* + * we are done probling so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + /*configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); @@ -679,6 +727,8 @@ out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); +out_nhlt_free: + skl_nhlt_free(skl->nhlt); out_free: skl->init_failed = 1; skl_free(ebus); @@ -719,16 +769,17 @@ static void skl_remove(struct pci_dev *pci) if (skl->tplg) release_firmware(skl->tplg); - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_i915_exit(&ebus->bus); - if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); - pci_dev_put(pci); + + /* codec removal, invoke bus_device_remove */ + snd_hdac_ext_bus_device_remove(ebus); + skl_platform_unregister(&pci->dev); skl_free_dsp(skl); skl_machine_device_unregister(skl); skl_dmic_device_unregister(skl); + skl_nhlt_free(skl->nhlt); skl_free(ebus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 39e16fa7a92b..4b4b3876aea9 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -66,7 +66,7 @@ struct skl { struct platform_device *dmic_dev; struct platform_device *i2s_dev; - void *nhlt; /* nhlt ptr */ + struct nhlt_acpi_table *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ struct skl_dsp_resource resource; @@ -103,8 +103,8 @@ struct skl_dsp_ops { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void *addr); +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(struct nhlt_acpi_table *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 132bb83f8e99..bc3c7b5ac752 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,7 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST + depends on HAS_DMA help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index f7e789e97fbc..3abf51c07851 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_MT8173_RT5650_RT5676 depends on SND_SOC_MEDIATEK && I2C select SND_SOC_RT5645 select SND_SOC_RT5677 + select SND_SOC_HDMI_CODEC help This adds ASoC driver for Mediatek MT8173 boards with the RT5650 and RT5676 codecs. diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173-rt5650-rt5676.c index 5c4c58c69c51..bb593926c62d 100644 --- a/sound/soc/mediatek/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173-rt5650-rt5676.c @@ -134,7 +134,9 @@ static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = { enum { DAI_LINK_PLAYBACK, DAI_LINK_CAPTURE, + DAI_LINK_HDMI, DAI_LINK_CODEC_I2S, + DAI_LINK_HDMI_I2S, DAI_LINK_INTERCODEC }; @@ -161,6 +163,16 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dynamic = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI] = { + .name = "HDMI", + .stream_name = "HDMI PCM", + .cpu_dai_name = "HDMI", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, /* Back End DAI links */ [DAI_LINK_CODEC_I2S] = { @@ -177,6 +189,13 @@ static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = { .dpcm_playback = 1, .dpcm_capture = 1, }, + [DAI_LINK_HDMI_I2S] = { + .name = "HDMI BE", + .cpu_dai_name = "HDMIO", + .no_pcm = 1, + .codec_dai_name = "i2s-hifi", + .dpcm_playback = 1, + }, /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */ [DAI_LINK_INTERCODEC] = { .name = "rt5650_rt5676 intercodec", @@ -251,6 +270,14 @@ static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev) mt8173_rt5650_rt5676_dais[DAI_LINK_INTERCODEC].codec_of_node = mt8173_rt5650_rt5676_codecs[1].of_node; + mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 2); + if (!mt8173_rt5650_rt5676_dais[DAI_LINK_HDMI_I2S].codec_of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index bb09bb1b7f1c..a27a6673dbe3 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -85,12 +85,29 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) { struct snd_soc_card *card = runtime->card; struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + const char *codec_capture_dai = runtime->codec_dais[1]->name; int ret; rt5645_sel_asrc_clk_src(codec, - RT5645_DA_STEREO_FILTER | - RT5645_AD_STEREO_FILTER, + RT5645_DA_STEREO_FILTER, RT5645_CLK_SEL_I2S1_ASRC); + + if (!strcmp(codec_capture_dai, "rt5645-aif1")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } else if (!strcmp(codec_capture_dai, "rt5645-aif2")) { + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else { + dev_warn(card->dev, + "Only one dai codec found in DTS, enabled rt5645 AD filter\n"); + rt5645_sel_asrc_clk_src(codec, + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + } + /* enable jack detection */ ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | @@ -110,6 +127,11 @@ static int mt8173_rt5650_init(struct snd_soc_pcm_runtime *runtime) static struct snd_soc_dai_link_component mt8173_rt5650_codecs[] = { { + /* Playback */ + .dai_name = "rt5645-aif1", + }, + { + /* Capture */ .dai_name = "rt5645-aif1", }, }; @@ -149,7 +171,7 @@ static struct snd_soc_dai_link mt8173_rt5650_dais[] = { .cpu_dai_name = "I2S", .no_pcm = 1, .codecs = mt8173_rt5650_codecs, - .num_codecs = 1, + .num_codecs = 2, .init = mt8173_rt5650_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -177,6 +199,8 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8173_rt5650_card; struct device_node *platform_node; + struct device_node *np; + const char *codec_capture_dai; int i, ret; platform_node = of_parse_phandle(pdev->dev.of_node, @@ -199,6 +223,26 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } + mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; + + if (of_find_node_by_name(platform_node, "codec-capture")) { + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (!np) { + dev_err(&pdev->dev, + "%s: Can't find codec-capture DT node\n", + __func__); + return -EINVAL; + } + ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + if (ret < 0) { + dev_err(&pdev->dev, + "%s codec_capture_dai name fail %d\n", + __func__, ret); + return ret; + } + mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2c12fb..2b5df2ef51a3 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -123,6 +123,7 @@ #define AFE_TDM_CON1_WLEN_32BIT (0x2 << 8) #define AFE_TDM_CON1_MSB_ALIGNED (0x1 << 4) #define AFE_TDM_CON1_1_BCK_DELAY (0x1 << 3) +#define AFE_TDM_CON1_LRCK_INV (0x1 << 2) #define AFE_TDM_CON1_BCK_INV (0x1 << 1) #define AFE_TDM_CON1_EN (0x1 << 0) @@ -449,6 +450,7 @@ static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, runtime->rate * runtime->channels * 32); val = AFE_TDM_CON1_BCK_INV | + AFE_TDM_CON1_LRCK_INV | AFE_TDM_CON1_1_BCK_DELAY | AFE_TDM_CON1_MSB_ALIGNED | /* I2S mode */ AFE_TDM_CON1_WLEN_32BIT | diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index c7563e230c7d..4a16e778966b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -260,6 +260,10 @@ static void omap_st_on(struct omap_mcbsp *mcbsp) if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 1); + /* Disable Sidetone clock auto-gating for normal operation */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE)); + /* Enable McBSP Sidetone */ w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); @@ -279,6 +283,10 @@ static void omap_st_off(struct omap_mcbsp *mcbsp) w = MCBSP_READ(mcbsp, SSELCR); MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + /* Enable Sidetone clock auto-gating to reduce power consumption */ + w = MCBSP_ST_READ(mcbsp, SYSCONFIG); + MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE); + if (mcbsp->pdata->enable_st_clock) mcbsp->pdata->enable_st_clock(mcbsp->id, 0); } diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 99381a27295b..a84f677234f0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -82,6 +82,8 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct dma_chan *chan; int err = 0; + memset(&config, 0x00, sizeof(config)); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index ec522e94b0e2..b6cb9950f05d 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -133,3 +133,4 @@ module_platform_driver(mmp_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("ALSA SoC Brownstone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 5c8f9db50a47..d1661fa6ee08 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver); MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)"); MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 51e790d006f5..96df9b2d8fc4 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("MMP Soc Audio DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index eca60c29791a..ca8b23f8c525 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver); MODULE_AUTHOR("Leo Yan <leoy@marvell.com>"); MODULE_DESCRIPTION("MMP SSPA SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4e74d9573f03..bcc81e920a67 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver); MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index da03fad1b9cd..3cad990dad2c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f3de615aacd7..9615e6de1306 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 9f390398d518..410d48b93031 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..574c6af28c06 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -75,29 +82,37 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_STOP); + if (!i2s->rx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC, - I2S_CLR_TXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_TXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } @@ -113,29 +128,37 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_STOP); + if (!i2s->tx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_RXC, - I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_RXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399de684d..49354d17ea55 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -492,9 +492,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, */ if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, req_rate); + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->clkout[CLKOUT] = clk; of_clk_add_provider(np, of_clk_src_simple_get, clk); @@ -506,9 +504,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, else { for (i = 0; i < CLKOUTMAX; i++) { clk = clk_register_fixed_rate(dev, clkout_name[i], - parent_clk_name, - (parent_clk_name) ? - 0 : CLK_IS_ROOT, + parent_clk_name, 0, req_rate); if (!IS_ERR(clk)) { adg->onecell.clks = adg->clkout; diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8fd7bdc..6bc93cbb3049 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67258ca..a8f61d79333b 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 15d6ffe8be74..e39f916d0f2f 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *priv) i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (ret) goto rsnd_src_probe_done; +skip: i++; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b159610..16369cad4803 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_dai *snd_soc_find_dai( +/** + * snd_soc_find_dai - Find a registered DAI + * + * @dlc: name of the DAI and optional component info to match + * + * This function will search all regsitered components and their DAIs to + * find the DAI of the same name. The component's of_node and name + * should also match if being specified. + * + * Return: pointer of DAI, or NULL if not found. + */ +struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { struct snd_soc_component *component; @@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_find_dai); static bool soc_is_dai_link_bound(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a81dfd..c4464858bf01 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,13 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + /* card won't be set for the dummy component, as a spot fix + * we're checking for that case specifically here but in future + * we will ensure that the dummy component looks like others. + */ + if (!cmpnt->card) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906af387..6cef3977507a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1cf94d7fb9f4..ee7f15aa46fc 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1476,6 +1481,8 @@ widget: widget->dobj.type = SND_SOC_DOBJ_WIDGET; widget->dobj.ops = tplg->ops; widget->dobj.index = tplg->index; + kfree(template.sname); + kfree(template.name); list_add(&widget->dobj.list, &tplg->comp->dobj_list); return 0; @@ -1499,10 +1506,17 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; @@ -1586,6 +1600,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, return snd_soc_register_dai(tplg->comp, dai_drv); } +/* create the FE DAI link */ static int soc_tplg_link_create(struct soc_tplg *tplg, struct snd_soc_tplg_pcm *pcm) { @@ -1598,6 +1613,16 @@ static int soc_tplg_link_create(struct soc_tplg *tplg, link->name = pcm->pcm_name; link->stream_name = pcm->pcm_name; + link->id = pcm->pcm_id; + + link->cpu_dai_name = pcm->dai_name; + link->codec_name = "snd-soc-dummy"; + link->codec_dai_name = "snd-soc-dummy-dai"; + + /* enable DPCM */ + link->dynamic = 1; + link->dpcm_playback = pcm->playback; + link->dpcm_capture = pcm->capture; /* pass control to component driver for optional further init */ ret = soc_tplg_dai_link_load(tplg, link); @@ -1639,8 +1664,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1650,7 +1673,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1670,6 +1699,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1686,6 +1720,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index d14bf411515b..a452ad7cec40 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,7 +15,6 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE - select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -23,9 +22,6 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. -config SND_USB_AUDIO_USE_MEDIA_CONTROLLER - bool - config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 8dca3c407f5a..2d2d122b069f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,8 +15,6 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o -snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o - snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bbba8c7..3fc63583a537 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,7 +66,6 @@ #include "format.h" #include "power.h" #include "stream.h" -#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -612,11 +611,6 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { - /* don't want to fail when media_snd_device_create() fails */ - media_snd_device_create(chip, intf); - } - usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -673,14 +667,6 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } - /* - * Nice to check quirk && quirk->media_device - * need some special handlings. Doesn't look like - * we have access to quirk here - * Acceses mixer_list - */ - media_snd_device_delete(chip); - /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 34a0898e2238..71778ca4b26a 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,8 +105,6 @@ struct snd_usb_endpoint { struct list_head list; }; -struct media_ctl; - struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -158,7 +156,6 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ - struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c deleted file mode 100644 index 93a50d01490c..000000000000 --- a/sound/usb/media.c +++ /dev/null @@ -1,318 +0,0 @@ -/* - * media.c - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh@osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ - -#include <linux/init.h> -#include <linux/list.h> -#include <linux/mutex.h> -#include <linux/slab.h> -#include <linux/usb.h> - -#include <sound/pcm.h> -#include <sound/core.h> - -#include "usbaudio.h" -#include "card.h" -#include "mixer.h" -#include "media.h" - -static int media_snd_enable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->enable_source) - return mctl->media_dev->enable_source(&mctl->media_entity, - &mctl->media_pipe); - return 0; -} - -static void media_snd_disable_source(struct media_ctl *mctl) -{ - if (mctl && mctl->media_dev->disable_source) - mctl->media_dev->disable_source(&mctl->media_entity); -} - -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream) -{ - struct media_device *mdev; - struct media_ctl *mctl; - struct device *pcm_dev = &pcm->streams[stream].dev; - u32 intf_type; - int ret = 0; - u16 mixer_pad; - struct media_entity *entity; - - mdev = subs->stream->chip->media_dev; - if (!mdev) - return -ENODEV; - - if (subs->media_ctl) - return 0; - - /* allocate media_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; - mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; - mixer_pad = 1; - } else { - intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; - mctl->media_pad.flags = MEDIA_PAD_FL_SINK; - mixer_pad = 2; - } - mctl->media_entity.name = pcm->name; - media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) - goto free_mctl; - - mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, - MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); - if (!mctl->intf_devnode) { - ret = -ENOMEM; - goto unregister_entity; - } - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - ret = -ENOMEM; - goto devnode_remove; - } - - /* create link between mixer and audio */ - media_device_for_each_entity(entity, mdev) { - switch (entity->function) { - case MEDIA_ENT_F_AUDIO_MIXER: - ret = media_create_pad_link(entity, mixer_pad, - &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); - if (ret) - goto remove_intf_link; - break; - } - } - - subs->media_ctl = mctl; - return 0; - -remove_intf_link: - media_remove_intf_link(mctl->intf_link); -devnode_remove: - media_devnode_remove(mctl->intf_devnode); -unregister_entity: - media_device_unregister_entity(&mctl->media_entity); -free_mctl: - kfree(mctl); - return ret; -} - -void media_snd_stream_delete(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl && mctl->media_dev) { - struct media_device *mdev; - - mdev = subs->stream->chip->media_dev; - if (mdev && media_devnode_is_registered(&mdev->devnode)) { - media_devnode_remove(mctl->intf_devnode); - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - subs->media_ctl = NULL; - } -} - -int media_snd_start_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - return media_snd_enable_source(mctl); - return 0; -} - -void media_snd_stop_pipeline(struct snd_usb_substream *subs) -{ - struct media_ctl *mctl = subs->media_ctl; - - if (mctl) - media_snd_disable_source(mctl); -} - -int media_snd_mixer_init(struct snd_usb_audio *chip) -{ - struct device *ctl_dev = &chip->card->ctl_dev; - struct media_intf_devnode *ctl_intf; - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - struct media_mixer_ctl *mctl; - u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; - int ret; - - if (!mdev) - return -ENODEV; - - ctl_intf = chip->ctl_intf_media_devnode; - if (!ctl_intf) { - ctl_intf = media_devnode_create(mdev, intf_type, 0, - MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); - if (!ctl_intf) - return -ENOMEM; - chip->ctl_intf_media_devnode = ctl_intf; - } - - list_for_each_entry(mixer, &chip->mixer_list, list) { - - if (mixer->media_mixer_ctl) - continue; - - /* allocate media_mixer_ctl */ - mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); - if (!mctl) - return -ENOMEM; - - mctl->media_dev = mdev; - mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; - mctl->media_entity.name = chip->card->mixername; - mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; - mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; - mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; - media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, - mctl->media_pad); - ret = media_device_register_entity(mctl->media_dev, - &mctl->media_entity); - if (ret) { - kfree(mctl); - return ret; - } - - mctl->intf_link = media_create_intf_link(&mctl->media_entity, - &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); - if (!mctl->intf_link) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - kfree(mctl); - return -ENOMEM; - } - mctl->intf_devnode = ctl_intf; - mixer->media_mixer_ctl = mctl; - } - return 0; -} - -static void media_snd_mixer_delete(struct snd_usb_audio *chip) -{ - struct usb_mixer_interface *mixer; - struct media_device *mdev = chip->media_dev; - - if (!mdev) - return; - - list_for_each_entry(mixer, &chip->mixer_list, list) { - struct media_mixer_ctl *mctl; - - mctl = mixer->media_mixer_ctl; - if (!mixer->media_mixer_ctl) - continue; - - if (media_devnode_is_registered(&mdev->devnode)) { - media_device_unregister_entity(&mctl->media_entity); - media_entity_cleanup(&mctl->media_entity); - } - kfree(mctl); - mixer->media_mixer_ctl = NULL; - } - if (media_devnode_is_registered(&mdev->devnode)) - media_devnode_remove(chip->ctl_intf_media_devnode); - chip->ctl_intf_media_devnode = NULL; -} - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) -{ - struct media_device *mdev; - struct usb_device *usbdev = interface_to_usbdev(iface); - int ret; - - mdev = media_device_get_devres(&usbdev->dev); - if (!mdev) - return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } - if (!media_devnode_is_registered(&mdev->devnode)) { - ret = media_device_register(mdev); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't register media device. Error: %d\n", - ret); - return ret; - } - } - - /* save media device - avoid lookups */ - chip->media_dev = mdev; - - /* Create media entities for mixer and control dev */ - ret = media_snd_mixer_init(chip); - if (ret) { - dev_err(&usbdev->dev, - "Couldn't create media mixer entities. Error: %d\n", - ret); - - /* clear saved media_dev */ - chip->media_dev = NULL; - - return ret; - } - return 0; -} - -void media_snd_device_delete(struct snd_usb_audio *chip) -{ - struct media_device *mdev = chip->media_dev; - - media_snd_mixer_delete(chip); - - if (mdev) { - if (media_devnode_is_registered(&mdev->devnode)) - media_device_unregister(mdev); - chip->media_dev = NULL; - } -} diff --git a/sound/usb/media.h b/sound/usb/media.h deleted file mode 100644 index 1dcdcdc5f7aa..000000000000 --- a/sound/usb/media.h +++ /dev/null @@ -1,72 +0,0 @@ -/* - * media.h - Media Controller specific ALSA driver code - * - * Copyright (c) 2016 Shuah Khan <shuahkh@osg.samsung.com> - * Copyright (c) 2016 Samsung Electronics Co., Ltd. - * - * This file is released under the GPLv2. - */ - -/* - * This file adds Media Controller support to ALSA driver - * to use the Media Controller API to share tuner with DVB - * and V4L2 drivers that control media device. Media device - * is created based on existing quirks framework. Using this - * approach, the media controller API usage can be added for - * a specific device. -*/ -#ifndef __MEDIA_H - -#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER - -#include <media/media-device.h> -#include <media/media-entity.h> -#include <sound/asound.h> - -struct media_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad; - struct media_pipeline media_pipe; -}; - -/* - * One source pad each for SNDRV_PCM_STREAM_CAPTURE and - * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link - * to AUDIO Source -*/ -#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) - -struct media_mixer_ctl { - struct media_device *media_dev; - struct media_entity media_entity; - struct media_intf_devnode *intf_devnode; - struct media_link *intf_link; - struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; - struct media_pipeline media_pipe; -}; - -int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface); -void media_snd_device_delete(struct snd_usb_audio *chip); -int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, - int stream); -void media_snd_stream_delete(struct snd_usb_substream *subs); -int media_snd_start_pipeline(struct snd_usb_substream *subs); -void media_snd_stop_pipeline(struct snd_usb_substream *subs); -#else -static inline int media_snd_device_create(struct snd_usb_audio *chip, - struct usb_interface *iface) - { return 0; } -static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } -static inline int media_snd_stream_init(struct snd_usb_substream *subs, - struct snd_pcm *pcm, int stream) - { return 0; } -static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } -static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) - { return 0; } -static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } -#endif -#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index f3789446ab9c..3417ef347e40 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,8 +3,6 @@ #include <sound/info.h> -struct media_mixer_ctl; - struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -24,7 +22,6 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; - struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index ddca6547399b..1f8fb0d904e0 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -349,6 +349,16 @@ static struct usbmix_name_map bose_companion5_map[] = { }; /* + * Dell usb dock with ALC4020 codec had a firmware problem where it got + * screwed up when zero volume is passed; just skip it as a workaround + */ +static const struct usbmix_name_map dell_alc4020_map[] = { + { 16, NULL }, + { 19, NULL }, + { 0 } +}; + +/* * Control map entries */ @@ -431,6 +441,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = aureon_51_2_map, }, { + .id = USB_ID(0x0bda, 0x4014), + .map = dell_alc4020_map, + }, + { .id = USB_ID(0x0dba, 0x1000), .map = mbox1_map, }, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0e4e0640c504..44d178ee9177 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,7 +35,6 @@ #include "pcm.h" #include "clock.h" #include "power.h" -#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -718,14 +717,10 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; - ret = media_snd_start_pipeline(subs); - if (ret) - return ret; - ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - goto err_ret; + return ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -739,27 +734,22 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - ret = -EINVAL; - goto err_ret; + return -EINVAL; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - goto err_ret; + return ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; - -err_ret: - media_snd_stop_pipeline(subs); - return ret; } /* @@ -771,7 +761,6 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; - media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1232,7 +1221,6 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; - int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1246,12 +1234,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - ret = setup_hw_info(runtime, subs); - if (ret == 0) - ret = media_snd_stream_init(subs, as->pcm, direction); - if (ret) - snd_usb_autosuspend(subs->stream->chip); - return ret; + return setup_hw_info(runtime, subs); } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1260,7 +1243,6 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); - media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9d087b19c70c..c60a776e815d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,7 +2886,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ - .media_device = 1, \ } \ } diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fb62bce2435c..6adde457b602 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -150,6 +150,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, usb_audio_err(chip, "cannot memdup\n"); return -ENOMEM; } + INIT_LIST_HEAD(&fp->list); if (fp->nr_rates > MAX_NR_RATES) { kfree(fp); return -EINVAL; @@ -193,6 +194,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return 0; error: + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp); kfree(rate_table); return err; @@ -469,6 +471,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = 0; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + INIT_LIST_HEAD(&fp->list); switch (fp->maxpacksize) { case 0x120: @@ -492,6 +495,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp); return err; } @@ -1130,9 +1134,14 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case USB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ + case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 51258a15f653..8e9548bc1f1a 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,7 +36,6 @@ #include "format.h" #include "clock.h" #include "stream.h" -#include "media.h" /* * free a substream @@ -53,7 +52,6 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); - media_snd_stream_delete(subs); } @@ -316,7 +314,9 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, /* * add this endpoint to the chip instance. * if a stream with the same endpoint already exists, append to it. - * if not, create a new pcm stream. + * if not, create a new pcm stream. note, fp is added to the substream + * fmt_list and will be freed on the chip instance release. do not free + * fp or do remove it from the substream fmt_list to avoid double-free. */ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, int stream, @@ -677,6 +677,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) * (fp->maxpacksize & 0x7ff); fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no); fp->clock = clock; + INIT_LIST_HEAD(&fp->list); /* some quirks for attributes here */ @@ -725,6 +726,7 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) dev_dbg(&dev->dev, "%u:%d: add audio endpoint %#x\n", iface_no, altno, fp->endpoint); err = snd_usb_add_audio_stream(chip, stream, fp); if (err < 0) { + list_del(&fp->list); /* unlink for avoiding double-free */ kfree(fp->rate_table); kfree(fp->chmap); kfree(fp); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..b665d85555cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,9 +30,6 @@ * */ -struct media_device; -struct media_intf_devnode; - struct snd_usb_audio { int index; struct usb_device *dev; @@ -63,8 +60,6 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ - struct media_device *media_dev; - struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -115,7 +110,6 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; - bool media_device; const void *data; }; |