From 0052b7dcf9d9ec6be4fc3fe815a2ceda623bb9d1 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 13 Sep 2015 19:00:05 +0900 Subject: ALSA: pcm: remove structure member of 'struct snd_pcm_hwptr_log *' type because this structure had been removed This structure was added by 4d96eb255c53 ('ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions') to store PCM pointers information of latest 10 pointer movements (=XRUN_LOG_CNT). When CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has 'hwptr_log' member with a pointer to the structure. When calling xrun_log() in pcm_lib.c, the structure was allocated to the pointer. When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer is released. In f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'), the pointer logging is replaced with using Linux Kernel Tracepoints. The structure was also removed, while it's just declared. The member and kfree still remains. This commit removes the member and related codes. I think this was overlooked because it brings no errors/warnings to C compilers. Fixes: f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints') Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 6 ------ 1 file changed, 6 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee0a510..a4fcc9456194 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -285,8 +285,6 @@ struct snd_pcm_hw_constraint_ranges { unsigned int mask; }; -struct snd_pcm_hwptr_log; - /* * userspace-provided audio timestamp config to kernel, * structure is for internal use only and filled with dedicated unpack routine @@ -428,10 +426,6 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif - -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_pcm_hwptr_log *hwptr_log; -#endif }; struct snd_pcm_group { /* keep linked substreams */ -- cgit v1.2.3 From 7486d80f7d853f50088124824bf62d9a4d14de75 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 10 Sep 2015 06:49:29 +0000 Subject: ASoC: rsnd: remove unneeded sh_clk header sh_clk header is not needed, and it will create confusion. Let's remove it. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 1 - sound/soc/sh/rcar/adg.c | 1 - 2 files changed, 2 deletions(-) (limited to 'include') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index bb7b2ebfee7b..d8e33d38da43 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -12,7 +12,6 @@ #ifndef RCAR_SND_H #define RCAR_SND_H -#include #define RSND_GEN1_SRU 0 #define RSND_GEN1_ADG 1 diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index fefc881dbac2..b512be82306e 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -7,7 +7,6 @@ * License. See the file "COPYING" in the main directory of this archive * for more details. */ -#include #include "rsnd.h" #define CLKA 0 -- cgit v1.2.3 From 6131084a0bc966107021d8c89489f9cd1663b902 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Wed, 9 Sep 2015 21:27:43 +0300 Subject: ASoC: simple-card: Add tdm slot mask support to simple-card Adds DT binding for explicitly choosing a tdm mask for DAI and uses it in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been changed. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tdm-slot.txt | 11 +++++++++- include/sound/simple_card.h | 2 ++ include/sound/soc.h | 2 ++ sound/soc/generic/simple-card.c | 8 +++++-- sound/soc/soc-core.c | 25 ++++++++++++++++++++++ 5 files changed, 45 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/tdm-slot.txt b/Documentation/devicetree/bindings/sound/tdm-slot.txt index 6a2c84247f91..34cf70e2cbc4 100644 --- a/Documentation/devicetree/bindings/sound/tdm-slot.txt +++ b/Documentation/devicetree/bindings/sound/tdm-slot.txt @@ -4,11 +4,15 @@ This specifies audio DAI's TDM slot. TDM slot properties: dai-tdm-slot-num : Number of slots in use. -dai-tdm-slot-width : Width in bits for each slot. +dai-tdm-slot-width : Width in bits for each slot. +dai-tdm-slot-tx-mask : Transmit direction slot mask, optional +dai-tdm-slot-rx-mask : Receive direction slot mask, optional For instance: dai-tdm-slot-num = <2>; dai-tdm-slot-width = <8>; + dai-tdm-slot-tx-mask = <0 1>; + dai-tdm-slot-rx-mask = <1 0>; And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() to specify a explicit mapping of the channels and the slots. If it's absent @@ -18,3 +22,8 @@ tx and rx masks. For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit for an active slot as default, and the default active bits are at the LSB of the masks. + +The explicit masks are given as array of integers, where the first +number presents bit-0 (LSB), second presents bit-1, etc. Any non zero +number is considered 1 and 0 is 0. snd_soc_of_xlate_tdm_slot_mask() +does not do anything, if either mask is set non zero value. diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index b9b4f289fe6b..0399352f3a62 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -19,6 +19,8 @@ struct asoc_simple_dai { unsigned int sysclk; int slots; int slot_width; + unsigned int tx_slot_mask; + unsigned int rx_slot_mask; struct clk *clk; }; diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..a76622d7bb2f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1601,6 +1601,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width); void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ff76d419436..54c33204541f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -151,7 +151,9 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, } if (set->slots) { - ret = snd_soc_dai_set_tdm_slot(dai, 0, 0, + ret = snd_soc_dai_set_tdm_slot(dai, + set->tx_slot_mask, + set->rx_slot_mask, set->slots, set->slot_width); if (ret && ret != -ENOTSUPP) { @@ -243,7 +245,9 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return ret; /* Parse TDM slot */ - ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); + ret = snd_soc_of_parse_tdm_slot(np, &dai->tx_slot_mask, + &dai->rx_slot_mask, + &dai->slots, &dai->slot_width); if (ret) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15236c3..c5e21ca0c015 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3291,13 +3291,38 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); +static int snd_soc_of_get_slot_mask(struct device_node *np, + const char *prop_name, + unsigned int *mask) +{ + u32 val; + const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + int i; + + if (!of_slot_mask) + return 0; + val /= sizeof(u32); + for (i = 0; i < val; i++) + if (be32_to_cpup(&of_slot_mask[i])) + *mask |= (1 << i); + + return val; +} + int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *tx_mask, + unsigned int *rx_mask, unsigned int *slots, unsigned int *slot_width) { u32 val; int ret; + if (tx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-tx-mask", tx_mask); + if (rx_mask) + snd_soc_of_get_slot_mask(np, "dai-tdm-slot-rx-mask", rx_mask); + if (of_property_read_bool(np, "dai-tdm-slot-num")) { ret = of_property_read_u32(np, "dai-tdm-slot-num", &val); if (ret) -- cgit v1.2.3 From a54e22f40480eb63533d3f0520e9c84b102dd09f Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 16 Sep 2015 13:59:42 +0100 Subject: ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro Add a version of the SOC_SINGLE_RANGE_TLV macro that allows a custom get and put to be specified. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- include/sound/soc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..9ffa28514fa7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -226,6 +226,18 @@ .info = snd_soc_info_volsw, \ .get = xhandler_get, .put = xhandler_put, \ .private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert, 0) } +#define SOC_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ -- cgit v1.2.3 From 917536aeb88d34e06c1353b0dd144f0987bb66bd Mon Sep 17 00:00:00 2001 From: John Lin Date: Mon, 21 Sep 2015 14:12:01 +0800 Subject: ASoC: rt5645: Add jd_invert for Broadwell Broadwell can not triger the IRQ falling and rising simultaneously, so it can not detect jack-in and jack-out simultaneously. We add a flag "jd_invert" to platform data. If this flag is set, codec IRQ will be set to invert that forces IRQ as pulse when jack-in and jack-out. Signed-off-by: John Lin Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- include/sound/rt5645.h | 2 ++ sound/soc/codecs/rt5645.c | 7 +++++++ sound/soc/codecs/rt5645.h | 4 ++++ 3 files changed, 13 insertions(+) (limited to 'include') diff --git a/include/sound/rt5645.h b/include/sound/rt5645.h index 22734bc3ffd4..a5cf6152e778 100644 --- a/include/sound/rt5645.h +++ b/include/sound/rt5645.h @@ -21,6 +21,8 @@ struct rt5645_platform_data { /* 0 = IN2P; 1 = GPIO6; 2 = GPIO10; 3 = GPIO12 */ unsigned int jd_mode; + /* Invert JD when jack insert */ + bool jd_invert; }; #endif diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index e83068886c70..b0d96b6d21de 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2829,6 +2829,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } + if (rt5645->pdata.jd_invert) + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); } else { /* jack out */ rt5645->jack_type = 0; @@ -2844,6 +2847,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_disable_pin(dapm, "LDO2"); snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); snd_soc_dapm_sync(dapm); + if (rt5645->pdata.jd_invert) + regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, + RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR); } return rt5645->jack_type; @@ -3213,6 +3219,7 @@ static struct rt5645_platform_data buddy_platform_data = { .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, .jd_mode = 3, + .jd_invert = true, }; static int buddy_quirk_cb(const struct dmi_system_id *id) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 0e4cfc6ac649..90325d9256ff 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1626,6 +1626,10 @@ #define RT5645_OT_P_NOR (0x0 << 10) #define RT5645_OT_P_INV (0x1 << 10) #define RT5645_IRQ_JD_1_1_EN (0x1 << 9) +#define RT5645_JD_1_1_MASK (0x1 << 7) +#define RT5645_JD_1_1_SFT 7 +#define RT5645_JD_1_1_NOR (0x0 << 7) +#define RT5645_JD_1_1_INV (0x1 << 7) /* IRQ Control 2 (0xbe) */ #define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) -- cgit v1.2.3 From 5334240c30cb0058fa8c373bf0d653337833230d Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Wed, 2 Sep 2015 14:11:38 +0800 Subject: drm/i915: Add audio sync_audio_rate callback Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang Reviewed-by: Jani Nikula Signed-off-by: Takashi Iwai --- include/drm/i915_component.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b2d56dd483d9..e6d35d7239c0 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,13 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + /** + * @sync_audio_rate: set n/cts based on the sample rate + * + * Called from audio driver. After audio driver sets the + * sample rate, it will call this function to set n/cts + */ + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; const struct i915_audio_component_audio_ops { -- cgit v1.2.3 From 7e8275c2f2bbb384e18af37066b8b2f32b7d092f Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Fri, 25 Sep 2015 09:36:12 +0800 Subject: drm/i915: set proper N/CTS in modeset When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang Reviewed-by: Jani Nikula Signed-off-by: Takashi Iwai --- drivers/gpu/drm/i915/intel_audio.c | 57 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 58 insertions(+), 9 deletions(-) (limited to 'include') diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index c5922f902aaf..30f6859dcb36 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,9 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + BUILD_BUG_ON(MAX_PORTS != I915_MAX_PORTS); + for (i = 0; i < ARRAY_SIZE(acomp->aud_sample_rate); i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7239c0..89dc7d6bc1cc 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- cgit v1.2.3 From b7631a12e726597cffde1b29cc3bcf811981c1fb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Sep 2015 12:19:08 +0200 Subject: ALSA: hda - Fix typos in snd_hdac_regmap_*() documents Fixes the wrong reference names to regmap amp functions. Signed-off-by: Takashi Iwai --- include/sound/hda_regmap.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'include') diff --git a/include/sound/hda_regmap.h b/include/sound/hda_regmap.h index df705908480a..2767c55a641e 100644 --- a/include/sound/hda_regmap.h +++ b/include/sound/hda_regmap.h @@ -67,7 +67,7 @@ int snd_hdac_regmap_update_raw(struct hdac_device *codec, unsigned int reg, * @reg: verb to write * @val: value to write * - * For writing an amp value, use snd_hda_regmap_amp_update(). + * For writing an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, @@ -85,7 +85,7 @@ snd_hdac_regmap_write(struct hdac_device *codec, hda_nid_t nid, * @mask: bit mask to update * @val: value to update * - * For updating an amp value, use snd_hda_regmap_amp_update(). + * For updating an amp value, use snd_hdac_regmap_update_amp(). */ static inline int snd_hdac_regmap_update(struct hdac_device *codec, hda_nid_t nid, -- cgit v1.2.3 From 660dd3d52ead45b8e60dcf966daf304de2121a28 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Sep 2015 09:39:21 +0900 Subject: ALSA: firewire-digi00x: add hwdep interface This commit adds hwdep interface so as the other sound drivers for units on IEEE 1394 bus have. This interface is designed for mixer/control applications. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 1 + sound/firewire/digi00x/Makefile | 2 +- sound/firewire/digi00x/digi00x-hwdep.c | 192 ++++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 23 +++- sound/firewire/digi00x/digi00x-stream.c | 39 +++++++ sound/firewire/digi00x/digi00x.c | 6 + sound/firewire/digi00x/digi00x.h | 14 +++ 9 files changed, 273 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a45be6bdcf5b..aa329132f6c4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -100,9 +100,10 @@ enum { SNDRV_HWDEP_IFACE_FW_FIREWORKS, /* Echo Audio Fireworks based device */ SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ + SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_OXFW + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DIGI00X }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index 49122df3b56b..f67d228f731b 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -56,6 +56,7 @@ union snd_firewire_event { #define SNDRV_FIREWIRE_TYPE_FIREWORKS 2 #define SNDRV_FIREWIRE_TYPE_BEBOB 3 #define SNDRV_FIREWIRE_TYPE_OXFW 4 +#define SNDRV_FIREWIRE_TYPE_DIGI00X 5 /* RME, MOTU, ... */ struct snd_firewire_get_info { diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 74f17a8263db..e61445e2ceaa 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -123,6 +123,7 @@ config SND_BEBOB config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" select SND_FIREWIRE_LIB + select SND_HWDEP help Say Y here to include support for Digidesign Digi 002/003 family. * Digi 002 Console diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile index 3c0ff6fff1f0..5bc3c7780a51 100644 --- a/sound/firewire/digi00x/Makefile +++ b/sound/firewire/digi00x/Makefile @@ -1,3 +1,3 @@ snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \ - digi00x-pcm.o digi00x.o + digi00x-pcm.o digi00x-hwdep.o digi00x.o obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c new file mode 100644 index 000000000000..d629e415dc8b --- /dev/null +++ b/sound/firewire/digi00x/digi00x-hwdep.c @@ -0,0 +1,192 @@ +/* + * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +/* + * This codes give three functionality. + * + * 1.get firewire node information + * 2.get notification about starting/stopping stream + * 3.lock/unlock stream + */ + +#include "digi00x.h" + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&dg00x->lock); + + while (!dg00x->dev_lock_changed) { + prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&dg00x->lock); + schedule(); + finish_wait(&dg00x->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&dg00x->lock); + } + + memset(&event, 0, sizeof(event)); + if (dg00x->dev_lock_changed) { + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (dg00x->dev_lock_count > 0); + dg00x->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + } + + spin_unlock_irq(&dg00x->lock); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + unsigned int events; + + poll_wait(file, &dg00x->hwdep_wait, wait); + + spin_lock_irq(&dg00x->lock); + if (dg00x->dev_lock_changed) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&dg00x->lock); + + return events; +} + +static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(dg00x->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_DIGI00X; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + if (dg00x->dev_lock_count == 0) { + dg00x->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&dg00x->lock); + + return err; +} + +static int hwdep_unlock(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + if (dg00x->dev_lock_count == -1) { + dg00x->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&dg00x->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + + spin_lock_irq(&dg00x->lock); + if (dg00x->dev_lock_count == -1) + dg00x->dev_lock_count = 0; + spin_unlock_irq(&dg00x->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_dg00x *dg00x = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(dg00x, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(dg00x); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(dg00x); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +static const struct snd_hwdep_ops hwdep_ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, +}; + +int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x) +{ + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep); + if (err < 0) + return err; + + strcpy(hwdep->name, "Digi00x"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X; + hwdep->ops = hwdep_ops; + hwdep->private_data = dg00x; + hwdep->exclusive = true; + + return err; +} diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c index 87c9aa234890..cac28f70aef7 100644 --- a/sound/firewire/digi00x/digi00x-pcm.c +++ b/sound/firewire/digi00x/digi00x-pcm.c @@ -118,21 +118,25 @@ static int pcm_open(struct snd_pcm_substream *substream) unsigned int rate; int err; + err = snd_dg00x_stream_lock_try(dg00x); + if (err < 0) + goto end; + err = pcm_init_hw_params(dg00x, substream); if (err < 0) - return err; + goto err_locked; /* Check current clock source. */ err = snd_dg00x_stream_get_clock(dg00x, &clock); if (err < 0) - return err; + goto err_locked; if (clock != SND_DG00X_CLOCK_INTERNAL) { err = snd_dg00x_stream_check_external_clock(dg00x, &detect); if (err < 0) - return err; + goto err_locked; if (!detect) { err = -EBUSY; - return err; + goto err_locked; } } @@ -141,18 +145,25 @@ static int pcm_open(struct snd_pcm_substream *substream) amdtp_stream_pcm_running(&dg00x->tx_stream)) { err = snd_dg00x_stream_get_external_rate(dg00x, &rate); if (err < 0) - return err; + goto err_locked; substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; } snd_pcm_set_sync(substream); - +end: + return err; +err_locked: + snd_dg00x_stream_lock_release(dg00x); return err; } static int pcm_close(struct snd_pcm_substream *substream) { + struct snd_dg00x *dg00x = substream->private_data; + + snd_dg00x_stream_lock_release(dg00x); + return 0; } diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c index 3278c3e08560..8aac31be3132 100644 --- a/sound/firewire/digi00x/digi00x-stream.c +++ b/sound/firewire/digi00x/digi00x-stream.c @@ -379,3 +379,42 @@ void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x) amdtp_stream_update(&dg00x->tx_stream); amdtp_stream_update(&dg00x->rx_stream); } + +void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x) +{ + dg00x->dev_lock_changed = true; + wake_up(&dg00x->hwdep_wait); +} + +int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x) +{ + int err; + + spin_lock_irq(&dg00x->lock); + + /* user land lock this */ + if (dg00x->dev_lock_count < 0) { + err = -EBUSY; + goto end; + } + + /* this is the first time */ + if (dg00x->dev_lock_count++ == 0) + snd_dg00x_stream_lock_changed(dg00x); + err = 0; +end: + spin_unlock_irq(&dg00x->lock); + return err; +} + +void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x) +{ + spin_lock_irq(&dg00x->lock); + + if (WARN_ON(dg00x->dev_lock_count <= 0)) + goto end; + if (--dg00x->dev_lock_count == 0) + snd_dg00x_stream_lock_changed(dg00x); +end: + spin_unlock_irq(&dg00x->lock); +} diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 166c1d43123f..cd93030df1cd 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -72,6 +72,8 @@ static int snd_dg00x_probe(struct fw_unit *unit, dg00x->unit = fw_unit_get(unit); mutex_init(&dg00x->mutex); + spin_lock_init(&dg00x->lock); + init_waitqueue_head(&dg00x->hwdep_wait); err = name_card(dg00x); if (err < 0) @@ -87,6 +89,10 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; + err = snd_dg00x_create_hwdep_device(dg00x); + if (err < 0) + goto error; + err = snd_card_register(card); if (err < 0) goto error; diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 199064504a55..5ba531806262 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -22,6 +22,8 @@ #include #include #include +#include +#include #include "../lib.h" #include "../iso-resources.h" @@ -32,6 +34,7 @@ struct snd_dg00x { struct fw_unit *unit; struct mutex mutex; + spinlock_t lock; struct amdtp_stream tx_stream; struct fw_iso_resources tx_resources; @@ -40,6 +43,12 @@ struct snd_dg00x { struct fw_iso_resources rx_resources; unsigned int substreams_counter; + + /* for uapi */ + int dev_lock_count; + bool dev_lock_changed; + wait_queue_head_t hwdep_wait; + }; #define DG00X_ADDR_BASE 0xffffe0000000ull @@ -118,8 +127,13 @@ void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x); void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x); +void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x); +int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x); +void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x); + void snd_dg00x_proc_init(struct snd_dg00x *dg00x); int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x); +int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x); #endif -- cgit v1.2.3 From 44b7308871ac6fd85fc840bfa3ddb466fe7aff23 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 30 Sep 2015 09:39:22 +0900 Subject: ALSA: firewire-digi00x: add support for asynchronous messaging Digi 002/003 family uses asynchronous transaction for messaging. The address to transmit this message is stored on a certain register. This commit allocates a range of address on OHCI 1394 host controller to handle the messaging. As long as I know, the purpose of this message seems to notify lost of synchronization. While, the meaning of content of the message is not clear. Actual examples of this messaging: * When clock source is set as internal: - 0x00007051 - 0x00007052 - 0x00007054 - 0x00007057 - 0x00007058 * When clock source is set as somewhat external: - 0x00009000 - 0x00009010 - 0x00009020 - 0x00009021 - 0x00009022 The lost often occurs when using internal clock source. In this case, users hear sounds with quite short gap every several minutes. In fact, the lost is recovered temporarily. When using with external clock source, the lost seems not to occur. The mechanism is not clear yet. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/firewire.h | 7 +++ sound/firewire/digi00x/Makefile | 3 +- sound/firewire/digi00x/digi00x-hwdep.c | 12 ++++- sound/firewire/digi00x/digi00x-transaction.c | 81 ++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 7 +++ sound/firewire/digi00x/digi00x.h | 7 +++ 6 files changed, 114 insertions(+), 3 deletions(-) create mode 100644 sound/firewire/digi00x/digi00x-transaction.c (limited to 'include') diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index f67d228f731b..deb041cb9af0 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -9,6 +9,7 @@ #define SNDRV_FIREWIRE_EVENT_LOCK_STATUS 0x000010cc #define SNDRV_FIREWIRE_EVENT_DICE_NOTIFICATION 0xd1ce004e #define SNDRV_FIREWIRE_EVENT_EFW_RESPONSE 0x4e617475 +#define SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE 0x746e736c struct snd_firewire_event_common { unsigned int type; /* SNDRV_FIREWIRE_EVENT_xxx */ @@ -40,11 +41,17 @@ struct snd_firewire_event_efw_response { __be32 response[0]; /* some responses */ }; +struct snd_firewire_event_digi00x_message { + unsigned int type; + __u32 message; /* Digi00x-specific message */ +}; + union snd_firewire_event { struct snd_firewire_event_common common; struct snd_firewire_event_lock_status lock_status; struct snd_firewire_event_dice_notification dice_notification; struct snd_firewire_event_efw_response efw_response; + struct snd_firewire_event_digi00x_message digi00x_message; }; diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile index 5bc3c7780a51..28e3d137ef57 100644 --- a/sound/firewire/digi00x/Makefile +++ b/sound/firewire/digi00x/Makefile @@ -1,3 +1,4 @@ snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \ - digi00x-pcm.o digi00x-hwdep.o digi00x.o + digi00x-pcm.o digi00x-hwdep.o \ + digi00x-transaction.o digi00x.o obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c index d629e415dc8b..f188e4758fd2 100644 --- a/sound/firewire/digi00x/digi00x-hwdep.c +++ b/sound/firewire/digi00x/digi00x-hwdep.c @@ -12,6 +12,7 @@ * 1.get firewire node information * 2.get notification about starting/stopping stream * 3.lock/unlock stream + * 4.get asynchronous messaging */ #include "digi00x.h" @@ -25,7 +26,7 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, spin_lock_irq(&dg00x->lock); - while (!dg00x->dev_lock_changed) { + while (!dg00x->dev_lock_changed && dg00x->msg == 0) { prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE); spin_unlock_irq(&dg00x->lock); schedule(); @@ -42,6 +43,13 @@ static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, dg00x->dev_lock_changed = false; count = min_t(long, count, sizeof(event.lock_status)); + } else { + event.digi00x_message.type = + SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE; + event.digi00x_message.message = dg00x->msg; + dg00x->msg = 0; + + count = min_t(long, count, sizeof(event.digi00x_message)); } spin_unlock_irq(&dg00x->lock); @@ -61,7 +69,7 @@ static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, poll_wait(file, &dg00x->hwdep_wait, wait); spin_lock_irq(&dg00x->lock); - if (dg00x->dev_lock_changed) + if (dg00x->dev_lock_changed || dg00x->msg) events = POLLIN | POLLRDNORM; else events = 0; diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c new file mode 100644 index 000000000000..49372901a1e1 --- /dev/null +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -0,0 +1,81 @@ +/* + * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family + * + * Copyright (c) 2014-2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include +#include "digi00x.h" + +static void handle_unknown_message(struct snd_dg00x *dg00x, + unsigned long long offset, __be32 *buf) +{ + unsigned long flags; + + spin_lock_irqsave(&dg00x->lock, flags); + dg00x->msg = be32_to_cpu(*buf); + spin_unlock_irqrestore(&dg00x->lock, flags); + + wake_up(&dg00x->hwdep_wait); +} + +static void handle_message(struct fw_card *card, struct fw_request *request, + int tcode, int destination, int source, + int generation, unsigned long long offset, + void *data, size_t length, void *callback_data) +{ + struct snd_dg00x *dg00x = callback_data; + __be32 *buf = (__be32 *)data; + + if (offset == dg00x->async_handler.offset) + handle_unknown_message(dg00x, offset, buf); + + fw_send_response(card, request, RCODE_COMPLETE); +} + +int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x) +{ + struct fw_device *device = fw_parent_device(dg00x->unit); + __be32 data[2]; + + /* Unknown. 4bytes. */ + data[0] = cpu_to_be32((device->card->node_id << 16) | + (dg00x->async_handler.offset >> 32)); + data[1] = cpu_to_be32(dg00x->async_handler.offset); + return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST, + DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR, + &data, sizeof(data), 0); +} + +int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) +{ + static const struct fw_address_region resp_register_region = { + .start = 0xffffe0000000ull, + .end = 0xffffe000ffffull, + }; + int err; + + dg00x->async_handler.length = 4; + dg00x->async_handler.address_callback = handle_message; + dg00x->async_handler.callback_data = dg00x; + + err = fw_core_add_address_handler(&dg00x->async_handler, + &resp_register_region); + if (err < 0) + return err; + + err = snd_dg00x_transaction_reregister(dg00x); + if (err < 0) { + fw_core_remove_address_handler(&dg00x->async_handler); + dg00x->async_handler.address_callback = NULL; + } + + return err; +} + +void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x) +{ + fw_core_remove_address_handler(&dg00x->async_handler); +} diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index cd93030df1cd..34937a26c198 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -46,6 +46,7 @@ static void dg00x_card_free(struct snd_card *card) struct snd_dg00x *dg00x = card->private_data; snd_dg00x_stream_destroy_duplex(dg00x); + snd_dg00x_transaction_unregister(dg00x); fw_unit_put(dg00x->unit); @@ -93,6 +94,10 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; + err = snd_dg00x_transaction_register(dg00x); + if (err < 0) + goto error; + err = snd_card_register(card); if (err < 0) goto error; @@ -109,6 +114,8 @@ static void snd_dg00x_update(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + snd_dg00x_transaction_reregister(dg00x); + mutex_lock(&dg00x->mutex); snd_dg00x_stream_update_duplex(dg00x); mutex_unlock(&dg00x->mutex); diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 5ba531806262..b960c868b59b 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -49,6 +49,9 @@ struct snd_dg00x { bool dev_lock_changed; wait_queue_head_t hwdep_wait; + /* For asynchronous messages. */ + struct fw_address_handler async_handler; + u32 msg; }; #define DG00X_ADDR_BASE 0xffffe0000000ull @@ -110,6 +113,10 @@ int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s, struct snd_pcm_runtime *runtime); void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format); +int snd_dg00x_transaction_register(struct snd_dg00x *dg00x); +int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x); +void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x); + extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT]; extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT]; int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x, -- cgit v1.2.3 From c25c79b468a61ad8a54f764553056e2e2a427ea8 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 29 Sep 2015 16:43:56 +0100 Subject: ASoC: Add SOC_DOUBLE_R_EXT _EXT version of SOC_DOUBLE_R required to allow for custom handlers. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/soc.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..2d79cb55aec5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -217,6 +217,13 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = \ SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert, 0) } +#define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ + xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_volsw, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ + xmax, xinvert) } #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\ xhandler_get, xhandler_put, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ -- cgit v1.2.3 From 58ceb57ec1be928bec2faeca11fe0752f930669d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 30 Sep 2015 22:51:52 +0200 Subject: ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm This patch removes the old PXA DMA API usage and switches over to generic functions provided by snd-soc-dmaengine-pcm. More cleanups may be done on top of this, and some function stubs can now be removed completetly. However, the intention here was to keep the transition as small as possible. This was tested on the mioa701 pxa27x board. Signed-off-by: Daniel Mack [trivial change from mmp-dma to pxa-dma] Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- include/sound/pxa2xx-lib.h | 1 - sound/arm/pxa2xx-ac97.c | 13 ++- sound/arm/pxa2xx-pcm-lib.c | 201 ++++++++------------------------------------ sound/arm/pxa2xx-pcm.c | 12 +-- sound/arm/pxa2xx-pcm.h | 2 - sound/soc/pxa/pxa2xx-ac97.c | 49 +++++++---- sound/soc/pxa/pxa2xx-i2s.c | 3 + sound/soc/pxa/pxa2xx-pcm.c | 32 ------- 8 files changed, 85 insertions(+), 228 deletions(-) (limited to 'include') diff --git a/include/sound/pxa2xx-lib.h b/include/sound/pxa2xx-lib.h index 56e818e4a1cb..6ef629bde164 100644 --- a/include/sound/pxa2xx-lib.h +++ b/include/sound/pxa2xx-lib.h @@ -12,7 +12,6 @@ extern int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd); extern snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream); -extern void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id); extern int __pxa2xx_pcm_open(struct snd_pcm_substream *substream); extern int __pxa2xx_pcm_close(struct snd_pcm_substream *substream); extern int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 38590b322c54..fbd5dad0c484 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include @@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_reset, }; -static unsigned long pxa2xx_ac97_pcm_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .filter_data = &pxa2xx_ac97_pcm_out_req, }; -static unsigned long pxa2xx_ac97_pcm_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 01f8fdc42b1b..e9b98af6b52c 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -8,6 +8,7 @@ #include #include #include +#include #include #include @@ -15,8 +16,6 @@ #include #include -#include - #include "pxa2xx-pcm.h" static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { @@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { .period_bytes_min = 32, .period_bytes_max = 8192 - 32, .periods_min = 1, - .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), + .periods_max = 256, .buffer_bytes_max = 128 * 1024, .fifo_size = 32, }; @@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - size_t totsize = params_buffer_bytes(params); - size_t period = params_period_bytes(params); - pxa_dma_desc *dma_desc; - dma_addr_t dma_buff_phys, next_desc_phys; - u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_dmaengine_dai_dma_data *dma_params; + struct dma_slave_config config; + int ret; - /* temporary transition hack */ - switch (rtd->params->addr_width) { - case DMA_SLAVE_BUSWIDTH_1_BYTE: - dcmd |= DCMD_WIDTH1; - break; - case DMA_SLAVE_BUSWIDTH_2_BYTES: - dcmd |= DCMD_WIDTH2; - break; - case DMA_SLAVE_BUSWIDTH_4_BYTES: - dcmd |= DCMD_WIDTH4; - break; - default: - /* can't happen */ - break; - } + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; - switch (rtd->params->maxburst) { - case 8: - dcmd |= DCMD_BURST8; - break; - case 16: - dcmd |= DCMD_BURST16; - break; - case 32: - dcmd |= DCMD_BURST32; - break; - } + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = totsize; + snd_dmaengine_pcm_set_config_from_dai_data(substream, + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream), + &config); - dma_desc = rtd->dma_desc_array; - next_desc_phys = rtd->dma_desc_array_phys; - dma_buff_phys = runtime->dma_addr; - do { - next_desc_phys += sizeof(pxa_dma_desc); - dma_desc->ddadr = next_desc_phys; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_desc->dsadr = dma_buff_phys; - dma_desc->dtadr = rtd->params->addr; - } else { - dma_desc->dsadr = rtd->params->addr; - dma_desc->dtadr = dma_buff_phys; - } - if (period > totsize) - period = totsize; - dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN; - dma_desc++; - dma_buff_phys += period; - } while (totsize -= period); - dma_desc[-1].ddadr = rtd->dma_desc_array_phys; + ret = dmaengine_slave_config(chan, &config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; } @@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params); int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - - if (rtd && rtd->params && rtd->params->filter_data) { - unsigned long req = *(unsigned long *) rtd->params->filter_data; - DRCMR(req) = 0; - } - snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free); int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) = DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - break; - - case SNDRV_PCM_TRIGGER_RESUME: - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; - DCSR(prtd->dma_ch) |= DCSR_RUN; - break; - - default: - ret = -EINVAL; - } - - return ret; + return snd_dmaengine_pcm_trigger(substream, cmd); } EXPORT_SYMBOL(pxa2xx_pcm_trigger); snd_pcm_uframes_t pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; - - dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); - snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); - - if (x == runtime->buffer_size) - x = 0; - return x; + return snd_dmaengine_pcm_pointer(substream); } EXPORT_SYMBOL(pxa2xx_pcm_pointer); int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - unsigned long req; - - if (!prtd || !prtd->params) - return 0; - - if (prtd->dma_ch == -1) - return -EINVAL; - - DCSR(prtd->dma_ch) &= ~DCSR_RUN; - DCSR(prtd->dma_ch) = 0; - DCMD(prtd->dma_ch) = 0; - req = *(unsigned long *) prtd->params->filter_data; - DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD; - return 0; } EXPORT_SYMBOL(__pxa2xx_pcm_prepare); -void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) -{ - struct snd_pcm_substream *substream = dev_id; - int dcsr; - - dcsr = DCSR(dma_ch); - DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; - - if (dcsr & DCSR_ENDINTR) { - snd_pcm_period_elapsed(substream); - } else { - printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n", - dma_ch, dcsr); - snd_pcm_stop_xrun(substream); - } -} -EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); - int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd; + struct snd_dmaengine_dai_dma_data *dma_params; int ret; runtime->hw = pxa2xx_pcm_hardware; + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + /* * For mysterious reasons (and despite what the manual says) * playback samples are lost if the DMA count is not a multiple @@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); if (ret) - goto out; + return ret; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) - goto out; - - ret = -ENOMEM; - rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); - if (!rtd) - goto out; - rtd->dma_desc_array = - dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, - &rtd->dma_desc_array_phys, GFP_KERNEL); - if (!rtd->dma_desc_array) - goto err1; + return ret; - rtd->dma_ch = -1; - runtime->private_data = rtd; - return 0; - - err1: - kfree(rtd); - out: - return ret; + return snd_dmaengine_pcm_open_request_chan(substream, + pxad_filter_fn, + dma_params->filter_data); } EXPORT_SYMBOL(__pxa2xx_pcm_open); int __pxa2xx_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *rtd = runtime->private_data; - - dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, - rtd->dma_desc_array, rtd->dma_desc_array_phys); - kfree(rtd); - return 0; + return snd_dmaengine_pcm_close_release_chan(substream); } EXPORT_SYMBOL(__pxa2xx_pcm_close); diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 83be8e3f095e..83fcfac97739 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -46,17 +46,13 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? client->playback_params : client->capture_params; - ret = pxa_request_dma("dma", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - goto err2; - rtd->dma_ch = ret; ret = client->startup(substream); if (!ret) - goto out; + goto err2; + + return 0; - pxa_free_dma(rtd->dma_ch); err2: __pxa2xx_pcm_close(substream); out: @@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) { struct pxa2xx_pcm_client *client = substream->private_data; - struct pxa2xx_runtime_data *rtd = substream->runtime->private_data; - pxa_free_dma(rtd->dma_ch); client->shutdown(substream); return __pxa2xx_pcm_close(substream); diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 00330985beec..8fa2b7c9e6b8 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -13,8 +13,6 @@ struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - struct pxa_dma_desc *dma_desc_array; - dma_addr_t dma_desc_array_phys; }; struct pxa2xx_pcm_client { diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9e4b04e0fbd1..f3de615aacd7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include @@ -49,7 +50,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; +static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 11, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +62,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; +static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 12, +}; + static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -65,7 +74,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 10, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -73,7 +85,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9; +static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 9, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -81,7 +96,10 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8; +static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { + .prio = PXAD_PRIO_LOWEST, + .drcmr = 8, +}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, @@ -89,9 +107,8 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -105,9 +122,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_aux_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -121,9 +137,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *cpu_dai) +static int pxa2xx_ac97_mic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -139,15 +154,15 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_48000) static const struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = { - .hw_params = pxa2xx_ac97_hw_params, + .startup = pxa2xx_ac97_hifi_startup, }; static const struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = { - .hw_params = pxa2xx_ac97_hw_aux_params, + .startup = pxa2xx_ac97_aux_startup, }; static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { - .hw_params = pxa2xx_ac97_hw_mic_params, + .startup = pxa2xx_ac97_mic_startup, }; /* diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b4e40036910..0389cf7b4b1e 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -319,6 +319,9 @@ static int pxa2xx_i2s_probe(struct snd_soc_dai *dai) /* Along with FIFO servicing */ SAIMR &= ~(SAIMR_RFS | SAIMR_TFS); + snd_soc_dai_init_dma_data(dai, &pxa2xx_i2s_pcm_stereo_out, + &pxa2xx_i2s_pcm_stereo_in); + return 0; } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 29a3fdbb7b59..9f390398d518 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,8 +15,6 @@ #include #include -#include - #include #include #include @@ -27,11 +25,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma; - int ret; dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -40,40 +35,13 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, if (!dma) return 0; - /* this may get called several times by oss emulation - * with different params */ - if (prtd->params == NULL) { - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } else if (prtd->params != dma) { - pxa_free_dma(prtd->dma_ch); - prtd->params = dma; - ret = pxa_request_dma("name", DMA_PRIO_LOW, - pxa2xx_pcm_dma_irq, substream); - if (ret < 0) - return ret; - prtd->dma_ch = ret; - } - return __pxa2xx_pcm_hw_params(substream, params); } static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) { - struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; - __pxa2xx_pcm_hw_free(substream); - if (prtd->dma_ch >= 0) { - pxa_free_dma(prtd->dma_ch); - prtd->dma_ch = -1; - prtd->params = NULL; - } - return 0; } -- cgit v1.2.3 From e5e0c3dd257bf34cf001e10422943f90437f0f1b Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Thu, 1 Oct 2015 22:02:17 +0900 Subject: ALSA: firewire-tascam: add hwdep interface This commit adds hwdep interface so as the other IEEE 1394 sound devices has. This interface is designed for mixer/control applications. By using this interface, an application can get information about firewire node, can lock/unlock kernel streaming and can get notification at starting/stopping kernel streaming. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 1 + sound/firewire/Kconfig | 1 + sound/firewire/tascam/Makefile | 2 +- sound/firewire/tascam/tascam-hwdep.c | 201 ++++++++++++++++++++++++++++++++++ sound/firewire/tascam/tascam-pcm.c | 17 ++- sound/firewire/tascam/tascam-stream.c | 39 +++++++ sound/firewire/tascam/tascam.c | 6 + sound/firewire/tascam/tascam.h | 13 +++ 9 files changed, 278 insertions(+), 5 deletions(-) create mode 100644 sound/firewire/tascam/tascam-hwdep.c (limited to 'include') diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index aa329132f6c4..a82108e5d1c0 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -101,9 +101,10 @@ enum { SNDRV_HWDEP_IFACE_FW_BEBOB, /* BridgeCo BeBoB based device */ SNDRV_HWDEP_IFACE_FW_OXFW, /* Oxford OXFW970/971 based device */ SNDRV_HWDEP_IFACE_FW_DIGI00X, /* Digidesign Digi 002/003 family */ + SNDRV_HWDEP_IFACE_FW_TASCAM, /* TASCAM FireWire series */ /* Don't forget to change the following: */ - SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_DIGI00X + SNDRV_HWDEP_IFACE_LAST = SNDRV_HWDEP_IFACE_FW_TASCAM }; struct snd_hwdep_info { diff --git a/include/uapi/sound/firewire.h b/include/uapi/sound/firewire.h index deb041cb9af0..db79a12fcc78 100644 --- a/include/uapi/sound/firewire.h +++ b/include/uapi/sound/firewire.h @@ -64,6 +64,7 @@ union snd_firewire_event { #define SNDRV_FIREWIRE_TYPE_BEBOB 3 #define SNDRV_FIREWIRE_TYPE_OXFW 4 #define SNDRV_FIREWIRE_TYPE_DIGI00X 5 +#define SNDRV_FIREWIRE_TYPE_TASCAM 6 /* RME, MOTU, ... */ struct snd_firewire_get_info { diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index bb3f2610c6e9..bee0e5f1a116 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -138,6 +138,7 @@ config SND_FIREWIRE_DIGI00X config SND_FIREWIRE_TASCAM tristate "TASCAM FireWire series support" select SND_FIREWIRE_LIB + select SND_HWDEP help Say Y here to include support for TASCAM. * FW-1884 diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile index f075b9bb723f..6beefc2ae8b0 100644 --- a/sound/firewire/tascam/Makefile +++ b/sound/firewire/tascam/Makefile @@ -1,3 +1,3 @@ snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \ - tascam-pcm.o tascam.o + tascam-pcm.o tascam-hwdep.o tascam.o obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c new file mode 100644 index 000000000000..131267c3a042 --- /dev/null +++ b/sound/firewire/tascam/tascam-hwdep.c @@ -0,0 +1,201 @@ +/* + * tascam-hwdep.c - a part of driver for TASCAM FireWire series + * + * Copyright (c) 2015 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +/* + * This codes give three functionality. + * + * 1.get firewire node information + * 2.get notification about starting/stopping stream + * 3.lock/unlock stream + */ + +#include "tascam.h" + +static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf, + long count) +{ + union snd_firewire_event event; + + memset(&event, 0, sizeof(event)); + + event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; + event.lock_status.status = (tscm->dev_lock_count > 0); + tscm->dev_lock_changed = false; + + count = min_t(long, count, sizeof(event.lock_status)); + + if (copy_to_user(buf, &event, count)) + return -EFAULT; + + return count; +} + +static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, + loff_t *offset) +{ + struct snd_tscm *tscm = hwdep->private_data; + DEFINE_WAIT(wait); + union snd_firewire_event event; + + spin_lock_irq(&tscm->lock); + + while (!tscm->dev_lock_changed) { + prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE); + spin_unlock_irq(&tscm->lock); + schedule(); + finish_wait(&tscm->hwdep_wait, &wait); + if (signal_pending(current)) + return -ERESTARTSYS; + spin_lock_irq(&tscm->lock); + } + + memset(&event, 0, sizeof(event)); + count = hwdep_read_locked(tscm, buf, count); + spin_unlock_irq(&tscm->lock); + + return count; +} + +static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file, + poll_table *wait) +{ + struct snd_tscm *tscm = hwdep->private_data; + unsigned int events; + + poll_wait(file, &tscm->hwdep_wait, wait); + + spin_lock_irq(&tscm->lock); + if (tscm->dev_lock_changed) + events = POLLIN | POLLRDNORM; + else + events = 0; + spin_unlock_irq(&tscm->lock); + + return events; +} + +static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg) +{ + struct fw_device *dev = fw_parent_device(tscm->unit); + struct snd_firewire_get_info info; + + memset(&info, 0, sizeof(info)); + info.type = SNDRV_FIREWIRE_TYPE_TASCAM; + info.card = dev->card->index; + *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]); + *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]); + strlcpy(info.device_name, dev_name(&dev->device), + sizeof(info.device_name)); + + if (copy_to_user(arg, &info, sizeof(info))) + return -EFAULT; + + return 0; +} + +static int hwdep_lock(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + if (tscm->dev_lock_count == 0) { + tscm->dev_lock_count = -1; + err = 0; + } else { + err = -EBUSY; + } + + spin_unlock_irq(&tscm->lock); + + return err; +} + +static int hwdep_unlock(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + if (tscm->dev_lock_count == -1) { + tscm->dev_lock_count = 0; + err = 0; + } else { + err = -EBADFD; + } + + spin_unlock_irq(&tscm->lock); + + return err; +} + +static int hwdep_release(struct snd_hwdep *hwdep, struct file *file) +{ + struct snd_tscm *tscm = hwdep->private_data; + + spin_lock_irq(&tscm->lock); + if (tscm->dev_lock_count == -1) + tscm->dev_lock_count = 0; + spin_unlock_irq(&tscm->lock); + + return 0; +} + +static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct snd_tscm *tscm = hwdep->private_data; + + switch (cmd) { + case SNDRV_FIREWIRE_IOCTL_GET_INFO: + return hwdep_get_info(tscm, (void __user *)arg); + case SNDRV_FIREWIRE_IOCTL_LOCK: + return hwdep_lock(tscm); + case SNDRV_FIREWIRE_IOCTL_UNLOCK: + return hwdep_unlock(tscm); + default: + return -ENOIOCTLCMD; + } +} + +#ifdef CONFIG_COMPAT +static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file, + unsigned int cmd, unsigned long arg) +{ + return hwdep_ioctl(hwdep, file, cmd, + (unsigned long)compat_ptr(arg)); +} +#else +#define hwdep_compat_ioctl NULL +#endif + +static const struct snd_hwdep_ops hwdep_ops = { + .read = hwdep_read, + .release = hwdep_release, + .poll = hwdep_poll, + .ioctl = hwdep_ioctl, + .ioctl_compat = hwdep_compat_ioctl, +}; + +int snd_tscm_create_hwdep_device(struct snd_tscm *tscm) +{ + struct snd_hwdep *hwdep; + int err; + + err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep); + if (err < 0) + return err; + + strcpy(hwdep->name, "Tascam"); + hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM; + hwdep->ops = hwdep_ops; + hwdep->private_data = tscm; + hwdep->exclusive = true; + + return err; +} diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c index 696b371e3c44..380d3db969a5 100644 --- a/sound/firewire/tascam/tascam-pcm.c +++ b/sound/firewire/tascam/tascam-pcm.c @@ -72,9 +72,13 @@ static int pcm_open(struct snd_pcm_substream *substream) unsigned int rate; int err; + err = snd_tscm_stream_lock_try(tscm); + if (err < 0) + goto end; + err = pcm_init_hw_params(tscm, substream); if (err < 0) - return err; + goto err_locked; err = snd_tscm_stream_get_clock(tscm, &clock); if (clock != SND_TSCM_CLOCK_INTERNAL || @@ -82,18 +86,25 @@ static int pcm_open(struct snd_pcm_substream *substream) amdtp_stream_pcm_running(&tscm->tx_stream)) { err = snd_tscm_stream_get_rate(tscm, &rate); if (err < 0) - return err; + goto err_locked; substream->runtime->hw.rate_min = rate; substream->runtime->hw.rate_max = rate; } snd_pcm_set_sync(substream); - +end: + return err; +err_locked: + snd_tscm_stream_lock_release(tscm); return err; } static int pcm_close(struct snd_pcm_substream *substream) { + struct snd_tscm *tscm = substream->private_data; + + snd_tscm_stream_lock_release(tscm); + return 0; } diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 0732f7f0c736..0e6dd5c61f53 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -455,3 +455,42 @@ void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm) finish_session(tscm); release_resources(tscm); } + +void snd_tscm_stream_lock_changed(struct snd_tscm *tscm) +{ + tscm->dev_lock_changed = true; + wake_up(&tscm->hwdep_wait); +} + +int snd_tscm_stream_lock_try(struct snd_tscm *tscm) +{ + int err; + + spin_lock_irq(&tscm->lock); + + /* user land lock this */ + if (tscm->dev_lock_count < 0) { + err = -EBUSY; + goto end; + } + + /* this is the first time */ + if (tscm->dev_lock_count++ == 0) + snd_tscm_stream_lock_changed(tscm); + err = 0; +end: + spin_unlock_irq(&tscm->lock); + return err; +} + +void snd_tscm_stream_lock_release(struct snd_tscm *tscm) +{ + spin_lock_irq(&tscm->lock); + + if (WARN_ON(tscm->dev_lock_count <= 0)) + goto end; + if (--tscm->dev_lock_count == 0) + snd_tscm_stream_lock_changed(tscm); +end: + spin_unlock_irq(&tscm->lock); +} diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index bb4f70656f2f..ee2f498dcce4 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -110,6 +110,8 @@ static int snd_tscm_probe(struct fw_unit *unit, tscm->spec = (const struct snd_tscm_spec *)entry->driver_data; mutex_init(&tscm->mutex); + spin_lock_init(&tscm->lock); + init_waitqueue_head(&tscm->hwdep_wait); err = check_name(tscm); if (err < 0) @@ -125,6 +127,10 @@ static int snd_tscm_probe(struct fw_unit *unit, if (err < 0) goto error; + err = snd_tscm_create_hwdep_device(tscm); + if (err < 0) + goto error; + err = snd_card_register(card); if (err < 0) goto error; diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 28c875f7808c..75a3b9a81fea 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -23,6 +23,8 @@ #include #include #include +#include +#include #include "../lib.h" #include "../amdtp-stream.h" @@ -44,6 +46,7 @@ struct snd_tscm { struct fw_unit *unit; struct mutex mutex; + spinlock_t lock; const struct snd_tscm_spec *spec; @@ -52,6 +55,10 @@ struct snd_tscm { struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; unsigned int substreams_counter; + + int dev_lock_count; + bool dev_lock_changed; + wait_queue_head_t hwdep_wait; }; #define TSCM_ADDR_BASE 0xffff00000000ull @@ -97,8 +104,14 @@ void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm); int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate); void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm); +void snd_tscm_stream_lock_changed(struct snd_tscm *tscm); +int snd_tscm_stream_lock_try(struct snd_tscm *tscm); +void snd_tscm_stream_lock_release(struct snd_tscm *tscm); + void snd_tscm_proc_init(struct snd_tscm *tscm); int snd_tscm_create_pcm_devices(struct snd_tscm *tscm); +int snd_tscm_create_hwdep_device(struct snd_tscm *tscm); + #endif -- cgit v1.2.3 From 5b2688a59af686f7c0a80edc49d7f190365ac090 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:28:47 +0800 Subject: ASoC: topology: ABI - Add PCM Support and bump ABI version to 4 The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm. This struct will now be used to handle data related to PCMs (FE DAI & DAI links). It's not for BE, because BE DAI mappings will be provided by ACPI/FDT data. Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using snd_soc_tplg_stream and snd_soc_stream_caps instead. Bump ABI version to 4. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 31 +++++++++++++++---------------- sound/soc/soc-topology.c | 4 ++-- 2 files changed, 17 insertions(+), 18 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 247c50bd60f0..2aa081ca95c1 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -83,7 +83,7 @@ #define SND_SOC_TPLG_NUM_TEXTS 16 /* ABI version */ -#define SND_SOC_TPLG_ABI_VERSION 0x3 +#define SND_SOC_TPLG_ABI_VERSION 0x4 /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 @@ -378,30 +378,29 @@ struct snd_soc_tplg_dapm_widget { */ } __attribute__((packed)); -struct snd_soc_tplg_pcm_cfg_caps { - struct snd_soc_tplg_stream_caps caps; - struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX]; - __le32 num_configs; /* number of configs */ -} __attribute__((packed)); /* - * Describes SW/FW specific features of PCM or DAI link. + * Describes SW/FW specific features of PCM (FE DAI & DAI link). * - * File block representation for PCM/DAI-Link :- + * File block representation for PCM :- * +-----------------------------------+-----+ * | struct snd_soc_tplg_hdr | 1 | * +-----------------------------------+-----+ - * | struct snd_soc_tplg_dapm_pcm_dai | N | + * | struct snd_soc_tplg_pcm | N | * +-----------------------------------+-----+ */ -struct snd_soc_tplg_pcm_dai { +struct snd_soc_tplg_pcm { __le32 size; /* in bytes of this structure */ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le32 id; /* unique ID - used to match */ - __le32 playback; /* supports playback mode */ - __le32 capture; /* supports capture mode */ - __le32 compress; /* 1 = compressed; 0 = PCM */ - struct snd_soc_tplg_pcm_cfg_caps capconf[2]; /* capabilities and configs */ + char pcm_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + __le32 pcm_id; /* unique ID - used to match */ + __le32 dai_id; /* unique ID - used to match */ + __le32 playback; /* supports playback mode */ + __le32 capture; /* supports capture mode */ + __le32 compress; /* 1 = compressed; 0 = PCM */ + struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */ + __le32 num_streams; /* number of streams */ + struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */ } __attribute__((packed)); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 69d01cd925ce..8d7ec80af51b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1558,7 +1558,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, pcm_dai = (struct snd_soc_tplg_pcm_dai *)tplg->pos; if (soc_tplg_check_elem_count(tplg, - sizeof(struct snd_soc_tplg_pcm_dai), count, + sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n", count); @@ -1566,7 +1566,7 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, } dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); - tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; + tplg->pos += sizeof(struct snd_soc_tplg_pcm) * count; dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL); if (dobj == NULL) -- cgit v1.2.3 From 7654855ef84d78079109bb38195a8db6b188117b Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:29:47 +0800 Subject: ASoC: topology: ABI - Remove unused struct snd_soc_tplg_stream_config The struct snd_soc_tplg_stream_config is no longer used in the ABI. We are using snd_soc_tplg_stream instead. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 10 ---------- 1 file changed, 10 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 2aa081ca95c1..4bef63f5e1ff 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -226,16 +226,6 @@ struct snd_soc_tplg_stream { __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ } __attribute__((packed)); -/* - * Duplex stream configuration supported by SW/FW. - */ -struct snd_soc_tplg_stream_config { - __le32 size; /* in bytes of this structure */ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - struct snd_soc_tplg_stream playback; - struct snd_soc_tplg_stream capture; -} __attribute__((packed)); - /* * Manifest. List totals for each payload type. Not used in parsing, but will * be passed to the component driver before any other objects in order for any -- cgit v1.2.3 From e8b3fe8f383bbf055cbd69b776fcbbb5ed3befcd Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:30:05 +0800 Subject: ASoC: topology: ABI - Use __le32 instead of __u32 in snd_soc_tplg_dapm_widget This fixes the endianness of the ABI parameters in the struct. The field 'num_kcontrols' is also extended from 16 bits to 32 bits. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 4bef63f5e1ff..88210a8e450f 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -356,11 +356,11 @@ struct snd_soc_tplg_dapm_widget { __le32 shift; /* bits to shift */ __le32 mask; /* non-shifted mask */ __le32 subseq; /* sort within widget type */ - __u32 invert; /* invert the power bit */ - __u32 ignore_suspend; /* kept enabled over suspend */ - __u16 event_flags; - __u16 event_type; - __u16 num_kcontrols; + __le32 invert; /* invert the power bit */ + __le32 ignore_suspend; /* kept enabled over suspend */ + __le16 event_flags; + __le16 event_type; + __le32 num_kcontrols; struct snd_soc_tplg_private priv; /* * kcontrols that relate to this widget -- cgit v1.2.3 From 731324f5cee3caf230427f754701212961fe0bb1 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:31:34 +0800 Subject: ASoC: topology: ABI - Add name element to snd_soc_tplg_stream For codec-codec links, this struct will be mapped to the DAI links's params, which is struct snd_soc_pcm_stream and it needs a stream name. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 88210a8e450f..218148000187 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -217,6 +217,7 @@ struct snd_soc_tplg_stream_caps { */ struct snd_soc_tplg_stream { __le32 size; /* in bytes of this structure */ + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* Name of the stream */ __le64 format; /* SNDRV_PCM_FMTBIT_* */ __le32 rate; /* SNDRV_PCM_RATE_* */ __le32 period_bytes; /* size of period in bytes */ -- cgit v1.2.3 From 478d0d39719127c42f21ac85b8719a87b8fa7bd7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 30 Sep 2015 17:31:44 +0800 Subject: ASoC: topology: ABI - Change stream formats to a bitwise flag The toplogy user space tool will generate this bitwise flag by using SNDRV_PCM_FORMAT_* exposed by asound.h, and the topology core will copy this flag when generating DAI streams. Signed-off-by: Mengdong Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 218148000187..eeb74a9cc2f8 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -198,7 +198,7 @@ struct snd_soc_tplg_ctl_hdr { struct snd_soc_tplg_stream_caps { __le32 size; /* in bytes of this structure */ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le64 formats; /* supported formats SNDRV_PCM_FMTBIT_* */ __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ __le32 rate_min; /* min rate */ __le32 rate_max; /* max rate */ -- cgit v1.2.3 From 7c545b327d54cdc3f693093f744459f6e0d8ce58 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:32:04 +0800 Subject: ASoC: topology: ABI - Add the type for BE DAI link Define the topology type for BE DAI link: SND_SOC_TPLG_TYPE_BACKEND_LINK. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index eeb74a9cc2f8..f75fd29bf580 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -103,7 +103,8 @@ #define SND_SOC_TPLG_TYPE_PCM 7 #define SND_SOC_TPLG_TYPE_MANIFEST 8 #define SND_SOC_TPLG_TYPE_CODEC_LINK 9 -#define SND_SOC_TPLG_TYPE_PDATA 10 +#define SND_SOC_TPLG_TYPE_BACKEND_LINK 10 +#define SND_SOC_TPLG_TYPE_PDATA 11 #define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA /* vendor block IDs - please add new vendor types to end */ -- cgit v1.2.3 From e2a9df656f28e42538562d090e2b6f3dec5b41f2 Mon Sep 17 00:00:00 2001 From: Vedang Patel Date: Wed, 30 Sep 2015 17:32:22 +0800 Subject: ASoC: topology: ABI - Add configuration for BE & Codec-Codec DAI Links struct snd_soc_tplg_link_config is defined to configure BE & CC links. Signed-off-by: Vedang Patel Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index f75fd29bf580..538ea1308008 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -395,4 +395,21 @@ struct snd_soc_tplg_pcm { struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */ } __attribute__((packed)); + +/* + * Describes the BE or CC link runtime supported configs or params + * + * File block representation for BE/CC link config :- + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_hdr | 1 | + * +-----------------------------------+-----+ + * | struct snd_soc_tplg_link_config | N | + * +-----------------------------------+-----+ + */ +struct snd_soc_tplg_link_config { + __le32 size; /* in bytes of this structure */ + __le32 id; /* unique ID - used to match */ + struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */ + __le32 num_streams; /* number of streams */ +} __attribute__((packed)); #endif -- cgit v1.2.3 From 76a822a6ae9c0c67309318bb60a4117329252fc4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 30 Sep 2015 17:32:40 +0800 Subject: ASoC: topology: ABI - Remove tdm_slot & dai_fmt from snd_soc_tplg_stream These two fields are line parameters for BE/CC links and should not be from toplogy but from ACPI. Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- include/uapi/sound/asoc.h | 2 -- 1 file changed, 2 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 538ea1308008..26539a7e4880 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -224,8 +224,6 @@ struct snd_soc_tplg_stream { __le32 period_bytes; /* size of period in bytes */ __le32 buffer_bytes; /* size of buffer in bytes */ __le32 channels; /* channels */ - __le32 tdm_slot; /* optional BE bitmask of supported TDM slots */ - __le32 dai_fmt; /* SND_SOC_DAIFMT_ */ } __attribute__((packed)); /* -- cgit v1.2.3 From 6d817c0e9fd7536be76690bfdee88e8a81c16f7d Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Tue, 29 Sep 2015 16:44:01 +0100 Subject: ASoC: codecs: Add da7219 codec driver This adds support for the DA7219 audio codec with built-in advanced accessory detect features. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7219-aad.h | 99 +++ include/sound/da7219.h | 55 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++++++++++ sound/soc/codecs/da7219-aad.h | 212 +++++ sound/soc/codecs/da7219.c | 1940 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++++++++++ 8 files changed, 3955 insertions(+) create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h (limited to 'include') diff --git a/include/sound/da7219-aad.h b/include/sound/da7219-aad.h new file mode 100644 index 000000000000..17802fb86ec4 --- /dev/null +++ b/include/sound/da7219-aad.h @@ -0,0 +1,99 @@ +/* + * da7219-aad.h - DA7322 ASoC Codec AAD Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_AAD_PDATA_H +#define __DA7219_AAD_PDATA_H + +enum da7219_aad_micbias_pulse_lvl { + DA7219_AAD_MICBIAS_PULSE_LVL_OFF = 0, + DA7219_AAD_MICBIAS_PULSE_LVL_2_8V = 6, + DA7219_AAD_MICBIAS_PULSE_LVL_2_9V, +}; + +enum da7219_aad_btn_cfg { + DA7219_AAD_BTN_CFG_2MS = 1, + DA7219_AAD_BTN_CFG_5MS, + DA7219_AAD_BTN_CFG_10MS, + DA7219_AAD_BTN_CFG_50MS, + DA7219_AAD_BTN_CFG_100MS, + DA7219_AAD_BTN_CFG_200MS, + DA7219_AAD_BTN_CFG_500MS, +}; + +enum da7219_aad_mic_det_thr { + DA7219_AAD_MIC_DET_THR_200_OHMS = 0, + DA7219_AAD_MIC_DET_THR_500_OHMS, + DA7219_AAD_MIC_DET_THR_750_OHMS, + DA7219_AAD_MIC_DET_THR_1000_OHMS, +}; + +enum da7219_aad_jack_ins_deb { + DA7219_AAD_JACK_INS_DEB_5MS = 0, + DA7219_AAD_JACK_INS_DEB_10MS, + DA7219_AAD_JACK_INS_DEB_20MS, + DA7219_AAD_JACK_INS_DEB_50MS, + DA7219_AAD_JACK_INS_DEB_100MS, + DA7219_AAD_JACK_INS_DEB_200MS, + DA7219_AAD_JACK_INS_DEB_500MS, + DA7219_AAD_JACK_INS_DEB_1S, +}; + +enum da7219_aad_jack_det_rate { + DA7219_AAD_JACK_DET_RATE_32_64MS = 0, + DA7219_AAD_JACK_DET_RATE_64_128MS, + DA7219_AAD_JACK_DET_RATE_128_256MS, + DA7219_AAD_JACK_DET_RATE_256_512MS, +}; + +enum da7219_aad_jack_rem_deb { + DA7219_AAD_JACK_REM_DEB_1MS = 0, + DA7219_AAD_JACK_REM_DEB_5MS, + DA7219_AAD_JACK_REM_DEB_10MS, + DA7219_AAD_JACK_REM_DEB_20MS, +}; + +enum da7219_aad_btn_avg { + DA7219_AAD_BTN_AVG_1 = 0, + DA7219_AAD_BTN_AVG_2, + DA7219_AAD_BTN_AVG_4, + DA7219_AAD_BTN_AVG_8, +}; + +enum da7219_aad_adc_1bit_rpt { + DA7219_AAD_ADC_1BIT_RPT_1 = 0, + DA7219_AAD_ADC_1BIT_RPT_2, + DA7219_AAD_ADC_1BIT_RPT_4, + DA7219_AAD_ADC_1BIT_RPT_8, +}; + +struct da7219_aad_pdata { + int irq; + + enum da7219_aad_micbias_pulse_lvl micbias_pulse_lvl; + u32 micbias_pulse_time; + enum da7219_aad_btn_cfg btn_cfg; + enum da7219_aad_mic_det_thr mic_det_thr; + enum da7219_aad_jack_ins_deb jack_ins_deb; + enum da7219_aad_jack_det_rate jack_det_rate; + enum da7219_aad_jack_rem_deb jack_rem_deb; + + u8 a_d_btn_thr; + u8 d_b_btn_thr; + u8 b_c_btn_thr; + u8 c_mic_btn_thr; + + enum da7219_aad_btn_avg btn_avg; + enum da7219_aad_adc_1bit_rpt adc_1bit_rpt; +}; + +#endif /* __DA7219_AAD_PDATA_H */ diff --git a/include/sound/da7219.h b/include/sound/da7219.h new file mode 100644 index 000000000000..3f39e135312d --- /dev/null +++ b/include/sound/da7219.h @@ -0,0 +1,55 @@ +/* + * da7219.h - DA7219 ASoC Codec Driver Platform Data + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_PDATA_H +#define __DA7219_PDATA_H + +/* LDO */ +enum da7219_ldo_lvl_sel { + DA7219_LDO_LVL_SEL_1_05V = 0, + DA7219_LDO_LVL_SEL_1_10V, + DA7219_LDO_LVL_SEL_1_20V, + DA7219_LDO_LVL_SEL_1_40V, +}; + +/* Mic Bias */ +enum da7219_micbias_voltage { + DA7219_MICBIAS_1_8V = 1, + DA7219_MICBIAS_2_0V, + DA7219_MICBIAS_2_2V, + DA7219_MICBIAS_2_4V, + DA7219_MICBIAS_2_6V, +}; + +/* Mic input type */ +enum da7219_mic_amp_in_sel { + DA7219_MIC_AMP_IN_SEL_DIFF = 0, + DA7219_MIC_AMP_IN_SEL_SE_P, + DA7219_MIC_AMP_IN_SEL_SE_N, +}; + +struct da7219_aad_pdata; + +struct da7219_pdata { + /* Internal LDO */ + enum da7219_ldo_lvl_sel ldo_lvl_sel; + + /* Mic */ + enum da7219_micbias_voltage micbias_lvl; + enum da7219_mic_amp_in_sel mic_amp_in_sel; + + /* AAD */ + struct da7219_aad_pdata *aad_pdata; +}; + +#endif /* __DA7219_PDATA_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733ecd17f..c9a895cc4eff 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C + select SND_SOC_DA7219 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_DMIC @@ -430,6 +431,9 @@ config SND_SOC_DA7210 config SND_SOC_DA7213 tristate +config SND_SOC_DA7219 + tristate + config SND_SOC_DA732X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077954ae..eb10a34d5e13 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-cs4349-objs := cs4349.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o +snd-soc-da7219-objs := da7219.o da7219-aad.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o @@ -241,6 +242,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o +obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c new file mode 100644 index 000000000000..9459593eef13 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.c @@ -0,0 +1,823 @@ +/* + * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson + * + * 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * Detection control + */ + +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219->aad->jack = jack; + da7219->aad->jack_inserted = false; + + /* Send an initial empty report */ + snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK); + + /* Enable/Disable jack detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_ACCDET_EN_MASK, + (jack ? DA7219_ACCDET_EN_MASK : 0)); +} +EXPORT_SYMBOL_GPL(da7219_aad_jack_det); + +/* + * Button/HPTest work + */ + +static void da7219_aad_btn_det_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, btn_det_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + u8 statusa, micbias_ctrl; + bool micbias_up = false; + int retries = 0; + + /* Drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + /* Make sure mic bias is up */ + snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + do { + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + if (statusa & DA7219_MICBIAS_UP_STS_MASK) + micbias_up = true; + else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES) + msleep(DA7219_AAD_MICBIAS_CHK_DELAY); + } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES)); + + if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) + dev_warn(codec->dev, "Mic bias status check timed out"); + + /* + * Mic bias pulse required to enable mic, must be done before enabling + * button detection to prevent erroneous button readings. + */ + if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) { + /* Pulse higher level voltage */ + micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL); + snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_LEVEL_MASK, + da7219_aad->micbias_pulse_lvl); + msleep(da7219_aad->micbias_pulse_time); + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl); + + } + + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, + da7219_aad->btn_cfg); +} + +static void da7219_aad_hptest_work(struct work_struct *work) +{ + struct da7219_aad_priv *da7219_aad = + container_of(work, struct da7219_aad_priv, hptest_work); + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u16 tonegen_freq_hptest; + u8 accdet_cfg8; + int report = 0; + + /* Lock DAPM and any Kcontrols that are affected by this test */ + snd_soc_dapm_mutex_lock(dapm); + mutex_lock(&da7219->lock); + + /* Bypass cache so it saves current settings */ + regcache_cache_bypass(da7219->regmap, true); + + /* Make sure Tone Generator is disabled */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + /* Enable HPTest block, 1KOhms check */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK, + DA7219_HPTEST_EN_MASK | + DA7219_HPTEST_RES_SEL_1KOHMS); + + /* Set gains to 0db */ + snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB); + snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB); + + /* Disable DAC filters, EQs and soft mute */ + snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK, + 0); + snd_soc_update_bits(codec, DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_MASK, 0); + + /* Enable HP left & right paths */ + snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK, + DA7219_CP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK, + DA7219_DAC_L_SRC_TONEGEN | + DA7219_DAC_R_SRC_TONEGEN); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, + DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK, + DA7219_DAC_L_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, + DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK, + DA7219_DAC_R_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_MASK, + DA7219_MIXOUT_L_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_MASK, + DA7219_MIXOUT_R_MIX_SELECT_MASK); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_OUTFILT_ST_1L_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1L); + snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R, + DA7219_OUTFILT_ST_1R_SRC_MASK, + DA7219_DMIX_ST_SRC_OUTFILT1R); + snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_MASK, + DA7219_MIXOUT_L_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_MASK, + DA7219_MIXOUT_R_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK); + snd_soc_write(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK); + + /* Configure & start Tone Generator */ + snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK); + tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ); + regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + &tonegen_freq_hptest, sizeof(tonegen_freq_hptest)); + snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2, + DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK, + DA7219_SWG_SEL_SRAMP | + DA7219_TONE_GEN_GAIN_MINUS_15DB); + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Grab comparator reading */ + accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8); + if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK) + report |= SND_JACK_HEADPHONE; + else + report |= SND_JACK_LINEOUT; + + /* Stop tone generator */ + snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0); + + msleep(DA7219_AAD_HPTEST_PERIOD); + + /* Restore original settings from cache */ + regcache_mark_dirty(da7219->regmap); + regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L, + DA7219_DROUTING_ST_OUTFILT_1R); + regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_R_SELECT); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC, + DA7219_DIG_ROUTING_DAC); + regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5, + DA7219_DAC_FILTERS5); + regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4, + DA7219_DAC_FILTERS1); + regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER, + DA7219_TONE_GEN_ON_PER); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L, + DA7219_TONE_GEN_FREQ1_U); + regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1, + DA7219_TONE_GEN_CFG2); + + regcache_cache_bypass(da7219->regmap, false); + + /* Disable HPTest block */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8, + DA7219_HPTEST_EN_MASK, 0); + + /* Drive Headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK, + DA7219_HP_L_AMP_OE_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, + DA7219_HP_R_AMP_OE_MASK); + + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + + /* + * Only send report if jack hasn't been removed during process, + * otherwise it's invalid and we drop it. + */ + if (da7219_aad->jack_inserted) + snd_soc_jack_report(da7219_aad->jack, report, + SND_JACK_HEADSET | SND_JACK_LINEOUT); +} + + +/* + * IRQ + */ + +static irqreturn_t da7219_aad_irq_thread(int irq, void *data) +{ + struct da7219_aad_priv *da7219_aad = data; + struct snd_soc_codec *codec = da7219_aad->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 events[DA7219_AAD_IRQ_REG_MAX]; + u8 statusa; + int i, report = 0, mask = 0; + + /* Read current IRQ events */ + regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B]) + return IRQ_NONE; + + /* Read status register for jack insertion & type status */ + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); + + /* Clear events */ + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A, + events, DA7219_AAD_IRQ_REG_MAX); + + dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n", + events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B], + statusa); + + if (statusa & DA7219_JACK_INSERTION_STS_MASK) { + /* Jack Insertion */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_INSERTED_MASK) { + report |= SND_JACK_MECHANICAL; + mask |= SND_JACK_MECHANICAL; + da7219_aad->jack_inserted = true; + } + + /* Jack type detection */ + if (events[DA7219_AAD_IRQ_REG_A] & + DA7219_E_JACK_DETECT_COMPLETE_MASK) { + /* + * If 4-pole, then enable button detection, else perform + * HP impedance test to determine output type to report. + * + * We schedule work here as the tasks themselves can + * take time to complete, and in particular for hptest + * we want to be able to check if the jack was removed + * during the procedure as this will invalidate the + * result. By doing this as work, the IRQ thread can + * handle a removal, and we can check at the end of + * hptest if we have a valid result or not. + */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + report |= SND_JACK_HEADSET; + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; + schedule_work(&da7219_aad->btn_det_work); + } else { + schedule_work(&da7219_aad->hptest_work); + } + } + + /* Button support for 4-pole jack */ + if (statusa & DA7219_JACK_TYPE_STS_MASK) { + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Press */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_PRESSED_MASK << i)) { + report |= SND_JACK_BTN_0 >> i; + mask |= SND_JACK_BTN_0 >> i; + } + } + snd_soc_jack_report(da7219_aad->jack, report, mask); + + for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) { + /* Button Release */ + if (events[DA7219_AAD_IRQ_REG_B] & + (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) { + report &= ~(SND_JACK_BTN_0 >> i); + mask |= SND_JACK_BTN_0 >> i; + } + } + } + } else { + /* Jack removal */ + if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) { + report = 0; + mask |= DA7219_AAD_REPORT_ALL_MASK; + da7219_aad->jack_inserted = false; + + /* Un-drive headphones/lineout */ + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_OE_MASK, 0); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_OE_MASK, 0); + + /* Ensure button detection disabled */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + /* Disable mic bias */ + snd_soc_dapm_disable_pin(dapm, "Mic Bias"); + snd_soc_dapm_sync(dapm); + + /* Cancel any pending work */ + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); + } + } + + snd_soc_jack_report(da7219_aad->jack, report, mask); + + return IRQ_HANDLED; +} + +/* + * DT to pdata conversion + */ + +static enum da7219_aad_micbias_pulse_lvl + da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2800: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V; + case 2900: + return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V; + default: + dev_warn(codec->dev, "Invalid micbias pulse level"); + return DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + } +} + +static enum da7219_aad_btn_cfg + da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 2: + return DA7219_AAD_BTN_CFG_2MS; + case 5: + return DA7219_AAD_BTN_CFG_5MS; + case 10: + return DA7219_AAD_BTN_CFG_10MS; + case 50: + return DA7219_AAD_BTN_CFG_50MS; + case 100: + return DA7219_AAD_BTN_CFG_100MS; + case 200: + return DA7219_AAD_BTN_CFG_200MS; + case 500: + return DA7219_AAD_BTN_CFG_500MS; + default: + dev_warn(codec->dev, "Invalid button config"); + return DA7219_AAD_BTN_CFG_10MS; + } +} + +static enum da7219_aad_mic_det_thr + da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 200: + return DA7219_AAD_MIC_DET_THR_200_OHMS; + case 500: + return DA7219_AAD_MIC_DET_THR_500_OHMS; + case 750: + return DA7219_AAD_MIC_DET_THR_750_OHMS; + case 1000: + return DA7219_AAD_MIC_DET_THR_1000_OHMS; + default: + dev_warn(codec->dev, "Invalid mic detect threshold"); + return DA7219_AAD_MIC_DET_THR_500_OHMS; + } +} + +static enum da7219_aad_jack_ins_deb + da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 5: + return DA7219_AAD_JACK_INS_DEB_5MS; + case 10: + return DA7219_AAD_JACK_INS_DEB_10MS; + case 20: + return DA7219_AAD_JACK_INS_DEB_20MS; + case 50: + return DA7219_AAD_JACK_INS_DEB_50MS; + case 100: + return DA7219_AAD_JACK_INS_DEB_100MS; + case 200: + return DA7219_AAD_JACK_INS_DEB_200MS; + case 500: + return DA7219_AAD_JACK_INS_DEB_500MS; + case 1000: + return DA7219_AAD_JACK_INS_DEB_1S; + default: + dev_warn(codec->dev, "Invalid jack insert debounce"); + return DA7219_AAD_JACK_INS_DEB_20MS; + } +} + +static enum da7219_aad_jack_det_rate + da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "32ms_64ms")) { + return DA7219_AAD_JACK_DET_RATE_32_64MS; + } else if (!strcmp(str, "64ms_128ms")) { + return DA7219_AAD_JACK_DET_RATE_64_128MS; + } else if (!strcmp(str, "128ms_256ms")) { + return DA7219_AAD_JACK_DET_RATE_128_256MS; + } else if (!strcmp(str, "256ms_512ms")) { + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } else { + dev_warn(codec->dev, "Invalid jack detect rate"); + return DA7219_AAD_JACK_DET_RATE_256_512MS; + } +} + +static enum da7219_aad_jack_rem_deb + da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_JACK_REM_DEB_1MS; + case 5: + return DA7219_AAD_JACK_REM_DEB_5MS; + case 10: + return DA7219_AAD_JACK_REM_DEB_10MS; + case 20: + return DA7219_AAD_JACK_REM_DEB_20MS; + default: + dev_warn(codec->dev, "Invalid jack removal debounce"); + return DA7219_AAD_JACK_REM_DEB_1MS; + } +} + +static enum da7219_aad_btn_avg + da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_BTN_AVG_1; + case 2: + return DA7219_AAD_BTN_AVG_2; + case 4: + return DA7219_AAD_BTN_AVG_4; + case 8: + return DA7219_AAD_BTN_AVG_8; + default: + dev_warn(codec->dev, "Invalid button average value"); + return DA7219_AAD_BTN_AVG_2; + } +} + +static enum da7219_aad_adc_1bit_rpt + da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1: + return DA7219_AAD_ADC_1BIT_RPT_1; + case 2: + return DA7219_AAD_ADC_1BIT_RPT_2; + case 4: + return DA7219_AAD_ADC_1BIT_RPT_4; + case 8: + return DA7219_AAD_ADC_1BIT_RPT_8; + default: + dev_warn(codec->dev, "Invalid ADC 1-bit repeat value"); + return DA7219_AAD_ADC_1BIT_RPT_1; + } +} + +static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad"); + struct da7219_aad_pdata *aad_pdata; + const char *of_str; + u32 of_val32; + + if (!aad_np) + return NULL; + + aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL); + if (!aad_pdata) + goto out; + + aad_pdata->irq = irq_of_parse_and_map(np, 0); + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl", + &of_val32) >= 0) + aad_pdata->micbias_pulse_lvl = + da7219_aad_of_micbias_pulse_lvl(codec, of_val32); + else + aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF; + + if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time", + &of_val32) >= 0) + aad_pdata->micbias_pulse_time = of_val32; + + if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0) + aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32); + else + aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS; + + if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0) + aad_pdata->mic_det_thr = + da7219_aad_of_mic_det_thr(codec, of_val32); + else + aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS; + + if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0) + aad_pdata->jack_ins_deb = + da7219_aad_of_jack_ins_deb(codec, of_val32); + else + aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS; + + if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str)) + aad_pdata->jack_det_rate = + da7219_aad_of_jack_det_rate(codec, of_str); + else + aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS; + + if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0) + aad_pdata->jack_rem_deb = + da7219_aad_of_jack_rem_deb(codec, of_val32); + else + aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS; + + if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0) + aad_pdata->a_d_btn_thr = (u8) of_val32; + else + aad_pdata->a_d_btn_thr = 0xA; + + if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0) + aad_pdata->d_b_btn_thr = (u8) of_val32; + else + aad_pdata->d_b_btn_thr = 0x16; + + if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0) + aad_pdata->b_c_btn_thr = (u8) of_val32; + else + aad_pdata->b_c_btn_thr = 0x21; + + if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0) + aad_pdata->c_mic_btn_thr = (u8) of_val32; + else + aad_pdata->c_mic_btn_thr = 0x3E; + + if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0) + aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32); + else + aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2; + + if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0) + aad_pdata->adc_1bit_rpt = + da7219_aad_of_adc_1bit_rpt(codec, of_val32); + else + aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + +out: + of_node_put(aad_np); + + return aad_pdata; +} + +static void da7219_aad_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + struct da7219_pdata *pdata = da7219->pdata; + + if ((pdata) && (pdata->aad_pdata)) { + struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata; + u8 cfg, mask; + + da7219_aad->irq = aad_pdata->irq; + + switch (aad_pdata->micbias_pulse_lvl) { + case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V: + case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V: + da7219_aad->micbias_pulse_lvl = + (aad_pdata->micbias_pulse_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + default: + break; + } + + da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time; + + switch (aad_pdata->btn_cfg) { + case DA7219_AAD_BTN_CFG_2MS: + case DA7219_AAD_BTN_CFG_5MS: + case DA7219_AAD_BTN_CFG_10MS: + case DA7219_AAD_BTN_CFG_50MS: + case DA7219_AAD_BTN_CFG_100MS: + case DA7219_AAD_BTN_CFG_200MS: + case DA7219_AAD_BTN_CFG_500MS: + da7219_aad->btn_cfg = (aad_pdata->btn_cfg << + DA7219_BUTTON_CONFIG_SHIFT); + } + + cfg = 0; + mask = 0; + switch (aad_pdata->mic_det_thr) { + case DA7219_AAD_MIC_DET_THR_200_OHMS: + case DA7219_AAD_MIC_DET_THR_500_OHMS: + case DA7219_AAD_MIC_DET_THR_750_OHMS: + case DA7219_AAD_MIC_DET_THR_1000_OHMS: + cfg |= (aad_pdata->mic_det_thr << + DA7219_MIC_DET_THRESH_SHIFT); + mask |= DA7219_MIC_DET_THRESH_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg); + + cfg = 0; + mask = 0; + switch (aad_pdata->jack_ins_deb) { + case DA7219_AAD_JACK_INS_DEB_5MS: + case DA7219_AAD_JACK_INS_DEB_10MS: + case DA7219_AAD_JACK_INS_DEB_20MS: + case DA7219_AAD_JACK_INS_DEB_50MS: + case DA7219_AAD_JACK_INS_DEB_100MS: + case DA7219_AAD_JACK_INS_DEB_200MS: + case DA7219_AAD_JACK_INS_DEB_500MS: + case DA7219_AAD_JACK_INS_DEB_1S: + cfg |= (aad_pdata->jack_ins_deb << + DA7219_JACKDET_DEBOUNCE_SHIFT); + mask |= DA7219_JACKDET_DEBOUNCE_MASK; + } + switch (aad_pdata->jack_det_rate) { + case DA7219_AAD_JACK_DET_RATE_32_64MS: + case DA7219_AAD_JACK_DET_RATE_64_128MS: + case DA7219_AAD_JACK_DET_RATE_128_256MS: + case DA7219_AAD_JACK_DET_RATE_256_512MS: + cfg |= (aad_pdata->jack_det_rate << + DA7219_JACK_DETECT_RATE_SHIFT); + mask |= DA7219_JACK_DETECT_RATE_MASK; + } + switch (aad_pdata->jack_rem_deb) { + case DA7219_AAD_JACK_REM_DEB_1MS: + case DA7219_AAD_JACK_REM_DEB_5MS: + case DA7219_AAD_JACK_REM_DEB_10MS: + case DA7219_AAD_JACK_REM_DEB_20MS: + cfg |= (aad_pdata->jack_rem_deb << + DA7219_JACKDET_REM_DEB_SHIFT); + mask |= DA7219_JACKDET_REM_DEB_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg); + + snd_soc_write(codec, DA7219_ACCDET_CONFIG_3, + aad_pdata->a_d_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_4, + aad_pdata->d_b_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_5, + aad_pdata->b_c_btn_thr); + snd_soc_write(codec, DA7219_ACCDET_CONFIG_6, + aad_pdata->c_mic_btn_thr); + + cfg = 0; + mask = 0; + switch (aad_pdata->btn_avg) { + case DA7219_AAD_BTN_AVG_1: + case DA7219_AAD_BTN_AVG_2: + case DA7219_AAD_BTN_AVG_4: + case DA7219_AAD_BTN_AVG_8: + cfg |= (aad_pdata->btn_avg << + DA7219_BUTTON_AVERAGE_SHIFT); + mask |= DA7219_BUTTON_AVERAGE_MASK; + } + switch (aad_pdata->adc_1bit_rpt) { + case DA7219_AAD_ADC_1BIT_RPT_1: + case DA7219_AAD_ADC_1BIT_RPT_2: + case DA7219_AAD_ADC_1BIT_RPT_4: + case DA7219_AAD_ADC_1BIT_RPT_8: + cfg |= (aad_pdata->adc_1bit_rpt << + DA7219_ADC_1_BIT_REPEAT_SHIFT); + mask |= DA7219_ADC_1_BIT_REPEAT_MASK; + } + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg); + } +} + + +/* + * Init/Exit + */ + +int da7219_aad_init(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + int ret; + + da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL); + if (!da7219_aad) + return -ENOMEM; + + da7219->aad = da7219_aad; + da7219_aad->codec = codec; + + /* Handle any DT/platform data */ + if ((codec->dev->of_node) && (da7219->pdata)) + da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec); + + da7219_aad_handle_pdata(codec); + + /* Disable button detection */ + snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, + DA7219_BUTTON_CONFIG_MASK, 0); + + INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work); + INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); + + ret = request_threaded_irq(da7219_aad->irq, NULL, + da7219_aad_irq_thread, + IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "da7219-aad", da7219_aad); + if (ret) { + dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; + } + + /* Unmask AAD IRQs */ + memset(mask, 0, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + &mask, DA7219_AAD_IRQ_REG_MAX); + + return 0; +} +EXPORT_SYMBOL_GPL(da7219_aad_init); + +void da7219_aad_exit(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_aad_priv *da7219_aad = da7219->aad; + u8 mask[DA7219_AAD_IRQ_REG_MAX]; + + /* Mask off AAD IRQs */ + memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX); + regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A, + mask, DA7219_AAD_IRQ_REG_MAX); + + free_irq(da7219_aad->irq, da7219_aad); + + cancel_work_sync(&da7219_aad->btn_det_work); + cancel_work_sync(&da7219_aad->hptest_work); +} +EXPORT_SYMBOL_GPL(da7219_aad_exit); + +MODULE_DESCRIPTION("ASoC DA7219 AAD Driver"); +MODULE_AUTHOR("Adam Thomson "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h new file mode 100644 index 000000000000..4fccf677cd06 --- /dev/null +++ b/sound/soc/codecs/da7219-aad.h @@ -0,0 +1,212 @@ +/* + * da7219-aad.h - DA7322 ASoC AAD Driver + * + * Copyright (c) 2015 Dialog Semiconductor Ltd. + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_AAD_H +#define __DA7219_AAD_H + +#include +#include +#include +#include + +/* + * Registers + */ + +#define DA7219_ACCDET_STATUS_A 0xC0 +#define DA7219_ACCDET_STATUS_B 0xC1 +#define DA7219_ACCDET_IRQ_EVENT_A 0xC2 +#define DA7219_ACCDET_IRQ_EVENT_B 0xC3 +#define DA7219_ACCDET_IRQ_MASK_A 0xC4 +#define DA7219_ACCDET_IRQ_MASK_B 0xC5 +#define DA7219_ACCDET_CONFIG_1 0xC6 +#define DA7219_ACCDET_CONFIG_2 0xC7 +#define DA7219_ACCDET_CONFIG_3 0xC8 +#define DA7219_ACCDET_CONFIG_4 0xC9 +#define DA7219_ACCDET_CONFIG_5 0xCA +#define DA7219_ACCDET_CONFIG_6 0xCB +#define DA7219_ACCDET_CONFIG_7 0xCC +#define DA7219_ACCDET_CONFIG_8 0xCD + + +/* + * Bit Fields + */ + +/* DA7219_ACCDET_STATUS_A = 0xC0 */ +#define DA7219_JACK_INSERTION_STS_SHIFT 0 +#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0) +#define DA7219_JACK_TYPE_STS_SHIFT 1 +#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1) +#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2 +#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2) +#define DA7219_MICBIAS_UP_STS_SHIFT 3 +#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3) + +/* DA7219_ACCDET_STATUS_B = 0xC1 */ +#define DA7219_BUTTON_TYPE_STS_SHIFT 0 +#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0) + +/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */ +#define DA7219_E_JACK_INSERTED_SHIFT 0 +#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_E_JACK_REMOVED_SHIFT 1 +#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */ +#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */ +#define DA7219_M_JACK_INSERTED_SHIFT 0 +#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0) +#define DA7219_M_JACK_REMOVED_SHIFT 1 +#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1) +#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2 +#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2) + +/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */ +#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0 +#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0) +#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1 +#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1) +#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2 +#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2) +#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3 +#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3) +#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4 +#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4) +#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5 +#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5) +#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6 +#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6) +#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7 +#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_1 = 0xC6 */ +#define DA7219_ACCDET_EN_SHIFT 0 +#define DA7219_ACCDET_EN_MASK (0x1 << 0) +#define DA7219_BUTTON_CONFIG_SHIFT 1 +#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1) +#define DA7219_MIC_DET_THRESH_SHIFT 4 +#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4) +#define DA7219_JACK_TYPE_DET_EN_SHIFT 6 +#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6) +#define DA7219_PIN_ORDER_DET_EN_SHIFT 7 +#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7) + +/* DA7219_ACCDET_CONFIG_2 = 0xC7 */ +#define DA7219_ACCDET_PAUSE_SHIFT 0 +#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0) +#define DA7219_JACKDET_DEBOUNCE_SHIFT 1 +#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1) +#define DA7219_JACK_DETECT_RATE_SHIFT 4 +#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4) +#define DA7219_JACKDET_REM_DEB_SHIFT 6 +#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6) + +/* DA7219_ACCDET_CONFIG_3 = 0xC8 */ +#define DA7219_A_D_BUTTON_THRESH_SHIFT 0 +#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_4 = 0xC9 */ +#define DA7219_D_B_BUTTON_THRESH_SHIFT 0 +#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_5 = 0xCA */ +#define DA7219_B_C_BUTTON_THRESH_SHIFT 0 +#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_6 = 0xCB */ +#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0 +#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0) + +/* DA7219_ACCDET_CONFIG_7 = 0xCC */ +#define DA7219_BUTTON_AVERAGE_SHIFT 0 +#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0) +#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2 +#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2) +#define DA7219_PIN_ORDER_FORCE_SHIFT 4 +#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4) +#define DA7219_JACK_TYPE_FORCE_SHIFT 5 +#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5) + +/* DA7219_ACCDET_CONFIG_8 = 0xCD */ +#define DA7219_HPTEST_EN_SHIFT 0 +#define DA7219_HPTEST_EN_MASK (0x1 << 0) +#define DA7219_HPTEST_RES_SEL_SHIFT 1 +#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1) +#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1) +#define DA7219_HPTEST_COMP_SHIFT 4 +#define DA7219_HPTEST_COMP_MASK (0x1 << 4) + + +#define DA7219_AAD_MAX_BUTTONS 4 +#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \ + SND_JACK_HEADSET | SND_JACK_LINEOUT | \ + SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +#define DA7219_AAD_MICBIAS_CHK_DELAY 10 +#define DA7219_AAD_MICBIAS_CHK_RETRIES 5 + +#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28 +#define DA7219_AAD_HPTEST_PERIOD 65 + +enum da7219_aad_event_regs { + DA7219_AAD_IRQ_REG_A = 0, + DA7219_AAD_IRQ_REG_B, + DA7219_AAD_IRQ_REG_MAX, +}; + +/* Private data */ +struct da7219_aad_priv { + struct snd_soc_codec *codec; + int irq; + + u8 micbias_pulse_lvl; + u32 micbias_pulse_time; + + u8 btn_cfg; + + struct work_struct btn_det_work; + struct work_struct hptest_work; + + struct snd_soc_jack *jack; + bool jack_inserted; +}; + +/* AAD control */ +void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +/* Init/Exit */ +int da7219_aad_init(struct snd_soc_codec *codec); +void da7219_aad_exit(struct snd_soc_codec *codec); + +#endif /* __DA7219_AAD_H */ diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c new file mode 100644 index 000000000000..76f8fc2c00fe --- /dev/null +++ b/sound/soc/codecs/da7219.c @@ -0,0 +1,1940 @@ +/* + * da7219.c - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson + * + * 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include "da7219.h" +#include "da7219-aad.h" + + +/* + * TLVs and Enums + */ + +/* Input TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0); +static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0); + +/* Output TLVs */ +static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0); + +static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv, + 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -77.25dB to 12dB */ + 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0) +); + +static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0); +static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0); + +/* Input Enums */ +static const char * const da7219_alc_attack_rate_txt[] = { + "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs", + "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs", + "30024/fs" +}; + +static const struct soc_enum da7219_alc_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT, + DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt); + +static const char * const da7219_alc_release_rate_txt[] = { + "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs", + "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs" +}; + +static const struct soc_enum da7219_alc_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT, + DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt); + +static const char * const da7219_alc_hold_time_txt[] = { + "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs", + "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs", + "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" +}; + +static const struct soc_enum da7219_alc_hold_time = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT, + DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt); + +static const char * const da7219_alc_env_rate_txt[] = { + "1/4", "1/16", "1/256", "1/65536" +}; + +static const struct soc_enum da7219_alc_env_attack_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const struct soc_enum da7219_alc_env_release_rate = + SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT, + DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt); + +static const char * const da7219_alc_anticlip_step_txt[] = { + "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs" +}; + +static const struct soc_enum da7219_alc_anticlip_step = + SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTICLIP_STEP_SHIFT, + DA7219_ALC_ANTICLIP_STEP_MAX, + da7219_alc_anticlip_step_txt); + +/* Input/Output Enums */ +static const char * const da7219_gain_ramp_rate_txt[] = { + "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8", + "Nominal Rate / 16" +}; + +static const struct soc_enum da7219_gain_ramp_rate = + SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT, + DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt); + +static const char * const da7219_hpf_mode_txt[] = { + "Disabled", "Audio", "Voice" +}; + +static const unsigned int da7219_hpf_mode_val[] = { + DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN, +}; + +static const struct soc_enum da7219_adc_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const struct soc_enum da7219_dac_hpf_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT, + DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX, + da7219_hpf_mode_txt, da7219_hpf_mode_val); + +static const char * const da7219_audio_hpf_corner_txt[] = { + "2Hz", "4Hz", "8Hz", "16Hz" +}; + +static const struct soc_enum da7219_adc_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const struct soc_enum da7219_dac_audio_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_AUDIO_HPF_CORNER_SHIFT, + DA7219_AUDIO_HPF_CORNER_MAX, + da7219_audio_hpf_corner_txt); + +static const char * const da7219_voice_hpf_corner_txt[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da7219_adc_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1, + DA7219_ADC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const struct soc_enum da7219_dac_voice_hpf_corner = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1, + DA7219_DAC_VOICE_HPF_CORNER_SHIFT, + DA7219_VOICE_HPF_CORNER_MAX, + da7219_voice_hpf_corner_txt); + +static const char * const da7219_tonegen_dtmf_key_txt[] = { + "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D", + "*", "#" +}; + +static const struct soc_enum da7219_tonegen_dtmf_key = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT, + DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt); + +static const char * const da7219_tonegen_swg_sel_txt[] = { + "Sum", "SWG1", "SWG2", "SWG1_1-Cos" +}; + +static const struct soc_enum da7219_tonegen_swg_sel = + SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT, + DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt); + +/* Output Enums */ +static const char * const da7219_dac_softmute_rate_txt[] = { + "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples", + "32 Samples", "64 Samples" +}; + +static const struct soc_enum da7219_dac_softmute_rate = + SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT, + DA7219_DAC_SOFTMUTE_RATE_MAX, + da7219_dac_softmute_rate_txt); + +static const char * const da7219_dac_ng_setup_time_txt[] = { + "256 Samples", "512 Samples", "1024 Samples", "2048 Samples" +}; + +static const struct soc_enum da7219_dac_ng_setup_time = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_SETUP_TIME_SHIFT, + DA7219_DAC_NG_SETUP_TIME_MAX, + da7219_dac_ng_setup_time_txt); + +static const char * const da7219_dac_ng_rampup_txt[] = { + "0.22ms/dB", "0.0138ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampup_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPUP_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampup_txt); + +static const char * const da7219_dac_ng_rampdown_txt[] = { + "0.88ms/dB", "14.08ms/dB" +}; + +static const struct soc_enum da7219_dac_ng_rampdown_rate = + SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME, + DA7219_DAC_NG_RAMPDN_RATE_SHIFT, + DA7219_DAC_NG_RAMP_RATE_MAX, + da7219_dac_ng_rampdown_txt); + + +static const char * const da7219_cp_track_mode_txt[] = { + "Largest Volume", "DAC Volume", "Signal Magnitude" +}; + +static const unsigned int da7219_cp_track_mode_val[] = { + DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL, + DA7219_CP_MCHANGE_SIG_MAG +}; + +static const struct soc_enum da7219_cp_track_mode = + SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT, + DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX, + da7219_cp_track_mode_txt, + da7219_cp_track_mode_val); + + +/* + * Control Functions + */ + +/* Locked Kcontrol calls */ +static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_volsw(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_get_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_lock(&da7219->lock); + ret = snd_soc_put_enum_double(kcontrol, ucontrol); + mutex_unlock(&da7219->lock); + + return ret; +} + +/* ALC */ +static void da7219_alc_calib(struct snd_soc_codec *codec) +{ + u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl; + + /* Save current state of mic control register */ + mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL); + + /* Save current state of input mixer control register */ + mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL); + + /* Save current state of input ADC control register */ + adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL); + + /* Enable then Mute MIC PGAs */ + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK, + DA7219_MIC_1_AMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_MASK, + DA7219_MIC_1_AMP_MUTE_EN_MASK); + + /* Enable input mixers unmuted */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_MASK | + DA7219_MIXIN_L_AMP_MUTE_EN_MASK, + DA7219_MIXIN_L_AMP_EN_MASK); + + /* Enable input filters unmuted */ + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK, + DA7219_ADC_L_EN_MASK); + + /* Perform auto calibration */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_AUTO_CALIB_EN_MASK, + DA7219_ALC_AUTO_CALIB_EN_MASK); + do { + calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1); + } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK); + + /* If auto calibration fails, disable DC offset, hybrid ALC */ + if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) { + dev_warn(codec->dev, + "ALC auto calibration failed with overflow\n"); + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, 0); + } else { + /* Enable DC offset cancellation, hybrid mode */ + snd_soc_update_bits(codec, DA7219_ALC_CTRL1, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK, + DA7219_ALC_OFFSET_EN_MASK | + DA7219_ALC_SYNC_MODE_MASK); + } + + /* Restore input filter control register to original state */ + snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl); + + /* Restore input mixer control registers to original state */ + snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl); + + /* Restore MIC control registers to original states */ + snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl); +} + +static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + + /* + * If ALC in operation and value of control has been updated, + * make sure calibrated offsets are updated. + */ + if ((ret == 1) && (da7219->alc_en)) + da7219_alc_calib(codec); + + return ret; +} + +static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + + /* Force ALC offset calibration if enabling ALC */ + if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) { + da7219_alc_calib(codec); + da7219->alc_en = false; + } else { + da7219->alc_en = false; + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + +/* ToneGen */ +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + mutex_lock(&da7219->lock); + ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + if (ret) + return ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to host endianness here. + */ + ucontrol->value.integer.value[0] = le16_to_cpu(val); + + return 0; +} + +static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct soc_mixer_control *mixer_ctrl = + (struct soc_mixer_control *) kcontrol->private_value; + unsigned int reg = mixer_ctrl->reg; + u16 val; + int ret; + + /* + * Frequency value spans two 8-bit registers, lower then upper byte. + * Therefore we need to convert to little endian here to align with + * HW registers. + */ + val = cpu_to_le16(ucontrol->value.integer.value[0]); + + mutex_lock(&da7219->lock); + ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + mutex_unlock(&da7219->lock); + + return ret; +} + + +/* + * KControls + */ + +static const struct snd_kcontrol_new da7219_snd_controls[] = { + /* Mics */ + SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN, + DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX, + DA7219_NO_INVERT, da7219_mic_gain_tlv), + SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Mixer Input */ + SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN, + DA7219_MIXIN_L_AMP_GAIN_SHIFT, + DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_mixin_gain_put, + da7219_mixin_gain_tlv), + SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ADC */ + SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN, + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_adc_dig_gain_tlv), + SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL, + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + + /* ALC */ + SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate), + SOC_ENUM("ALC Release Rate", da7219_alc_release_rate), + SOC_ENUM("ALC Hold Time", da7219_alc_hold_time), + SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate), + SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate), + SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE, + DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN, + DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX, + DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX, + DA7219_INVERT, da7219_alc_threshold_tlv), + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS, + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MIN_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS, + DA7219_ALC_ANA_GAIN_MAX_SHIFT, + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), + SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step), + SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL, + DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT), + SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT, + snd_soc_get_volsw, da7219_alc_sw_put), + + /* Input High-Pass Filters */ + SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode), + SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner), + SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner), + + /* Sidetone Filter */ + SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN, + DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX, + DA7219_NO_INVERT, da7219_sidetone_gain_tlv), + SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT), + + /* Tone Generator */ + SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2, + DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put, da7219_tonegen_gain_tlv), + SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1, + DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L, + DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L, + DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT, + da7219_tonegen_freq_get, da7219_tonegen_freq_put), + SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER, + DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER, + DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX, + DA7219_NO_INVERT), + + /* Gain ramping */ + SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate), + + /* DAC High-Pass Filter */ + SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode, + da7219_enum_locked_get, da7219_enum_locked_put), + SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner), + SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner), + + /* DAC 5-Band Equaliser */ + SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2, + DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3, + DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX, + DA7219_NO_INVERT, da7219_dac_eq_band_tlv), + SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4, + DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Softmute */ + SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate), + SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5, + DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_NO_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + + /* DAC Noise Gate */ + SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time), + SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate), + SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate), + SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH, + DA7219_DAC_NG_OFF_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH, + DA7219_DAC_NG_ON_THRESHOLD_SHIFT, + DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT, + da7219_dac_ng_threshold_tlv), + SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* DACs */ + SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN, + DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT, + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_dac_dig_gain_tlv), + SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put), + SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL, + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + + /* CP */ + SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode), + SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1, + DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX, + DA7219_NO_INVERT), + + /* Headphones */ + SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN, + DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT, + DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT, + da7219_volsw_locked_get, da7219_volsw_locked_put, + da7219_hp_gain_tlv), + SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL, + DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, + DA7219_INVERT, da7219_volsw_locked_get, + da7219_volsw_locked_put), + SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), + SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL, + DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + + +/* + * DAPM Mux Controls + */ + +static const char * const da7219_out_sel_txt[] = { + "ADC", "Tone Generator", "DAIL", "DAIR" +}; + +static const struct soc_enum da7219_out_dail_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dail_sel_mux = + SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel); + +static const struct soc_enum da7219_out_dair_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI, + DA7219_DAI_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dair_sel_mux = + SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel); + +static const struct soc_enum da7219_out_dacl_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_L_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacl_sel_mux = + SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel); + +static const struct soc_enum da7219_out_dacr_sel = + SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC, + DA7219_DAC_R_SRC_SHIFT, + DA7219_OUT_SRC_MAX, + da7219_out_sel_txt); + +static const struct snd_kcontrol_new da7219_out_dacr_sel_mux = + SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel); + + +/* + * DAPM Mixer Controls + */ + +static const struct snd_kcontrol_new da7219_mixin_controls[] = { + SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT, + DA7219_MIXIN_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_l_controls[] = { + SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT, + DA7219_MIXOUT_L_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +static const struct snd_kcontrol_new da7219_mixout_r_controls[] = { + SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT, + DA7219_MIXOUT_R_MIX_SELECT_SHIFT, + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), +}; + +#define DA7219_DMIX_ST_CTRLS(reg) \ + SOC_DAPM_SINGLE("Out FilterL Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Out FilterR Switch", reg, \ + DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \ + SOC_DAPM_SINGLE("Sidetone Switch", reg, \ + DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \ + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \ + +static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L), +}; + +static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = { + DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R), +}; + + +/* + * DAPM Events + */ + +static int da7219_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 da7219_priv *da7219 = 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: + if (da7219->master) + /* Enable DAI clks for master mode */ + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_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, DA7219_PC_COUNT, + DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7219->master) + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* + * DAPM Widgets + */ + +static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = { + /* Input Supplies */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL, + DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC"), + + /* Input PGAs */ + SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Filters */ + SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, + DA7219_NO_INVERT), + + /* Tone Generator */ + SND_SOC_DAPM_SIGGEN("TONE"), + SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1, + DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Sidetone Input */ + SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL, + DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT), + + /* Input Mixer Supply */ + SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + + /* Input Mixer */ + SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0, + da7219_mixin_controls, + ARRAY_SIZE(da7219_mixin_controls)), + + /* Input Muxes */ + SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dail_sel_mux), + SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dair_sel_mux), + + /* DAI Supply */ + SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT, + DA7219_NO_INVERT, da7219_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DAI */ + SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Muxes */ + SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacl_sel_mux), + SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0, + &da7219_out_dacr_sel_mux), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_mixout_l_controls, + ARRAY_SIZE(da7219_mixout_l_controls)), + SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0, + da7219_mixout_r_controls, + ARRAY_SIZE(da7219_mixout_r_controls)), + + /* Sidetone Mixers */ + SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0, + da7219_st_out_filtl_mix_controls, + ARRAY_SIZE(da7219_st_out_filtl_mix_controls)), + SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0, + 0, da7219_st_out_filtr_mix_controls, + ARRAY_SIZE(da7219_st_out_filtr_mix_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT, + DA7219_NO_INVERT), + SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT, + DA7219_NO_INVERT), + + /* Output PGAs */ + SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL, + DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL, + DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0), + + /* Output Supplies */ + SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT, + DA7219_NO_INVERT, NULL, 0), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), +}; + + +/* + * DAPM Mux Routes + */ + +#define DA7219_OUT_DAI_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIOUT"}, \ + {name, "DAIR", "DAIOUT"} + +#define DA7219_OUT_DAC_MUX_ROUTES(name) \ + {name, "ADC", "Mixer In"}, \ + {name, "Tone Generator", "Tone Generator"}, \ + {name, "DAIL", "DAIIN"}, \ + {name, "DAIR", "DAIIN"} + +/* + * DAPM Mixer Routes + */ + +#define DA7219_DMIX_ST_ROUTES(name) \ + {name, "Out FilterL Switch", "Mixer Out FilterL"}, \ + {name, "Out FilterR Switch", "Mixer Out FilterR"}, \ + {name, "Sidetone Switch", "Sidetone Filter"} + + +/* + * DAPM audio route definition + */ + +static const struct snd_soc_dapm_route da7219_audio_map[] = { + /* Input paths */ + {"MIC", NULL, "Mic Bias"}, + {"Mic PGA", NULL, "MIC"}, + {"Mixin PGA", NULL, "Mic PGA"}, + {"ADC", NULL, "Mixin PGA"}, + + {"Sidetone Filter", NULL, "ADC"}, + {"Mixer In", NULL, "Mixer In Supply"}, + {"Mixer In", "Mic Switch", "ADC"}, + + {"Tone Generator", NULL, "TONE"}, + + DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"), + DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"), + + {"DAIOUT", NULL, "Out DAIL Mux"}, + {"DAIOUT", NULL, "Out DAIR Mux"}, + {"DAIOUT", NULL, "DAI"}, + + /* Output paths */ + {"DAIIN", NULL, "DAI"}, + + DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"), + DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"), + + {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"}, + {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"}, + + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"), + DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"), + + {"DACL", NULL, "ST Mixer Out FilterL"}, + {"DACR", NULL, "ST Mixer Out FilterR"}, + + {"Mixout Left PGA", NULL, "DACL"}, + {"Mixout Right PGA", NULL, "DACR"}, + + {"Headphone Left PGA", NULL, "Mixout Left PGA"}, + {"Headphone Right PGA", NULL, "Mixout Right PGA"}, + + {"HPL", NULL, "Headphone Left PGA"}, + {"HPR", NULL, "Headphone Right PGA"}, + + {"HPL", NULL, "Charge Pump"}, + {"HPR", NULL, "Charge Pump"}, +}; + + +/* + * DAI operations + */ + +static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) + return 0; + + if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + + switch (clk_id) { + case DA7219_CLKSRC_MCLK_SQR: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, + DA7219_PLL_MCLK_SQR_EN_MASK); + break; + case DA7219_CLKSRC_MCLK: + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_MCLK_SQR_EN_MASK, 0); + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } + + da7219->clk_src = clk_id; + + if (da7219->mclk) { + freq = clk_round_rate(da7219->mclk, freq); + ret = clk_set_rate(da7219->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7219->mclk_rate = freq; + + return 0; +} + +static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + u8 pll_ctrl, indiv_bits, indiv; + u8 pll_frac_top, pll_frac_bot, pll_integer; + u32 freq_ref; + u64 frac_div; + + /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ + if (da7219->mclk_rate == 32768) { + indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; + indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; + } else if (da7219->mclk_rate < 2000000) { + 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 <= 54000000) { + indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; + indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + } else { + dev_err(codec->dev, "PLL input clock %d above valid range\n", + da7219->mclk_rate); + return -EINVAL; + } + freq_ref = (da7219->mclk_rate / indiv); + pll_ctrl = indiv_bits; + + /* Configure PLL */ + switch (source) { + case DA7219_SYSCLK_MCLK: + pll_ctrl |= DA7219_PLL_MODE_BYPASS; + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | + DA7219_PLL_MODE_MASK, pll_ctrl); + return 0; + case DA7219_SYSCLK_PLL: + pll_ctrl |= DA7219_PLL_MODE_NORMAL; + break; + case DA7219_SYSCLK_PLL_SRM: + pll_ctrl |= DA7219_PLL_MODE_SRM; + break; + case DA7219_SYSCLK_PLL_32KHZ: + pll_ctrl |= DA7219_PLL_MODE_32KHZ; + break; + default: + dev_err(codec->dev, "Invalid PLL config\n"); + return -EINVAL; + } + + /* Calculate dividers for PLL */ + pll_integer = fout / freq_ref; + frac_div = (u64)(fout % freq_ref) * 8192ULL; + do_div(frac_div, freq_ref); + pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK; + pll_frac_bot = (frac_div) & DA7219_BYTE_MASK; + + /* Write PLL config & dividers */ + snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top); + snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot); + snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer); + snd_soc_update_bits(codec, DA7219_PLL_CTRL, + DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK, + pll_ctrl); + + return 0; +} + +static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_clk_mode = 0, dai_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da7219->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + da7219->master = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + dai_clk_mode |= DA7219_DAI_CLK_POL_INV; + break; + case SND_SOC_DAIFMT_IB_IF: + dai_clk_mode |= DA7219_DAI_WCLK_POL_INV | + DA7219_DAI_CLK_POL_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + dai_ctrl |= DA7219_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + dai_ctrl |= DA7219_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + /* By default 64 BCLKs per WCLK is supported */ + dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64; + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK | + DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK, + dai_clk_mode); + snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK, + dai_ctrl); + + return 0; +} + +static int da7219_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; + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + u8 dai_bclks_per_wclk; + u16 offset; + u32 frame_size; + + /* No channels enabled so disable TDM, revert to 64-bit frames */ + if (!tx_mask) { + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, 0); + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + DA7219_DAI_BCLKS_PER_WCLK_64); + return 0; + } + + /* Check we have valid slots */ + if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) { + dev_err(codec->dev, "Invalid number of slots, max = %d\n", + DA7219_DAI_TDM_MAX_SLOTS); + return -EINVAL; + } + + /* Check we have a valid offset given */ + if (rx_mask > DA7219_DAI_OFFSET_MAX) { + dev_err(codec->dev, "Invalid slot offset, max = %d\n", + DA7219_DAI_OFFSET_MAX); + return -EINVAL; + } + + /* Calculate & validate frame size based on slot info provided. */ + frame_size = slots * slot_width; + switch (frame_size) { + case 32: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32; + break; + case 64: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64; + break; + case 128: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128; + break; + case 256: + dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256; + break; + default: + dev_err(codec->dev, "Invalid frame size %d\n", frame_size); + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE, + DA7219_DAI_BCLKS_PER_WCLK_MASK, + dai_bclks_per_wclk); + + offset = cpu_to_le16(rx_mask); + regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER, + &offset, sizeof(offset)); + + snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL, + DA7219_DAI_TDM_CH_EN_MASK | + DA7219_DAI_TDM_MODE_EN_MASK, + (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) | + DA7219_DAI_TDM_MODE_EN_MASK); + + return 0; +} + +static int da7219_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; + u8 dai_ctrl = 0, fs; + unsigned int channels; + + switch (params_width(params)) { + case 16: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE; + break; + case 20: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE; + break; + case 24: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE; + break; + case 32: + dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE; + break; + default: + return -EINVAL; + } + + channels = params_channels(params); + if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) { + dev_err(codec->dev, + "Invalid number of channels, only 1 to %d supported\n", + DA7219_DAI_CH_NUM_MAX); + return -EINVAL; + } + dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT; + + switch (params_rate(params)) { + case 8000: + fs = DA7219_SR_8000; + break; + case 11025: + fs = DA7219_SR_11025; + break; + case 12000: + fs = DA7219_SR_12000; + break; + case 16000: + fs = DA7219_SR_16000; + break; + case 22050: + fs = DA7219_SR_22050; + break; + case 24000: + fs = DA7219_SR_24000; + break; + case 32000: + fs = DA7219_SR_32000; + break; + case 44100: + fs = DA7219_SR_44100; + break; + case 48000: + fs = DA7219_SR_48000; + break; + case 88200: + fs = DA7219_SR_88200; + break; + case 96000: + fs = DA7219_SR_96000; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, DA7219_DAI_CTRL, + DA7219_DAI_WORD_LENGTH_MASK | + DA7219_DAI_CH_NUM_MASK, + dai_ctrl); + snd_soc_write(codec, DA7219_SR, fs); + + return 0; +} + +static const struct snd_soc_dai_ops da7219_dai_ops = { + .hw_params = da7219_hw_params, + .set_sysclk = da7219_set_dai_sysclk, + .set_pll = da7219_set_dai_pll, + .set_fmt = da7219_set_dai_fmt, + .set_tdm_slot = da7219_set_dai_tdm_slot, +}; + +#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver da7219_dai = { + .name = "da7219-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = DA7219_DAI_CH_NUM_MAX, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = DA7219_FORMATS, + }, + .ops = &da7219_dai_ops, + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, +}; + + +/* + * DT + */ + +static const struct of_device_id da7219_of_match[] = { + { .compatible = "dlg,da7219", }, + { } +}; +MODULE_DEVICE_TABLE(of, da7219_of_match); + +static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec, + u32 val) +{ + switch (val) { + case 1050: + return DA7219_LDO_LVL_SEL_1_05V; + case 1100: + return DA7219_LDO_LVL_SEL_1_10V; + case 1200: + return DA7219_LDO_LVL_SEL_1_20V; + case 1400: + return DA7219_LDO_LVL_SEL_1_40V; + default: + dev_warn(codec->dev, "Invalid LDO level"); + return DA7219_LDO_LVL_SEL_1_05V; + } +} + +static enum da7219_micbias_voltage + da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) +{ + switch (val) { + case 1800: + return DA7219_MICBIAS_1_8V; + case 2000: + return DA7219_MICBIAS_2_0V; + case 2200: + return DA7219_MICBIAS_2_2V; + case 2400: + return DA7219_MICBIAS_2_4V; + case 2600: + return DA7219_MICBIAS_2_6V; + default: + dev_warn(codec->dev, "Invalid micbias level"); + return DA7219_MICBIAS_2_2V; + } +} + +static enum da7219_mic_amp_in_sel + da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str) +{ + if (!strcmp(str, "diff")) { + return DA7219_MIC_AMP_IN_SEL_DIFF; + } else if (!strcmp(str, "se_p")) { + return DA7219_MIC_AMP_IN_SEL_SE_P; + } else if (!strcmp(str, "se_n")) { + return DA7219_MIC_AMP_IN_SEL_SE_N; + } else { + dev_warn(codec->dev, "Invalid mic input type selection"); + return DA7219_MIC_AMP_IN_SEL_DIFF; + } +} + +static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec) +{ + struct device_node *np = codec->dev->of_node; + struct da7219_pdata *pdata; + const char *of_str; + u32 of_val32; + + pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return NULL; + + if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0) + pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32); + + if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0) + pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32); + else + pdata->micbias_lvl = DA7219_MICBIAS_2_2V; + + if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str)) + pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str); + else + pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + + return pdata; +} + + +/* + * Codec driver functions + */ + +static int da7219_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + clk_prepare_enable(da7219->mclk); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, + DA7219_BIAS_EN_MASK); + + /* Enable Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, + DA7219_LDO_EN_MASK); + } + break; + case SND_SOC_BIAS_OFF: + /* Only disable if jack detection not active */ + if (!da7219->aad->jack) { + /* Bypass Internal Digital LDO */ + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_EN_MASK, 0); + + /* Master bias */ + snd_soc_update_bits(codec, DA7219_REFERENCES, + DA7219_BIAS_EN_MASK, 0); + } + + /* MCLK */ + clk_disable_unprepare(da7219->mclk); + break; + } + + return 0; +} + +static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = { + [DA7219_SUPPLY_VDD] = "VDD", + [DA7219_SUPPLY_VDDMIC] = "VDDMIC", + [DA7219_SUPPLY_VDDIO] = "VDDIO", +}; + +static int da7219_handle_supplies(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct regulator *vddio; + u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V; + int i, ret; + + /* Get required supplies */ + for (i = 0; i < DA7219_NUM_SUPPLIES; ++i) + da7219->supplies[i].supply = da7219_supply_names[i]; + + ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES, + da7219->supplies); + if (ret) + return ret; + + /* Determine VDDIO voltage provided */ + vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; + ret = regulator_get_voltage(vddio); + if (ret < 1200000) + dev_warn(codec->dev, "Invalid VDDIO voltage\n"); + else if (ret < 2800000) + io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; + + /* Enable main supplies */ + ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies); + + /* Ensure device in active mode */ + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK); + + /* Update IO voltage level range */ + snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl); + + return ret; +} + +static void da7219_handle_pdata(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + struct da7219_pdata *pdata = da7219->pdata; + + if (pdata) { + u8 micbias_lvl = 0; + + /* Internal LDO */ + switch (pdata->ldo_lvl_sel) { + case DA7219_LDO_LVL_SEL_1_05V: + case DA7219_LDO_LVL_SEL_1_10V: + case DA7219_LDO_LVL_SEL_1_20V: + case DA7219_LDO_LVL_SEL_1_40V: + snd_soc_update_bits(codec, DA7219_LDO_CTRL, + DA7219_LDO_LEVEL_SELECT_MASK, + (pdata->ldo_lvl_sel << + DA7219_LDO_LEVEL_SELECT_SHIFT)); + break; + } + + /* Mic Bias voltages */ + switch (pdata->micbias_lvl) { + case DA7219_MICBIAS_1_8V: + case DA7219_MICBIAS_2_0V: + case DA7219_MICBIAS_2_2V: + case DA7219_MICBIAS_2_4V: + case DA7219_MICBIAS_2_6V: + micbias_lvl |= (pdata->micbias_lvl << + DA7219_MICBIAS1_LEVEL_SHIFT); + break; + } + + snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl); + + /* Mic */ + switch (pdata->mic_amp_in_sel) { + case DA7219_MIC_AMP_IN_SEL_DIFF: + case DA7219_MIC_AMP_IN_SEL_SE_P: + case DA7219_MIC_AMP_IN_SEL_SE_N: + snd_soc_write(codec, DA7219_MIC_1_SELECT, + pdata->mic_amp_in_sel); + break; + } + } +} + +static int da7219_probe(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + int ret; + + mutex_init(&da7219->lock); + + /* Regulator configuration */ + ret = da7219_handle_supplies(codec); + if (ret) + return ret; + + /* Handle DT/Platform data */ + if (codec->dev->of_node) + da7219->pdata = da7219_of_to_pdata(codec); + else + da7219->pdata = dev_get_platdata(codec->dev); + + da7219_handle_pdata(codec); + + /* Check if MCLK provided */ + da7219->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7219->mclk)) { + if (PTR_ERR(da7219->mclk) != -ENOENT) + return PTR_ERR(da7219->mclk); + else + da7219->mclk = NULL; + } + + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, + DA7219_PC_FREERUN_MASK); + + /* Default gain ramping */ + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK, + DA7219_MIXIN_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK, + DA7219_ADC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK, + DA7219_DAC_L_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK, + DA7219_DAC_R_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, + DA7219_HP_L_AMP_RAMP_EN_MASK, + DA7219_HP_L_AMP_RAMP_EN_MASK); + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, + DA7219_HP_R_AMP_RAMP_EN_MASK, + DA7219_HP_R_AMP_RAMP_EN_MASK); + + /* Default infinite tone gen, start/stop by Kcontrol */ + snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK); + + /* Initialise AAD block */ + return da7219_aad_init(codec); +} + +static int da7219_remove(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + da7219_aad_exit(codec); + + /* Supplies */ + return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); +} + +#ifdef CONFIG_PM +static int da7219_suspend(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + + /* Put device into standby mode if jack detection disabled */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0); + + return 0; +} + +static int da7219_resume(struct snd_soc_codec *codec) +{ + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + + /* Put device into active mode if previously pushed to standby */ + if (!da7219->aad->jack) + snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, + DA7219_SYSTEM_ACTIVE_MASK); + + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define da7219_suspend NULL +#define da7219_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_da7219 = { + .probe = da7219_probe, + .remove = da7219_remove, + .suspend = da7219_suspend, + .resume = da7219_resume, + .set_bias_level = da7219_set_bias_level, + + .controls = da7219_snd_controls, + .num_controls = ARRAY_SIZE(da7219_snd_controls), + + .dapm_widgets = da7219_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets), + .dapm_routes = da7219_audio_map, + .num_dapm_routes = ARRAY_SIZE(da7219_audio_map), +}; + + +/* + * Regmap configs + */ + +static struct reg_default da7219_reg_defaults[] = { + { DA7219_MIC_1_SELECT, 0x00 }, + { DA7219_CIF_TIMEOUT_CTRL, 0x01 }, + { DA7219_SR_24_48, 0x00 }, + { DA7219_SR, 0x0A }, + { DA7219_CIF_I2C_ADDR_CFG, 0x02 }, + { DA7219_PLL_CTRL, 0x10 }, + { DA7219_PLL_FRAC_TOP, 0x00 }, + { DA7219_PLL_FRAC_BOT, 0x00 }, + { DA7219_PLL_INTEGER, 0x20 }, + { DA7219_DIG_ROUTING_DAI, 0x10 }, + { DA7219_DAI_CLK_MODE, 0x01 }, + { DA7219_DAI_CTRL, 0x28 }, + { DA7219_DAI_TDM_CTRL, 0x40 }, + { DA7219_DIG_ROUTING_DAC, 0x32 }, + { DA7219_DAI_OFFSET_LOWER, 0x00 }, + { DA7219_DAI_OFFSET_UPPER, 0x00 }, + { DA7219_REFERENCES, 0x00 }, + { DA7219_MIXIN_L_SELECT, 0x00 }, + { DA7219_MIXIN_L_GAIN, 0x03 }, + { DA7219_ADC_L_GAIN, 0x6F }, + { DA7219_ADC_FILTERS1, 0x80 }, + { DA7219_MIC_1_GAIN, 0x01 }, + { DA7219_SIDETONE_CTRL, 0x40 }, + { DA7219_SIDETONE_GAIN, 0x0E }, + { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 }, + { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 }, + { DA7219_DAC_FILTERS5, 0x00 }, + { DA7219_DAC_FILTERS2, 0x88 }, + { DA7219_DAC_FILTERS3, 0x88 }, + { DA7219_DAC_FILTERS4, 0x08 }, + { DA7219_DAC_FILTERS1, 0x80 }, + { DA7219_DAC_L_GAIN, 0x6F }, + { DA7219_DAC_R_GAIN, 0x6F }, + { DA7219_CP_CTRL, 0x20 }, + { DA7219_HP_L_GAIN, 0x39 }, + { DA7219_HP_R_GAIN, 0x39 }, + { DA7219_MIXOUT_L_SELECT, 0x00 }, + { DA7219_MIXOUT_R_SELECT, 0x00 }, + { DA7219_MICBIAS_CTRL, 0x03 }, + { DA7219_MIC_1_CTRL, 0x40 }, + { DA7219_MIXIN_L_CTRL, 0x40 }, + { DA7219_ADC_L_CTRL, 0x40 }, + { DA7219_DAC_L_CTRL, 0x40 }, + { DA7219_DAC_R_CTRL, 0x40 }, + { DA7219_HP_L_CTRL, 0x40 }, + { DA7219_HP_R_CTRL, 0x40 }, + { DA7219_MIXOUT_L_CTRL, 0x10 }, + { DA7219_MIXOUT_R_CTRL, 0x10 }, + { DA7219_CHIP_ID1, 0x23 }, + { DA7219_CHIP_ID2, 0x93 }, + { DA7219_CHIP_REVISION, 0x00 }, + { DA7219_LDO_CTRL, 0x00 }, + { DA7219_IO_CTRL, 0x00 }, + { DA7219_GAIN_RAMP_CTRL, 0x00 }, + { DA7219_PC_COUNT, 0x02 }, + { DA7219_CP_VOL_THRESHOLD1, 0x0E }, + { DA7219_DIG_CTRL, 0x00 }, + { DA7219_ALC_CTRL2, 0x00 }, + { DA7219_ALC_CTRL3, 0x00 }, + { DA7219_ALC_NOISE, 0x3F }, + { DA7219_ALC_TARGET_MIN, 0x3F }, + { DA7219_ALC_TARGET_MAX, 0x00 }, + { DA7219_ALC_GAIN_LIMITS, 0xFF }, + { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 }, + { DA7219_ALC_ANTICLIP_CTRL, 0x00 }, + { DA7219_ALC_ANTICLIP_LEVEL, 0x00 }, + { DA7219_DAC_NG_SETUP_TIME, 0x00 }, + { DA7219_DAC_NG_OFF_THRESH, 0x00 }, + { DA7219_DAC_NG_ON_THRESH, 0x00 }, + { DA7219_DAC_NG_CTRL, 0x00 }, + { DA7219_TONE_GEN_CFG1, 0x00 }, + { DA7219_TONE_GEN_CFG2, 0x00 }, + { DA7219_TONE_GEN_CYCLES, 0x00 }, + { DA7219_TONE_GEN_FREQ1_L, 0x55 }, + { DA7219_TONE_GEN_FREQ1_U, 0x15 }, + { DA7219_TONE_GEN_FREQ2_L, 0x00 }, + { DA7219_TONE_GEN_FREQ2_U, 0x40 }, + { DA7219_TONE_GEN_ON_PER, 0x02 }, + { DA7219_TONE_GEN_OFF_PER, 0x01 }, + { DA7219_ACCDET_IRQ_MASK_A, 0x00 }, + { DA7219_ACCDET_IRQ_MASK_B, 0x00 }, + { DA7219_ACCDET_CONFIG_1, 0xD6 }, + { DA7219_ACCDET_CONFIG_2, 0x34 }, + { DA7219_ACCDET_CONFIG_3, 0x0A }, + { DA7219_ACCDET_CONFIG_4, 0x16 }, + { DA7219_ACCDET_CONFIG_5, 0x21 }, + { DA7219_ACCDET_CONFIG_6, 0x3E }, + { DA7219_ACCDET_CONFIG_7, 0x01 }, + { DA7219_SYSTEM_ACTIVE, 0x00 }, +}; + +static bool da7219_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7219_MIC_1_GAIN_STATUS: + case DA7219_MIXIN_L_GAIN_STATUS: + case DA7219_ADC_L_GAIN_STATUS: + case DA7219_DAC_L_GAIN_STATUS: + case DA7219_DAC_R_GAIN_STATUS: + case DA7219_HP_L_GAIN_STATUS: + case DA7219_HP_R_GAIN_STATUS: + case DA7219_CIF_CTRL: + case DA7219_PLL_SRM_STS: + case DA7219_ALC_CTRL1: + case DA7219_SYSTEM_MODES_INPUT: + case DA7219_SYSTEM_MODES_OUTPUT: + case DA7219_ALC_OFFSET_AUTO_M_L: + case DA7219_ALC_OFFSET_AUTO_U_L: + case DA7219_TONE_GEN_CFG1: + case DA7219_ACCDET_STATUS_A: + case DA7219_ACCDET_STATUS_B: + case DA7219_ACCDET_IRQ_EVENT_A: + case DA7219_ACCDET_IRQ_EVENT_B: + case DA7219_ACCDET_CONFIG_8: + case DA7219_SYSTEM_STATUS: + return 1; + default: + return 0; + } +} + +static const struct regmap_config da7219_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA7219_SYSTEM_ACTIVE, + .reg_defaults = da7219_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults), + .volatile_reg = da7219_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + + +/* + * I2C layer + */ + +static int da7219_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da7219_priv *da7219; + int ret; + + da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv), + GFP_KERNEL); + if (!da7219) + return -ENOMEM; + + i2c_set_clientdata(i2c, da7219); + + da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config); + if (IS_ERR(da7219->regmap)) { + ret = PTR_ERR(da7219->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219, + &da7219_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n", + ret); + } + return ret; +} + +static int da7219_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id da7219_i2c_id[] = { + { "da7219", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, da7219_i2c_id); + +static struct i2c_driver da7219_i2c_driver = { + .driver = { + .name = "da7219", + .of_match_table = da7219_of_match, + }, + .probe = da7219_i2c_probe, + .remove = da7219_i2c_remove, + .id_table = da7219_i2c_id, +}; + +module_i2c_driver(da7219_i2c_driver); + +MODULE_DESCRIPTION("ASoC DA7219 Codec Driver"); +MODULE_AUTHOR("Adam Thomson "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h new file mode 100644 index 000000000000..b514268c6c56 --- /dev/null +++ b/sound/soc/codecs/da7219.h @@ -0,0 +1,820 @@ +/* + * da7219.h - DA7219 ALSA SoC Codec Driver + * + * Copyright (c) 2015 Dialog Semiconductor + * + * Author: Adam Thomson + * + * 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. + */ + +#ifndef __DA7219_H +#define __DA7219_H + +#include +#include +#include + +/* + * Registers + */ + +#define DA7219_MIC_1_GAIN_STATUS 0x6 +#define DA7219_MIXIN_L_GAIN_STATUS 0x8 +#define DA7219_ADC_L_GAIN_STATUS 0xA +#define DA7219_DAC_L_GAIN_STATUS 0xC +#define DA7219_DAC_R_GAIN_STATUS 0xD +#define DA7219_HP_L_GAIN_STATUS 0xE +#define DA7219_HP_R_GAIN_STATUS 0xF +#define DA7219_MIC_1_SELECT 0x10 +#define DA7219_CIF_TIMEOUT_CTRL 0x12 +#define DA7219_CIF_CTRL 0x13 +#define DA7219_SR_24_48 0x16 +#define DA7219_SR 0x17 +#define DA7219_CIF_I2C_ADDR_CFG 0x1B +#define DA7219_PLL_CTRL 0x20 +#define DA7219_PLL_FRAC_TOP 0x22 +#define DA7219_PLL_FRAC_BOT 0x23 +#define DA7219_PLL_INTEGER 0x24 +#define DA7219_PLL_SRM_STS 0x25 +#define DA7219_DIG_ROUTING_DAI 0x2A +#define DA7219_DAI_CLK_MODE 0x2B +#define DA7219_DAI_CTRL 0x2C +#define DA7219_DAI_TDM_CTRL 0x2D +#define DA7219_DIG_ROUTING_DAC 0x2E +#define DA7219_ALC_CTRL1 0x2F +#define DA7219_DAI_OFFSET_LOWER 0x30 +#define DA7219_DAI_OFFSET_UPPER 0x31 +#define DA7219_REFERENCES 0x32 +#define DA7219_MIXIN_L_SELECT 0x33 +#define DA7219_MIXIN_L_GAIN 0x34 +#define DA7219_ADC_L_GAIN 0x36 +#define DA7219_ADC_FILTERS1 0x38 +#define DA7219_MIC_1_GAIN 0x39 +#define DA7219_SIDETONE_CTRL 0x3A +#define DA7219_SIDETONE_GAIN 0x3B +#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C +#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D +#define DA7219_DAC_FILTERS5 0x40 +#define DA7219_DAC_FILTERS2 0x41 +#define DA7219_DAC_FILTERS3 0x42 +#define DA7219_DAC_FILTERS4 0x43 +#define DA7219_DAC_FILTERS1 0x44 +#define DA7219_DAC_L_GAIN 0x45 +#define DA7219_DAC_R_GAIN 0x46 +#define DA7219_CP_CTRL 0x47 +#define DA7219_HP_L_GAIN 0x48 +#define DA7219_HP_R_GAIN 0x49 +#define DA7219_MIXOUT_L_SELECT 0x4B +#define DA7219_MIXOUT_R_SELECT 0x4C +#define DA7219_SYSTEM_MODES_INPUT 0x50 +#define DA7219_SYSTEM_MODES_OUTPUT 0x51 +#define DA7219_MICBIAS_CTRL 0x62 +#define DA7219_MIC_1_CTRL 0x63 +#define DA7219_MIXIN_L_CTRL 0x65 +#define DA7219_ADC_L_CTRL 0x67 +#define DA7219_DAC_L_CTRL 0x69 +#define DA7219_DAC_R_CTRL 0x6A +#define DA7219_HP_L_CTRL 0x6B +#define DA7219_HP_R_CTRL 0x6C +#define DA7219_MIXOUT_L_CTRL 0x6E +#define DA7219_MIXOUT_R_CTRL 0x6F +#define DA7219_CHIP_ID1 0x81 +#define DA7219_CHIP_ID2 0x82 +#define DA7219_CHIP_REVISION 0x83 +#define DA7219_LDO_CTRL 0x90 +#define DA7219_IO_CTRL 0x91 +#define DA7219_GAIN_RAMP_CTRL 0x92 +#define DA7219_PC_COUNT 0x94 +#define DA7219_CP_VOL_THRESHOLD1 0x95 +#define DA7219_CP_DELAY 0x96 +#define DA7219_DIG_CTRL 0x99 +#define DA7219_ALC_CTRL2 0x9A +#define DA7219_ALC_CTRL3 0x9B +#define DA7219_ALC_NOISE 0x9C +#define DA7219_ALC_TARGET_MIN 0x9D +#define DA7219_ALC_TARGET_MAX 0x9E +#define DA7219_ALC_GAIN_LIMITS 0x9F +#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0 +#define DA7219_ALC_ANTICLIP_CTRL 0xA1 +#define DA7219_ALC_ANTICLIP_LEVEL 0xA2 +#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3 +#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4 +#define DA7219_DAC_NG_SETUP_TIME 0xAF +#define DA7219_DAC_NG_OFF_THRESH 0xB0 +#define DA7219_DAC_NG_ON_THRESH 0xB1 +#define DA7219_DAC_NG_CTRL 0xB2 +#define DA7219_TONE_GEN_CFG1 0xB4 +#define DA7219_TONE_GEN_CFG2 0xB5 +#define DA7219_TONE_GEN_CYCLES 0xB6 +#define DA7219_TONE_GEN_FREQ1_L 0xB7 +#define DA7219_TONE_GEN_FREQ1_U 0xB8 +#define DA7219_TONE_GEN_FREQ2_L 0xB9 +#define DA7219_TONE_GEN_FREQ2_U 0xBA +#define DA7219_TONE_GEN_ON_PER 0xBB +#define DA7219_TONE_GEN_OFF_PER 0xBC +#define DA7219_SYSTEM_STATUS 0xE0 +#define DA7219_SYSTEM_ACTIVE 0xFD + + +/* + * Bit Fields + */ + +#define DA7219_SWITCH_EN_MAX 0x1 + +/* DA7219_MIC_1_GAIN_STATUS = 0x6 */ +#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0) +#define DA7219_MIC_1_AMP_GAIN_MAX 0x7 + +/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */ +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0) + +/* DA7219_ADC_L_GAIN_STATUS = 0xA */ +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_L_GAIN_STATUS = 0xC */ +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_DAC_R_GAIN_STATUS = 0xD */ +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0) + +/* DA7219_HP_L_GAIN_STATUS = 0xE */ +#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_HP_R_GAIN_STATUS = 0xF */ +#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0) + +/* DA7219_MIC_1_SELECT = 0x10 */ +#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0 +#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0) + +/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */ +#define DA7219_I2C_TIMEOUT_EN_SHIFT 0 +#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0) + +/* DA7219_CIF_CTRL = 0x13 */ +#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0 +#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0) +#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7 +#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7) + +/* DA7219_SR_24_48 = 0x16 */ +#define DA7219_SR_24_48_SHIFT 0 +#define DA7219_SR_24_48_MASK (0x1 << 0) + +/* DA7219_SR = 0x17 */ +#define DA7219_SR_SHIFT 0 +#define DA7219_SR_MASK (0xF << 0) +#define DA7219_SR_8000 (0x01 << 0) +#define DA7219_SR_11025 (0x02 << 0) +#define DA7219_SR_12000 (0x03 << 0) +#define DA7219_SR_16000 (0x05 << 0) +#define DA7219_SR_22050 (0x06 << 0) +#define DA7219_SR_24000 (0x07 << 0) +#define DA7219_SR_32000 (0x09 << 0) +#define DA7219_SR_44100 (0x0A << 0) +#define DA7219_SR_48000 (0x0B << 0) +#define DA7219_SR_88200 (0x0E << 0) +#define DA7219_SR_96000 (0x0F << 0) + +/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */ +#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0 +#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0) + +/* 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_MCLK_SQR_EN_SHIFT 5 +#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) +#define DA7219_PLL_MODE_SHIFT 6 +#define DA7219_PLL_MODE_MASK (0x3 << 6) +#define DA7219_PLL_MODE_BYPASS (0x0 << 6) +#define DA7219_PLL_MODE_NORMAL (0x1 << 6) +#define DA7219_PLL_MODE_SRM (0x2 << 6) +#define DA7219_PLL_MODE_32KHZ (0x3 << 6) + +/* DA7219_PLL_FRAC_TOP = 0x22 */ +#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0) + +/* DA7219_PLL_FRAC_BOT = 0x23 */ +#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0 +#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0) + +/* DA7219_PLL_INTEGER = 0x24 */ +#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0 +#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0) + +/* DA7219_PLL_SRM_STS = 0x25 */ +#define DA7219_PLL_SRM_STATE_SHIFT 0 +#define DA7219_PLL_SRM_STATE_MASK (0xF << 0) +#define DA7219_PLL_SRM_STATUS_SHIFT 4 +#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4) +#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7) + +/* DA7219_DIG_ROUTING_DAI = 0x2A */ +#define DA7219_DAI_L_SRC_SHIFT 0 +#define DA7219_DAI_L_SRC_MASK (0x3 << 0) +#define DA7219_DAI_R_SRC_SHIFT 4 +#define DA7219_DAI_R_SRC_MASK (0x3 << 4) +#define DA7219_OUT_SRC_MAX 4 + +/* DA7219_DAI_CLK_MODE = 0x2B */ +#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0 +#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0) +#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0) +#define DA7219_DAI_CLK_POL_SHIFT 2 +#define DA7219_DAI_CLK_POL_MASK (0x1 << 2) +#define DA7219_DAI_CLK_POL_INV (0x1 << 2) +#define DA7219_DAI_WCLK_POL_SHIFT 3 +#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3) +#define DA7219_DAI_WCLK_POL_INV (0x1 << 3) +#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4 +#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4) +#define DA7219_DAI_CLK_EN_SHIFT 7 +#define DA7219_DAI_CLK_EN_MASK (0x1 << 7) + +/* DA7219_DAI_CTRL = 0x2C */ +#define DA7219_DAI_FORMAT_SHIFT 0 +#define DA7219_DAI_FORMAT_MASK (0x3 << 0) +#define DA7219_DAI_FORMAT_I2S (0x0 << 0) +#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0) +#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0) +#define DA7219_DAI_FORMAT_DSP (0x3 << 0) +#define DA7219_DAI_WORD_LENGTH_SHIFT 2 +#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2) +#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2) +#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2) +#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2) +#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2) +#define DA7219_DAI_CH_NUM_SHIFT 4 +#define DA7219_DAI_CH_NUM_MASK (0x3 << 4) +#define DA7219_DAI_CH_NUM_MAX 2 +#define DA7219_DAI_EN_SHIFT 7 +#define DA7219_DAI_EN_MASK (0x1 << 7) + +/* DA7219_DAI_TDM_CTRL = 0x2D */ +#define DA7219_DAI_TDM_CH_EN_SHIFT 0 +#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0) +#define DA7219_DAI_OE_SHIFT 6 +#define DA7219_DAI_OE_MASK (0x1 << 6) +#define DA7219_DAI_TDM_MODE_EN_SHIFT 7 +#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7) +#define DA7219_DAI_TDM_MAX_SLOTS 2 + +/* DA7219_DIG_ROUTING_DAC = 0x2E */ +#define DA7219_DAC_L_SRC_SHIFT 0 +#define DA7219_DAC_L_SRC_MASK (0x3 << 0) +#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0) +#define DA7219_DAC_L_MONO_SHIFT 3 +#define DA7219_DAC_L_MONO_MASK (0x1 << 3) +#define DA7219_DAC_R_SRC_SHIFT 4 +#define DA7219_DAC_R_SRC_MASK (0x3 << 4) +#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4) +#define DA7219_DAC_R_MONO_SHIFT 7 +#define DA7219_DAC_R_MONO_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL1 = 0x2F */ +#define DA7219_ALC_OFFSET_EN_SHIFT 0 +#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0) +#define DA7219_ALC_SYNC_MODE_SHIFT 1 +#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1) +#define DA7219_ALC_EN_SHIFT 3 +#define DA7219_ALC_EN_MASK (0x1 << 3) +#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4 +#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4) +#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5 +#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5) + +/* DA7219_DAI_OFFSET_LOWER = 0x30 */ +#define DA7219_DAI_OFFSET_LOWER_SHIFT 0 +#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0) + +/* DA7219_DAI_OFFSET_UPPER = 0x31 */ +#define DA7219_DAI_OFFSET_UPPER_SHIFT 0 +#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0) +#define DA7219_DAI_OFFSET_MAX 0x2FF + +/* DA7219_REFERENCES = 0x32 */ +#define DA7219_BIAS_EN_SHIFT 3 +#define DA7219_BIAS_EN_MASK (0x1 << 3) +#define DA7219_VMID_FAST_CHARGE_SHIFT 4 +#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4) + +/* DA7219_MIXIN_L_SELECT = 0x33 */ +#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXIN_L_GAIN = 0x34 */ +#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0 +#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0) +#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF + +/* DA7219_ADC_L_GAIN = 0x36 */ +#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F + +/* DA7219_ADC_FILTERS1 = 0x38 */ +#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_VOICE_HPF_CORNER_MAX 8 +#define DA7219_ADC_VOICE_EN_SHIFT 3 +#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_AUDIO_HPF_CORNER_MAX 4 +#define DA7219_ADC_HPF_EN_SHIFT 7 +#define DA7219_ADC_HPF_EN_MASK (0x1 << 7) +#define DA7219_HPF_MODE_SHIFT 0 +#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7)) +#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7)) +#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7)) +#define DA7219_HPF_MODE_MAX 3 + +/* DA7219_MIC_1_GAIN = 0x39 */ +#define DA7219_MIC_1_AMP_GAIN_SHIFT 0 +#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0) + +/* DA7219_SIDETONE_CTRL = 0x3A */ +#define DA7219_SIDETONE_MUTE_EN_SHIFT 6 +#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6) +#define DA7219_SIDETONE_EN_SHIFT 7 +#define DA7219_SIDETONE_EN_MASK (0x1 << 7) + +/* DA7219_SIDETONE_GAIN = 0x3B */ +#define DA7219_SIDETONE_GAIN_SHIFT 0 +#define DA7219_SIDETONE_GAIN_MASK (0xF << 0) +#define DA7219_SIDETONE_GAIN_MAX 0xE + +/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */ +#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0 +#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1 +#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2 +#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0) +#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1) + +/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */ +#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0 +#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0) + +/* DA7219_DAC_FILTERS5 = 0x40 */ +#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4 +#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4) +#define DA7219_DAC_SOFTMUTE_RATE_MAX 7 +#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7 +#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS2 = 0x41 */ +#define DA7219_DAC_EQ_BAND1_SHIFT 0 +#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND2_SHIFT 4 +#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4) +#define DA7219_DAC_EQ_BAND_MAX 0xF + +/* DA7219_DAC_FILTERS3 = 0x42 */ +#define DA7219_DAC_EQ_BAND3_SHIFT 0 +#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0) +#define DA7219_DAC_EQ_BAND4_SHIFT 4 +#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4) + +/* DA7219_DAC_FILTERS4 = 0x43 */ +#define DA7219_DAC_EQ_BAND5_SHIFT 0 +#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0) +#define DA7219_DAC_EQ_EN_SHIFT 7 +#define DA7219_DAC_EQ_EN_MASK (0x1 << 7) + +/* DA7219_DAC_FILTERS1 = 0x44 */ +#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0 +#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0) +#define DA7219_DAC_VOICE_EN_SHIFT 3 +#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3) +#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4 +#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4) +#define DA7219_DAC_HPF_EN_SHIFT 7 +#define DA7219_DAC_HPF_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_GAIN = 0x45 */ +#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0) +#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F +#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0) + +/* DA7219_DAC_R_GAIN = 0x46 */ +#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0 +#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0) + +/* DA7219_CP_CTRL = 0x47 */ +#define DA7219_CP_MCHANGE_SHIFT 4 +#define DA7219_CP_MCHANGE_MASK (0x3 << 4) +#define DA7219_CP_MCHANGE_REL_MASK 0x3 +#define DA7219_CP_MCHANGE_MAX 3 +#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1 +#define DA7219_CP_MCHANGE_DAC_VOL 0x2 +#define DA7219_CP_MCHANGE_SIG_MAG 0x3 +#define DA7219_CP_EN_SHIFT 7 +#define DA7219_CP_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_GAIN = 0x48 */ +#define DA7219_HP_L_AMP_GAIN_SHIFT 0 +#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0) +#define DA7219_HP_AMP_GAIN_MAX 0x3F +#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0) + +/* DA7219_HP_R_GAIN = 0x49 */ +#define DA7219_HP_R_AMP_GAIN_SHIFT 0 +#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0) + +/* DA7219_MIXOUT_L_SELECT = 0x4B */ +#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_MIXOUT_R_SELECT = 0x4C */ +#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0 +#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0) + +/* DA7219_SYSTEM_MODES_INPUT = 0x50 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_ADC_MODE_SHIFT 1 +#define DA7219_ADC_MODE_MASK (0x7F << 1) + +/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */ +#define DA7219_MODE_SUBMIT_SHIFT 0 +#define DA7219_MODE_SUBMIT_MASK (0x1 << 0) +#define DA7219_DAC_MODE_SHIFT 1 +#define DA7219_DAC_MODE_MASK (0x7F << 1) + +/* DA7219_MICBIAS_CTRL = 0x62 */ +#define DA7219_MICBIAS1_LEVEL_SHIFT 0 +#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0) +#define DA7219_MICBIAS1_EN_SHIFT 3 +#define DA7219_MICBIAS1_EN_MASK (0x1 << 3) + +/* DA7219_MIC_1_CTRL = 0x63 */ +#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIC_1_AMP_EN_SHIFT 7 +#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXIN_L_CTRL = 0x65 */ +#define DA7219_MIXIN_L_MIX_EN_SHIFT 3 +#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3) +#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_MIXIN_L_AMP_EN_SHIFT 7 +#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_ADC_L_CTRL = 0x67 */ +#define DA7219_ADC_L_BIAS_SHIFT 0 +#define DA7219_ADC_L_BIAS_MASK (0x3 << 0) +#define DA7219_ADC_L_RAMP_EN_SHIFT 5 +#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_ADC_L_MUTE_EN_SHIFT 6 +#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_ADC_L_EN_SHIFT 7 +#define DA7219_ADC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_L_CTRL = 0x69 */ +#define DA7219_DAC_L_RAMP_EN_SHIFT 5 +#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_L_MUTE_EN_SHIFT 6 +#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_L_EN_SHIFT 7 +#define DA7219_DAC_L_EN_MASK (0x1 << 7) + +/* DA7219_DAC_R_CTRL = 0x6A */ +#define DA7219_DAC_R_RAMP_EN_SHIFT 5 +#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5) +#define DA7219_DAC_R_MUTE_EN_SHIFT 6 +#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6) +#define DA7219_DAC_R_EN_SHIFT 7 +#define DA7219_DAC_R_EN_MASK (0x1 << 7) + +/* DA7219_HP_L_CTRL = 0x6B */ +#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_L_AMP_OE_SHIFT 3 +#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_L_AMP_EN_SHIFT 7 +#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_HP_R_CTRL = 0x6C */ +#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2 +#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2) +#define DA7219_HP_R_AMP_OE_SHIFT 3 +#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3) +#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4 +#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4) +#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5 +#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5) +#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6 +#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6) +#define DA7219_HP_R_AMP_EN_SHIFT 7 +#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_L_CTRL = 0x6E */ +#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7) + +/* DA7219_MIXOUT_R_CTRL = 0x6F */ +#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7 +#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7) + +/* DA7219_CHIP_ID1 = 0x81 */ +#define DA7219_CHIP_ID1_SHIFT 0 +#define DA7219_CHIP_ID1_MASK (0xFF << 0) + +/* DA7219_CHIP_ID2 = 0x82 */ +#define DA7219_CHIP_ID2_SHIFT 0 +#define DA7219_CHIP_ID2_MASK (0xFF << 0) + +/* DA7219_CHIP_REVISION = 0x83 */ +#define DA7219_CHIP_MINOR_SHIFT 0 +#define DA7219_CHIP_MINOR_MASK (0xF << 0) +#define DA7219_CHIP_MAJOR_SHIFT 4 +#define DA7219_CHIP_MAJOR_MASK (0xF << 4) + +/* DA7219_LDO_CTRL = 0x90 */ +#define DA7219_LDO_LEVEL_SELECT_SHIFT 4 +#define DA7219_LDO_LEVEL_SELECT_MASK (0x3 << 4) +#define DA7219_LDO_EN_SHIFT 7 +#define DA7219_LDO_EN_MASK (0x1 << 7) + +/* DA7219_IO_CTRL = 0x91 */ +#define DA7219_IO_VOLTAGE_LEVEL_SHIFT 0 +#define DA7219_IO_VOLTAGE_LEVEL_MASK (0x1 << 0) +#define DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V 0 +#define DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V 1 + +/* DA7219_GAIN_RAMP_CTRL = 0x92 */ +#define DA7219_GAIN_RAMP_RATE_SHIFT 0 +#define DA7219_GAIN_RAMP_RATE_MASK (0x3 << 0) +#define DA7219_GAIN_RAMP_RATE_MAX 4 + +/* DA7219_PC_COUNT = 0x94 */ +#define DA7219_PC_FREERUN_SHIFT 0 +#define DA7219_PC_FREERUN_MASK (0x1 << 0) +#define DA7219_PC_RESYNC_AUTO_SHIFT 1 +#define DA7219_PC_RESYNC_AUTO_MASK (0x1 << 1) + +/* DA7219_CP_VOL_THRESHOLD1 = 0x95 */ +#define DA7219_CP_THRESH_VDD2_SHIFT 0 +#define DA7219_CP_THRESH_VDD2_MASK (0x3F << 0) +#define DA7219_CP_THRESH_VDD2_MAX 0x3F + +/* DA7219_DIG_CTRL = 0x99 */ +#define DA7219_DAC_L_INV_SHIFT 3 +#define DA7219_DAC_L_INV_MASK (0x1 << 3) +#define DA7219_DAC_R_INV_SHIFT 7 +#define DA7219_DAC_R_INV_MASK (0x1 << 7) + +/* DA7219_ALC_CTRL2 = 0x9A */ +#define DA7219_ALC_ATTACK_SHIFT 0 +#define DA7219_ALC_ATTACK_MASK (0xF << 0) +#define DA7219_ALC_ATTACK_MAX 13 +#define DA7219_ALC_RELEASE_SHIFT 4 +#define DA7219_ALC_RELEASE_MASK (0xF << 4) +#define DA7219_ALC_RELEASE_MAX 11 + +/* DA7219_ALC_CTRL3 = 0x9B */ +#define DA7219_ALC_HOLD_SHIFT 0 +#define DA7219_ALC_HOLD_MASK (0xF << 0) +#define DA7219_ALC_HOLD_MAX 16 +#define DA7219_ALC_INTEG_ATTACK_SHIFT 4 +#define DA7219_ALC_INTEG_ATTACK_MASK (0x3 << 4) +#define DA7219_ALC_INTEG_RELEASE_SHIFT 6 +#define DA7219_ALC_INTEG_RELEASE_MASK (0x3 << 6) +#define DA7219_ALC_INTEG_MAX 4 + +/* DA7219_ALC_NOISE = 0x9C */ +#define DA7219_ALC_NOISE_SHIFT 0 +#define DA7219_ALC_NOISE_MASK (0x3F << 0) +#define DA7219_ALC_THRESHOLD_MAX 0x3F + +/* DA7219_ALC_TARGET_MIN = 0x9D */ +#define DA7219_ALC_THRESHOLD_MIN_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MIN_MASK (0x3F << 0) + +/* DA7219_ALC_TARGET_MAX = 0x9E */ +#define DA7219_ALC_THRESHOLD_MAX_SHIFT 0 +#define DA7219_ALC_THRESHOLD_MAX_MASK (0x3F << 0) + +/* DA7219_ALC_GAIN_LIMITS = 0x9F */ +#define DA7219_ALC_ATTEN_MAX_SHIFT 0 +#define DA7219_ALC_ATTEN_MAX_MASK (0xF << 0) +#define DA7219_ALC_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_GAIN_MAX_MASK (0xF << 4) +#define DA7219_ALC_ATTEN_GAIN_MAX 0xF + +/* DA7219_ALC_ANA_GAIN_LIMITS = 0xA0 */ +#define DA7219_ALC_ANA_GAIN_MIN_SHIFT 0 +#define DA7219_ALC_ANA_GAIN_MIN_MASK (0x7 << 0) +#define DA7219_ALC_ANA_GAIN_MIN 0x1 +#define DA7219_ALC_ANA_GAIN_MAX_SHIFT 4 +#define DA7219_ALC_ANA_GAIN_MAX_MASK (0x7 << 4) +#define DA7219_ALC_ANA_GAIN_MAX 0x7 + +/* DA7219_ALC_ANTICLIP_CTRL = 0xA1 */ +#define DA7219_ALC_ANTICLIP_STEP_SHIFT 0 +#define DA7219_ALC_ANTICLIP_STEP_MASK (0x3 << 0) +#define DA7219_ALC_ANTICLIP_STEP_MAX 4 +#define DA7219_ALC_ANTIPCLIP_EN_SHIFT 7 +#define DA7219_ALC_ANTIPCLIP_EN_MASK (0x1 << 7) + +/* DA7219_ALC_ANTICLIP_LEVEL = 0xA2 */ +#define DA7219_ALC_ANTICLIP_LEVEL_SHIFT 0 +#define DA7219_ALC_ANTICLIP_LEVEL_MASK (0x7F << 0) + +/* DA7219_ALC_OFFSET_AUTO_M_L = 0xA3 */ +#define DA7219_ALC_OFFSET_AUTO_M_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_M_L_MASK (0xFF << 0) + +/* DA7219_ALC_OFFSET_AUTO_U_L = 0xA4 */ +#define DA7219_ALC_OFFSET_AUTO_U_L_SHIFT 0 +#define DA7219_ALC_OFFSET_AUTO_U_L_MASK (0xF << 0) + +/* DA7219_DAC_NG_SETUP_TIME = 0xAF */ +#define DA7219_DAC_NG_SETUP_TIME_SHIFT 0 +#define DA7219_DAC_NG_SETUP_TIME_MASK (0x3 << 0) +#define DA7219_DAC_NG_SETUP_TIME_MAX 4 +#define DA7219_DAC_NG_RAMPUP_RATE_SHIFT 2 +#define DA7219_DAC_NG_RAMPUP_RATE_MASK (0x1 << 2) +#define DA7219_DAC_NG_RAMPDN_RATE_SHIFT 3 +#define DA7219_DAC_NG_RAMPDN_RATE_MASK (0x1 << 3) +#define DA7219_DAC_NG_RAMP_RATE_MAX 2 + +/* DA7219_DAC_NG_OFF_THRESH = 0xB0 */ +#define DA7219_DAC_NG_OFF_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_OFF_THRESHOLD_MASK (0x7 << 0) +#define DA7219_DAC_NG_THRESHOLD_MAX 0x7 + +/* DA7219_DAC_NG_ON_THRESH = 0xB1 */ +#define DA7219_DAC_NG_ON_THRESHOLD_SHIFT 0 +#define DA7219_DAC_NG_ON_THRESHOLD_MASK (0x7 << 0) + +/* DA7219_DAC_NG_CTRL = 0xB2 */ +#define DA7219_DAC_NG_EN_SHIFT 7 +#define DA7219_DAC_NG_EN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG1 = 0xB4 */ +#define DA7219_DTMF_REG_SHIFT 0 +#define DA7219_DTMF_REG_MASK (0xF << 0) +#define DA7219_DTMF_REG_MAX 16 +#define DA7219_DTMF_EN_SHIFT 4 +#define DA7219_DTMF_EN_MASK (0x1 << 4) +#define DA7219_START_STOPN_SHIFT 7 +#define DA7219_START_STOPN_MASK (0x1 << 7) + +/* DA7219_TONE_GEN_CFG2 = 0xB5 */ +#define DA7219_SWG_SEL_SHIFT 0 +#define DA7219_SWG_SEL_MASK (0x3 << 0) +#define DA7219_SWG_SEL_MAX 4 +#define DA7219_SWG_SEL_SRAMP (0x3 << 0) +#define DA7219_TONE_GEN_GAIN_SHIFT 4 +#define DA7219_TONE_GEN_GAIN_MASK (0xF << 4) +#define DA7219_TONE_GEN_GAIN_MAX 0xF +#define DA7219_TONE_GEN_GAIN_MINUS_9DB (0x3 << 4) +#define DA7219_TONE_GEN_GAIN_MINUS_15DB (0x5 << 4) + +/* DA7219_TONE_GEN_CYCLES = 0xB6 */ +#define DA7219_BEEP_CYCLES_SHIFT 0 +#define DA7219_BEEP_CYCLES_MASK (0x7 << 0) + +/* DA7219_TONE_GEN_FREQ1_L = 0xB7 */ +#define DA7219_FREQ1_L_SHIFT 0 +#define DA7219_FREQ1_L_MASK (0xFF << 0) +#define DA7219_FREQ_MAX 0xFFFF + +/* DA7219_TONE_GEN_FREQ1_U = 0xB8 */ +#define DA7219_FREQ1_U_SHIFT 0 +#define DA7219_FREQ1_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_L = 0xB9 */ +#define DA7219_FREQ2_L_SHIFT 0 +#define DA7219_FREQ2_L_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_FREQ2_U = 0xBA */ +#define DA7219_FREQ2_U_SHIFT 0 +#define DA7219_FREQ2_U_MASK (0xFF << 0) + +/* DA7219_TONE_GEN_ON_PER = 0xBB */ +#define DA7219_BEEP_ON_PER_SHIFT 0 +#define DA7219_BEEP_ON_PER_MASK (0x3F << 0) +#define DA7219_BEEP_ON_OFF_MAX 0x3F + +/* DA7219_TONE_GEN_OFF_PER = 0xBC */ +#define DA7219_BEEP_OFF_PER_SHIFT 0 +#define DA7219_BEEP_OFF_PER_MASK (0x3F << 0) + +/* DA7219_SYSTEM_STATUS = 0xE0 */ +#define DA7219_SC1_BUSY_SHIFT 0 +#define DA7219_SC1_BUSY_MASK (0x1 << 0) +#define DA7219_SC2_BUSY_SHIFT 1 +#define DA7219_SC2_BUSY_MASK (0x1 << 1) + +/* DA7219_SYSTEM_ACTIVE = 0xFD */ +#define DA7219_SYSTEM_ACTIVE_SHIFT 0 +#define DA7219_SYSTEM_ACTIVE_MASK (0x1 << 0) + + +/* + * General defines & data + */ + +/* Register inversion */ +#define DA7219_NO_INVERT 0 +#define DA7219_INVERT 1 + +/* Byte related defines */ +#define DA7219_BYTE_SHIFT 8 +#define DA7219_BYTE_MASK 0xFF + +/* PLL Output Frequencies */ +#define DA7219_PLL_FREQ_OUT_90316 90316800 +#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 + +/* SRM */ +#define DA7219_SRM_CHECK_RETRIES 8 + +enum da7219_clk_src { + DA7219_CLKSRC_MCLK = 0, + DA7219_CLKSRC_MCLK_SQR, +}; + +enum da7219_sys_clk { + DA7219_SYSCLK_MCLK = 0, + DA7219_SYSCLK_PLL, + DA7219_SYSCLK_PLL_SRM, + DA7219_SYSCLK_PLL_32KHZ +}; + +/* Regulators */ +enum da7219_supplies { + DA7219_SUPPLY_VDD = 0, + DA7219_SUPPLY_VDDMIC, + DA7219_SUPPLY_VDDIO, + DA7219_NUM_SUPPLIES, +}; + +struct da7219_aad_priv; + +/* Private data */ +struct da7219_priv { + struct da7219_aad_priv *aad; + struct da7219_pdata *pdata; + + struct regulator_bulk_data supplies[DA7219_NUM_SUPPLIES]; + struct regmap *regmap; + struct mutex lock; + + struct clk *mclk; + unsigned int mclk_rate; + int clk_src; + + bool master; + bool alc_en; +}; + +#endif /* __DA7219_H */ -- cgit v1.2.3 From 1d957d862ac782eaf5803d4d4cf167708e4dc147 Mon Sep 17 00:00:00 2001 From: Maruthi Srinivas Bayyavarapu Date: Fri, 25 Sep 2015 17:48:22 -0400 Subject: ASoC: dwc: support dw i2s in slave mode dw i2s controller can work in slave mode, codec being master. dw i2s is made to support master/slave operation, by reading dwc register. Signed-off-by: Maruthi Bayyavarapu Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- include/sound/designware_i2s.h | 2 + sound/soc/dwc/designware_i2s.c | 92 +++++++++++++++++++++++++----------------- 2 files changed, 57 insertions(+), 37 deletions(-) (limited to 'include') diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 3a8fca9409a7..8966ba7c9629 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -38,6 +38,8 @@ struct i2s_clk_config_data { struct i2s_platform_data { #define DWC_I2S_PLAY (1 << 0) #define DWC_I2S_RECORD (1 << 1) + #define DW_I2S_SLAVE (1 << 2) + #define DW_I2S_MASTER (1 << 3) unsigned int cap; int channel; u32 snd_fmts; diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b46b64e..3a52f82b5523 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -273,23 +273,25 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, config->sample_rate = params_rate(params); - if (dev->i2s_clk_cfg) { - ret = dev->i2s_clk_cfg(config); - if (ret < 0) { - dev_err(dev->dev, "runtime audio clk config fail\n"); - return ret; - } - } else { - u32 bitclk = config->sample_rate * config->data_width * 2; - - ret = clk_set_rate(dev->clk, bitclk); - if (ret) { - dev_err(dev->dev, "Can't set I2S clock rate: %d\n", - ret); - return ret; + if (dev->capability & DW_I2S_MASTER) { + if (dev->i2s_clk_cfg) { + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + } else { + u32 bitclk = config->sample_rate * + config->data_width * 2; + + ret = clk_set_rate(dev->clk, bitclk); + if (ret) { + dev_err(dev->dev, "Can't set I2S clock rate: %d\n", + ret); + return ret; + } } } - return 0; } @@ -357,7 +359,8 @@ static int dw_i2s_suspend(struct snd_soc_dai *dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - clk_disable(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable(dev->clk); return 0; } @@ -365,7 +368,8 @@ static int dw_i2s_resume(struct snd_soc_dai *dai) { struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); - clk_enable(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_enable(dev->clk); return 0; } @@ -443,6 +447,14 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, dw_i2s_dai->capture.rates = rates; } + if (COMP1_MODE_EN(comp1)) { + dev_dbg(dev->dev, "designware: i2s master mode supported\n"); + dev->capability |= DW_I2S_MASTER; + } else { + dev_dbg(dev->dev, "designware: i2s slave mode supported\n"); + dev->capability |= DW_I2S_SLAVE; + } + return 0; } @@ -529,6 +541,7 @@ static int dw_i2s_probe(struct platform_device *pdev) struct resource *res; int ret; struct snd_soc_dai_driver *dw_i2s_dai; + const char *clk_id; dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); if (!dev) { @@ -550,32 +563,35 @@ static int dw_i2s_probe(struct platform_device *pdev) return PTR_ERR(dev->i2s_base); dev->dev = &pdev->dev; + if (pdata) { + dev->capability = pdata->cap; + clk_id = NULL; ret = dw_configure_dai_by_pd(dev, dw_i2s_dai, res, pdata); - if (ret < 0) - return ret; + } else { + clk_id = "i2sclk"; + ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); + } + if (ret < 0) + return ret; - dev->capability = pdata->cap; - dev->i2s_clk_cfg = pdata->i2s_clk_cfg; - if (!dev->i2s_clk_cfg) { - dev_err(&pdev->dev, "no clock configure method\n"); - return -ENODEV; + if (dev->capability & DW_I2S_MASTER) { + if (pdata) { + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + if (!dev->i2s_clk_cfg) { + dev_err(&pdev->dev, "no clock configure method\n"); + return -ENODEV; + } } + dev->clk = devm_clk_get(&pdev->dev, clk_id); - dev->clk = devm_clk_get(&pdev->dev, NULL); - } else { - ret = dw_configure_dai_by_dt(dev, dw_i2s_dai, res); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + ret = clk_prepare_enable(dev->clk); if (ret < 0) return ret; - - dev->clk = devm_clk_get(&pdev->dev, "i2sclk"); } - if (IS_ERR(dev->clk)) - return PTR_ERR(dev->clk); - - ret = clk_prepare_enable(dev->clk); - if (ret < 0) - return ret; dev_set_drvdata(&pdev->dev, dev); ret = devm_snd_soc_register_component(&pdev->dev, &dw_i2s_component, @@ -597,7 +613,8 @@ static int dw_i2s_probe(struct platform_device *pdev) return 0; err_clk_disable: - clk_disable_unprepare(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable_unprepare(dev->clk); return ret; } @@ -605,7 +622,8 @@ static int dw_i2s_remove(struct platform_device *pdev) { struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); - clk_disable_unprepare(dev->clk); + if (dev->capability & DW_I2S_MASTER) + clk_disable_unprepare(dev->clk); return 0; } -- cgit v1.2.3 From dcc448e619194098b24477a6d56af50c57f26f1d Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Wed, 7 Oct 2015 12:42:22 +0200 Subject: ASoC: rsnd: Remove obsolete platform data support Since commit 3d7608e4c169af03 ("ARM: shmobile: bockw: remove legacy board file and config"), Renesas R-Car SoCs are only supported in generic DT-only ARM multi-platform builds. The driver doesn't need to use platform data anymore, hence remove platform data configuration. Move to sound/soc/sh/rcar/, as it's no longer needed by platform code. Signed-off-by: Geert Uytterhoeven Signed-off-by: Mark Brown --- include/sound/rcar_snd.h | 117 ------------------------------------------- sound/soc/sh/rcar/core.c | 19 ++----- sound/soc/sh/rcar/rcar_snd.h | 117 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 3 +- sound/soc/sh/rcar/ssi.c | 3 -- 5 files changed, 124 insertions(+), 135 deletions(-) delete mode 100644 include/sound/rcar_snd.h create mode 100644 sound/soc/sh/rcar/rcar_snd.h (limited to 'include') diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h deleted file mode 100644 index d8e33d38da43..000000000000 --- a/include/sound/rcar_snd.h +++ /dev/null @@ -1,117 +0,0 @@ -/* - * Renesas R-Car SRU/SCU/SSIU/SSI support - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto - * - * 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 RCAR_SND_H -#define RCAR_SND_H - - -#define RSND_GEN1_SRU 0 -#define RSND_GEN1_ADG 1 -#define RSND_GEN1_SSI 2 - -#define RSND_GEN2_SCU 0 -#define RSND_GEN2_ADG 1 -#define RSND_GEN2_SSIU 2 -#define RSND_GEN2_SSI 3 - -#define RSND_BASE_MAX 4 - -/* - * flags - * - * 0xAB000000 - * - * A : clock sharing settings - * B : SSI direction - */ -#define RSND_SSI_CLK_PIN_SHARE (1 << 31) -#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ - -#define RSND_SSI(_dma_id, _irq, _flags) \ -{ .dma_id = _dma_id, .irq = _irq, .flags = _flags } -#define RSND_SSI_UNUSED \ -{ .dma_id = -1, .irq = -1, .flags = 0 } - -struct rsnd_ssi_platform_info { - int dma_id; - int irq; - u32 flags; -}; - -#define RSND_SRC(rate, _dma_id) \ -{ .convert_rate = rate, .dma_id = _dma_id, } -#define RSND_SRC_UNUSED \ -{ .convert_rate = 0, .dma_id = -1, } - -struct rsnd_src_platform_info { - u32 convert_rate; /* sampling rate convert */ - int dma_id; /* for Gen2 SCU */ - int irq; -}; - -/* - * flags - */ -struct rsnd_ctu_platform_info { - u32 flags; -}; - -struct rsnd_mix_platform_info { - u32 flags; -}; - -struct rsnd_dvc_platform_info { - u32 flags; -}; - -struct rsnd_dai_path_info { - struct rsnd_ssi_platform_info *ssi; - struct rsnd_src_platform_info *src; - struct rsnd_ctu_platform_info *ctu; - struct rsnd_mix_platform_info *mix; - struct rsnd_dvc_platform_info *dvc; -}; - -struct rsnd_dai_platform_info { - struct rsnd_dai_path_info playback; - struct rsnd_dai_path_info capture; -}; - -/* - * flags - * - * 0x0000000A - * - * A : generation - */ -#define RSND_GEN_MASK (0xF << 0) -#define RSND_GEN1 (1 << 0) /* fixme */ -#define RSND_GEN2 (2 << 0) /* fixme */ - -struct rcar_snd_info { - u32 flags; - struct rsnd_ssi_platform_info *ssi_info; - int ssi_info_nr; - struct rsnd_src_platform_info *src_info; - int src_info_nr; - struct rsnd_ctu_platform_info *ctu_info; - int ctu_info_nr; - struct rsnd_mix_platform_info *mix_info; - int mix_info_nr; - struct rsnd_dvc_platform_info *dvc_info; - int dvc_info_nr; - struct rsnd_dai_platform_info *dai_info; - int dai_info_nr; - int (*start)(int id); - int (*stop)(int id); -}; - -#endif diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index eec294da81e3..6ef9a884ca7c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1236,20 +1236,11 @@ static int rsnd_probe(struct platform_device *pdev) }; int ret, i; - info = NULL; - of_data = NULL; - if (of_id) { - info = devm_kzalloc(&pdev->dev, - sizeof(struct rcar_snd_info), GFP_KERNEL); - of_data = of_id->data; - } else { - info = pdev->dev.platform_data; - } - - if (!info) { - dev_err(dev, "driver needs R-Car sound information\n"); - return -ENODEV; - } + info = devm_kzalloc(&pdev->dev, sizeof(struct rcar_snd_info), + GFP_KERNEL); + if (!info) + return -ENOMEM; + of_data = of_id->data; /* * init priv data diff --git a/sound/soc/sh/rcar/rcar_snd.h b/sound/soc/sh/rcar/rcar_snd.h new file mode 100644 index 000000000000..d8e33d38da43 --- /dev/null +++ b/sound/soc/sh/rcar/rcar_snd.h @@ -0,0 +1,117 @@ +/* + * Renesas R-Car SRU/SCU/SSIU/SSI support + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * 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 RCAR_SND_H +#define RCAR_SND_H + + +#define RSND_GEN1_SRU 0 +#define RSND_GEN1_ADG 1 +#define RSND_GEN1_SSI 2 + +#define RSND_GEN2_SCU 0 +#define RSND_GEN2_ADG 1 +#define RSND_GEN2_SSIU 2 +#define RSND_GEN2_SSI 3 + +#define RSND_BASE_MAX 4 + +/* + * flags + * + * 0xAB000000 + * + * A : clock sharing settings + * B : SSI direction + */ +#define RSND_SSI_CLK_PIN_SHARE (1 << 31) +#define RSND_SSI_NO_BUSIF (1 << 30) /* SSI+DMA without BUSIF */ + +#define RSND_SSI(_dma_id, _irq, _flags) \ +{ .dma_id = _dma_id, .irq = _irq, .flags = _flags } +#define RSND_SSI_UNUSED \ +{ .dma_id = -1, .irq = -1, .flags = 0 } + +struct rsnd_ssi_platform_info { + int dma_id; + int irq; + u32 flags; +}; + +#define RSND_SRC(rate, _dma_id) \ +{ .convert_rate = rate, .dma_id = _dma_id, } +#define RSND_SRC_UNUSED \ +{ .convert_rate = 0, .dma_id = -1, } + +struct rsnd_src_platform_info { + u32 convert_rate; /* sampling rate convert */ + int dma_id; /* for Gen2 SCU */ + int irq; +}; + +/* + * flags + */ +struct rsnd_ctu_platform_info { + u32 flags; +}; + +struct rsnd_mix_platform_info { + u32 flags; +}; + +struct rsnd_dvc_platform_info { + u32 flags; +}; + +struct rsnd_dai_path_info { + struct rsnd_ssi_platform_info *ssi; + struct rsnd_src_platform_info *src; + struct rsnd_ctu_platform_info *ctu; + struct rsnd_mix_platform_info *mix; + struct rsnd_dvc_platform_info *dvc; +}; + +struct rsnd_dai_platform_info { + struct rsnd_dai_path_info playback; + struct rsnd_dai_path_info capture; +}; + +/* + * flags + * + * 0x0000000A + * + * A : generation + */ +#define RSND_GEN_MASK (0xF << 0) +#define RSND_GEN1 (1 << 0) /* fixme */ +#define RSND_GEN2 (2 << 0) /* fixme */ + +struct rcar_snd_info { + u32 flags; + struct rsnd_ssi_platform_info *ssi_info; + int ssi_info_nr; + struct rsnd_src_platform_info *src_info; + int src_info_nr; + struct rsnd_ctu_platform_info *ctu_info; + int ctu_info_nr; + struct rsnd_mix_platform_info *mix_info; + int mix_info_nr; + struct rsnd_dvc_platform_info *dvc_info; + int dvc_info_nr; + struct rsnd_dai_platform_info *dai_info; + int dai_info_nr; + int (*start)(int id); + int (*stop)(int id); +}; + +#endif diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index e4068d78616c..e9fef53968b4 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -21,10 +21,11 @@ #include #include #include -#include #include #include +#include "rcar_snd.h" + /* * pseudo register * diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 5e05f9422073..842a35b1363a 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -700,9 +700,6 @@ static void rsnd_of_parse_ssi(struct platform_device *pdev, struct device *dev = &pdev->dev; int nr, i; - if (!of_data) - return; - node = rsnd_ssi_of_node(priv); if (!node) return; -- cgit v1.2.3 From 6e7c444318699496e6e6f30c875cf67534aeccc6 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Wed, 7 Oct 2015 14:27:11 +0100 Subject: ASoC: da7213: Add support to handle mclk data provided to driver Driver now can make use of mclk data, if provided, to set, enable and disable the clock source. As part of this, the choice to enable clock squaring is dealt with as part of dai_sysclk() call rather than as platform data. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- include/sound/da7213.h | 3 --- sound/soc/codecs/da7213.c | 67 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 8 ++++-- 3 files changed, 62 insertions(+), 16 deletions(-) (limited to 'include') diff --git a/include/sound/da7213.h b/include/sound/da7213.h index 673f5c39cbf2..e7eac8979995 100644 --- a/include/sound/da7213.h +++ b/include/sound/da7213.h @@ -44,9 +44,6 @@ struct da7213_platform_data { enum da7213_dmic_data_sel dmic_data_sel; enum da7213_dmic_samplephase dmic_samplephase; enum da7213_dmic_clk_rate dmic_clk_rate; - - /* MCLK squaring config */ - bool mclk_squaring; }; #endif /* _DA7213_PDATA_H */ diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index ab1486b04c30..7278f93460c1 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -12,6 +12,7 @@ * option) any later version. */ +#include #include #include #include @@ -1222,23 +1223,44 @@ static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq)) + return 0; + + if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) { + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } switch (clk_id) { case DA7213_CLKSRC_MCLK: - if ((freq == 32768) || - ((freq >= 5000000) && (freq <= 54000000))) { - da7213->mclk_rate = freq; - return 0; - } else { - dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", - freq); - return -EINVAL; - } + da7213->mclk_squarer_en = false; + break; + case DA7213_CLKSRC_MCLK_SQR: + da7213->mclk_squarer_en = true; break; default: dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } + + da7213->clk_src = clk_id; + + if (da7213->mclk) { + freq = clk_round_rate(da7213->mclk, freq); + ret = clk_set_rate(da7213->mclk, freq); + if (ret) { + dev_err(codec_dai->dev, "Failed to set clock rate %d\n", + freq); + return ret; + } + } + + da7213->mclk_rate = freq; + + return 0; } /* Supported PLL input frequencies are 5MHz - 54MHz. */ @@ -1366,12 +1388,25 @@ static struct snd_soc_dai_driver da7213_dai = { static int da7213_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + /* MCLK */ + if (da7213->mclk) { + ret = clk_prepare_enable(da7213->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable mclk\n"); + return ret; + } + } + /* Enable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, @@ -1382,6 +1417,10 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, /* Disable VMID reference & master bias */ snd_soc_update_bits(codec, DA7213_REFERENCES, DA7213_VMID_EN | DA7213_BIAS_EN, 0); + + /* MCLK */ + if (da7213->mclk) + clk_disable_unprepare(da7213->mclk); break; } return 0; @@ -1618,9 +1657,15 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_MASK | DA7213_DMIC_SAMPLEPHASE_MASK | DA7213_DMIC_CLK_RATE_MASK, dmic_cfg); + } - /* Set MCLK squaring */ - da7213->mclk_squarer_en = pdata->mclk_squaring; + /* Check if MCLK provided */ + da7213->mclk = devm_clk_get(codec->dev, "mclk"); + if (IS_ERR(da7213->mclk)) { + if (PTR_ERR(da7213->mclk) != -ENOENT) + return PTR_ERR(da7213->mclk); + else + da7213->mclk = NULL; } return 0; diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 9cb9ddd01282..030fd691b076 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -13,6 +13,7 @@ #ifndef _DA7213_H #define _DA7213_H +#include #include #include @@ -504,14 +505,17 @@ #define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 #define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 -enum clk_src { - DA7213_CLKSRC_MCLK +enum da7213_clk_src { + DA7213_CLKSRC_MCLK = 0, + DA7213_CLKSRC_MCLK_SQR, }; /* Codec private data */ struct da7213_priv { struct regmap *regmap; + struct clk *mclk; unsigned int mclk_rate; + int clk_src; bool master; bool mclk_squarer_en; bool srm_en; -- cgit v1.2.3 From 1b5e6167c27e1d3be33155baf9660768ac74aae0 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 8 Oct 2015 09:48:05 +0100 Subject: ALSA: hdac: Copy codec helpers to core The current codec helpers are local to hda code and needs to be moved to core so that other users can use it. The helpers to read/write the codec and to check the power state of widgets is copied Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 6 ++++ sound/hda/hdac_device.c | 81 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 49bc836fcd84..26e956f4b7c6 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -147,6 +147,12 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, unsigned int format); +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm); +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state); /** * snd_hdac_read_parm - read a codec parameter * @codec: the codec object diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index db96042a497f..b3b0ad289df1 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -952,3 +952,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, return true; } EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format); + +static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + unsigned int res; + + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) + return -1; + + return res; +} + +static int codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); + + return snd_hdac_exec_verb(hdac, cmd, flags, NULL); +} + +/** + * snd_hdac_codec_read - send a command and get the response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command and read the corresponding response. + * + * Returns the obtained response value, or -1 for an error. + */ +int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_read(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_read); + +/** + * snd_hdac_codec_write - send a single command without waiting for response + * @hdac: the HDAC device + * @nid: NID to send the command + * @flags: optional bit flags + * @verb: the verb to send + * @parm: the parameter for the verb + * + * Send a single command without waiting for response. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, + int flags, unsigned int verb, unsigned int parm) +{ + return codec_write(hdac, nid, flags, verb, parm); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_write); + +/* + * snd_hdac_check_power_state: check whether the actual power state matches + * with the target state + * + * @hdac: the HDAC device + * @nid: NID to send the command + * @target_state: target state to check for + * + * Return true if state matches, false if not + */ +bool snd_hdac_check_power_state(struct hdac_device *hdac, + hda_nid_t nid, unsigned int target_state) +{ + unsigned int state = codec_read(hdac, nid, 0, + AC_VERB_GET_POWER_STATE, 0); + + if (state & AC_PWRST_ERROR) + return true; + state = (state >> 4) & 0x0f; + return (state == target_state); +} +EXPORT_SYMBOL_GPL(snd_hdac_check_power_state); -- cgit v1.2.3 From a82d24f83de2c63199acead488259fcdf947e90e Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Thu, 15 Oct 2015 07:55:55 +0200 Subject: ALSA: emu10k1: added EMU10K1 version of DECLARE_BITMAP macro for UAPI MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation error: error: expected specifier-qualifier-list before ‘DECLARE_BITMAP’ DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ DECLARE_BITMAP macro is not meant for userspace headers and thus added here as private copy for emu10k.h. Fix was suggested by Arnd Bergmann in message <2168807.4Yxh5gl11Q@wuerfel> and Takashi Iwai in message on lkml. Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index ec1535bb6aed..5175e166987d 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -34,6 +34,14 @@ #define EMU10K1_FX8010_PCM_COUNT 8 +/* + * Following definition is copied from linux/types.h to support compiling + * this header file in userspace since they are not generally available for + * uapi headers. + */ +#define __EMU10K1_DECLARE_BITMAP(name,bits) \ + unsigned long name[(bits) / (sizeof(unsigned long) * 8)] + /* instruction set */ #define iMAC0 0x00 /* R = A + (X * Y >> 31) ; saturation */ #define iMAC1 0x01 /* R = A + (-X * Y >> 31) ; saturation */ @@ -300,7 +308,7 @@ struct snd_emu10k1_fx8010_control_old_gpr { struct snd_emu10k1_fx8010_code { char name[128]; - DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(gpr_valid, 0x200); /* bitmask of valid initializers */ __u32 __user *gpr_map; /* initializers */ unsigned int gpr_add_control_count; /* count of GPR controls to add/replace */ @@ -313,11 +321,11 @@ struct snd_emu10k1_fx8010_code { unsigned int gpr_list_control_total; /* total count of GPR controls */ struct snd_emu10k1_fx8010_control_gpr __user *gpr_list_controls; /* listed GPR controls */ - DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ + __EMU10K1_DECLARE_BITMAP(tram_valid, 0x100); /* bitmask of valid initializers */ __u32 __user *tram_data_map; /* data initializers */ __u32 __user *tram_addr_map; /* map initializers */ - DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ + __EMU10K1_DECLARE_BITMAP(code_valid, 1024); /* bitmask of valid instructions */ __u32 __user *code; /* one instruction - 64 bits */ }; -- cgit v1.2.3 From ffc287c89169705d9a01d48e05453ab0eda631e4 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Thu, 15 Oct 2015 07:56:06 +0200 Subject: ALSA: hdspm: use __u8, __u32 and __u64 from linux/types.h instead of stdint.h Kernel headers should use linux/types.h based definitions. Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 40 ++++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 22 deletions(-) (limited to 'include') diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 5737332d38f2..c4db6f5b306e 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,11 +20,7 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ -#ifdef __KERNEL__ #include -#else -#include -#endif /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 @@ -46,15 +42,15 @@ enum hdspm_speed { /* -------------------- IOCTL Peak/RMS Meters -------------------- */ struct hdspm_peak_rms { - uint32_t input_peaks[64]; - uint32_t playback_peaks[64]; - uint32_t output_peaks[64]; + __u32 input_peaks[64]; + __u32 playback_peaks[64]; + __u32 output_peaks[64]; - uint64_t input_rms[64]; - uint64_t playback_rms[64]; - uint64_t output_rms[64]; + __u64 input_rms[64]; + __u64 playback_rms[64]; + __u64 output_rms[64]; - uint8_t speed; /* enum {ss, ds, qs} */ + __u8 speed; /* enum {ss, ds, qs} */ int status2; }; @@ -155,21 +151,21 @@ enum hdspm_syncsource { }; struct hdspm_status { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ enum hdspm_syncsource autosync_source; - uint64_t card_clock; - uint32_t master_period; + __u64 card_clock; + __u32 master_period; union { struct { - uint8_t sync_wc; /* enum hdspm_sync */ - uint8_t sync_madi; /* enum hdspm_sync */ - uint8_t sync_tco; /* enum hdspm_sync */ - uint8_t sync_in; /* enum hdspm_sync */ - uint8_t madi_input; /* enum hdspm_madi_input */ - uint8_t channel_format; /* enum hdspm_madi_channel_format */ - uint8_t frame_format; /* enum hdspm_madi_frame_format */ + __u8 sync_wc; /* enum hdspm_sync */ + __u8 sync_madi; /* enum hdspm_sync */ + __u8 sync_tco; /* enum hdspm_sync */ + __u8 sync_in; /* enum hdspm_sync */ + __u8 madi_input; /* enum hdspm_madi_input */ + __u8 channel_format; /* enum hdspm_madi_channel_format */ + __u8 frame_format; /* enum hdspm_madi_frame_format */ } madi; } card_specific; }; @@ -184,7 +180,7 @@ struct hdspm_status { #define HDSPM_ADDON_TCO 1 struct hdspm_version { - uint8_t card_type; /* enum hdspm_io_type */ + __u8 card_type; /* enum hdspm_io_type */ char cardname[20]; unsigned int serial; unsigned short firmware_rev; -- cgit v1.2.3 From ded255be2276d365a91af2de7c7f8e2c233d4fa2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Oct 2015 17:59:43 +0200 Subject: ALSA: hda - consolidate chip rename functions A few multiple codec drivers do renaming the chip_name string but all these are open-coded and some of them have even no error check. Let's make common helpers to do it properly. Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 22 ++++++++++++++++++++++ sound/pci/hda/hda_bind.c | 35 +++++++++++++++++++++++------------ sound/pci/hda/hda_codec.h | 2 ++ sound/pci/hda/hda_sysfs.c | 3 +-- sound/pci/hda/patch_realtek.c | 12 +----------- sound/pci/hda/patch_via.c | 18 ++++-------------- 7 files changed, 54 insertions(+), 39 deletions(-) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 26e956f4b7c6..49df61c7afdc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -117,6 +117,7 @@ int snd_hdac_device_init(struct hdac_device *dev, struct hdac_bus *bus, void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b3b0ad289df1..4b06b26cee06 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -163,6 +163,28 @@ void snd_hdac_device_unregister(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_device_unregister); +/** + * snd_hdac_device_set_chip_name - set/update the codec name + * @codec: the HDAC device + * @name: name string to set + * + * Returns 0 if the name is set or updated, or a negative error code. + */ +int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name) +{ + char *newname; + + if (!name) + return 0; + newname = kstrdup(name, GFP_KERNEL); + if (!newname) + return -ENOMEM; + kfree(codec->chip_name); + codec->chip_name = newname; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); + /** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index d5ac25cc7fee..ef6b8f836a87 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -45,15 +45,31 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev) codec->patch_ops.unsol_event(codec, ev); } -/* reset the codec name from the preset */ -static int codec_refresh_name(struct hda_codec *codec, const char *name) +/** + * snd_hda_codec_set_name - set the codec name + * @codec: the HDA codec + * @name: name string to set + */ +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name) { - if (name) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); + int err; + + if (!name) + return 0; + err = snd_hdac_device_set_chip_name(&codec->core, name); + if (err < 0) + return err; + + /* update the mixer name */ + if (!*codec->card->mixername) { + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", + codec->core.vendor_name, codec->core.chip_name); } - return codec->core.chip_name ? 0 : -ENOMEM; + + return 0; } +EXPORT_SYMBOL_GPL(snd_hda_codec_set_name); static int hda_codec_driver_probe(struct device *dev) { @@ -64,7 +80,7 @@ static int hda_codec_driver_probe(struct device *dev) if (WARN_ON(!codec->preset)) return -EINVAL; - err = codec_refresh_name(codec, codec->preset->name); + err = snd_hda_codec_set_name(codec, codec->preset->name); if (err < 0) goto error; err = snd_hdac_regmap_init(&codec->core); @@ -251,11 +267,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) } } - /* audio codec should override the mixer name */ - if (codec->core.afg || !*codec->card->mixername) - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), "%s %s", - codec->core.vendor_name, codec->core.chip_name); return 0; error: diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 95991e463abb..b6d937784afa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -463,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus); void snd_hda_bus_reset(struct hda_bus *bus); void snd_hda_bus_reset_codecs(struct hda_bus *bus); +int snd_hda_codec_set_name(struct hda_codec *codec, const char *name); + /* * power management */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index a6e3d9b511ab..64e0d1d81ca5 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus, static void parse_chip_name_mode(char *buf, struct hda_bus *bus, struct hda_codec **codecp) { - kfree((*codecp)->core.chip_name); - (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL); + snd_hda_codec_set_name(*codecp, buf); } #define DEFINE_PARSE_ID_MODE(name) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 16b8dcba5c12..e1ffb0202ebc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = { }; -/* replace the codec chip_name with the given string */ -static int alc_codec_rename(struct hda_codec *codec, const char *name) -{ - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup(name, GFP_KERNEL); - if (!codec->core.chip_name) { - alc_free(codec); - return -ENOMEM; - } - return 0; -} +#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name) /* * Rename codecs appropriately from COEF value or subvendor id diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index da5366405eda..d714a57e9460 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec) override_mic_boost(codec, 0x1e, 0, 3, 40); /* correct names for VT1708BCE */ - if (get_codec_type(codec) == VT1708BCE) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (get_codec_type(codec) == VT1708BCE) + snd_hda_codec_set_name(codec, "VT1708BCE"); /* correct names for VT1705 */ - if (codec->core.vendor_id == 0x11064397) { - kfree(codec->core.chip_name); - codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->card->mixername, - sizeof(codec->card->mixername), - "%s %s", codec->core.vendor_name, codec->core.chip_name); - } + if (codec->core.vendor_id == 0x11064397) + snd_hda_codec_set_name(codec, "VT1705"); /* automatic parse from the BIOS config */ err = via_parse_auto_config(codec); -- cgit v1.2.3 From 90bbaf66ee7b946952f1e82a0069639dea5fd893 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Fri, 16 Oct 2015 17:57:46 +0800 Subject: ALSA: timer: add config item to export PCM timer disabling for expert PCM timer is not always used. For embedded device, we need an interface to disable it when it is not needed, to shrink the kernel size and memory footprint, here add CONFIG_SND_PCM_TIMER for it. When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected, about 25KB saving bonus we can get. Please be noted that when disabled, those stubs who using pcm timer (e.g. dmix, dsnoop & co) may work incorrectlly. Suggested-by: Takashi Iwai Signed-off-by: Jie Yang Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 8 +++++++- sound/core/Kconfig | 13 ++++++++++++- sound/core/Makefile | 3 ++- sound/core/pcm_lib.c | 2 ++ sound/core/pcm_native.c | 36 ++++++++++++++++-------------------- 5 files changed, 39 insertions(+), 23 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index a4fcc9456194..2882dddfc91c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1111,10 +1111,16 @@ static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substrea * Timer interface */ +#ifdef CONFIG_SND_PCM_TIMER void snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream); void snd_pcm_timer_init(struct snd_pcm_substream *substream); void snd_pcm_timer_done(struct snd_pcm_substream *substream); - +#else +static inline void +snd_pcm_timer_resolution_change(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_init(struct snd_pcm_substream *substream) {} +static inline void snd_pcm_timer_done(struct snd_pcm_substream *substream) {} +#endif /** * snd_pcm_gettime - Fill the timespec depending on the timestamp mode * @runtime: PCM runtime instance diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6c96feeaf01e..e3e949126a56 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -4,7 +4,7 @@ config SND_TIMER config SND_PCM tristate - select SND_TIMER + select SND_TIMER if SND_PCM_TIMER config SND_PCM_ELD bool @@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS support conversion of channels, formats and rates. It will behave like most of new OSS/Free drivers in 2.4/2.6 kernels. +config SND_PCM_TIMER + bool "PCM timer interface" if EXPERT + default y + help + If you disable this option, pcm timer will be inavailable, so + those stubs used pcm timer (e.g. dmix, dsnoop & co) may work + incorrectlly. + + For some embedded device, we may disable it to reduce memory + footprint, about 20KB on x86_64 platform. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND_SEQUENCER diff --git a/sound/core/Makefile b/sound/core/Makefile index 3354f91e003a..48ab4b8f8279 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o -snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \ +snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o +snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 7d45645f10ba..6dc4277937b8 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1883,8 +1883,10 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; +#ifdef CONFIG_SND_PCM_TIMER if (substream->timer_running) snd_timer_interrupt(substream->timer, 1); +#endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); if (runtime->transfer_ack_end) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 139887011ba2..a8b27cdc2844 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state) snd_pcm_stream_unlock_irq(substream); } +static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream, + int event) +{ +#ifdef CONFIG_SND_PCM_TIMER + if (substream->timer) + snd_timer_notify(substream->timer, event, + &substream->runtime->trigger_tstamp); +#endif +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -1043,9 +1053,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART); } static struct action_ops snd_pcm_action_start = { @@ -1093,9 +1101,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) if (runtime->status->state != state) { snd_pcm_trigger_tstamp(substream); runtime->status->state = state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP); } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); @@ -1209,18 +1215,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) snd_pcm_trigger_tstamp(substream); if (push) { runtime->status->state = SNDRV_PCM_STATE_PAUSED; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MPAUSE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; - if (substream->timer) - snd_timer_notify(substream->timer, - SNDRV_TIMER_EVENT_MCONTINUE, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE); } } @@ -1268,9 +1268,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) snd_pcm_trigger_tstamp(substream); runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND); wake_up(&runtime->sleep); wake_up(&runtime->tsleep); } @@ -1374,9 +1372,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->status->state = runtime->status->suspended_state; - if (substream->timer) - snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME, - &runtime->trigger_tstamp); + snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME); } static struct action_ops snd_pcm_action_resume = { -- cgit v1.2.3 From 93ed8560e98afc486df94f5a6238c1f0894b38b8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Oct 2015 21:35:53 +0200 Subject: ALSA: hda - Add api_version to hda_device_id struct For distinguishing the difference between HDA legacy and ext codec driver entries, we need to expose the value corresponding to type field. This patch adds a new field, api_version, to hda_device_id struct, so that this information is embedded in modalias string. Although the information is basically redundant (struct hdac_device already has type field), the helper that extracts from MODULE_DEVICE_TABLE() won't take it account except for the exported table entries themselves. So we need to put the same information in the table, too. Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + 1 file changed, 1 insertion(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 49df61c7afdc..b35bf59a1ecc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -33,6 +33,7 @@ extern struct bus_type snd_hda_bus_type; struct hda_device_id { __u32 vendor_id; __u32 rev_id; + __u8 api_version; const char *name; unsigned long driver_data; }; -- cgit v1.2.3 From da23ac1e40ce844d1a9553906bdacce160af76f6 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 29 Sep 2015 13:56:10 +0530 Subject: ALSA: hda - Add hduadio support to DEVTABLE For generating modalias entries automatically, move the definition of struct hda_device_id to linux/mod_devicetable.h and add the handling of this record in file2alias helper. The new modalias is represented with combination of vendor id, device id, and api version as "hdaudio:vNrNaN". This patch itself doesn't convert the existing modaliases. Since they were added manually, this patch won't give any regression by itself at this point. [Modified the modalias format to adapt the api_version field, and drop invalid ANY_ID definition by tiwai] Signed-off-by: Subhransu S. Prusty Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/linux/mod_devicetable.h | 8 ++++++++ include/sound/hdaudio.h | 12 +----------- scripts/mod/devicetable-offsets.c | 5 +++++ scripts/mod/file2alias.c | 17 +++++++++++++++++ sound/hda/hda_bus_type.c | 1 + 5 files changed, 32 insertions(+), 11 deletions(-) (limited to 'include') diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 688997a24aad..00825672d256 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -219,6 +219,14 @@ struct serio_device_id { __u8 proto; }; +struct hda_device_id { + __u32 vendor_id; + __u32 rev_id; + __u8 api_version; + const char *name; + unsigned long driver_data; +}; + /* * Struct used for matching a device */ diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b35bf59a1ecc..ddca48eb02e0 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -21,23 +21,13 @@ struct hdac_stream; struct hdac_device; struct hdac_driver; struct hdac_widget_tree; +struct hda_device_id; /* * exported bus type */ extern struct bus_type snd_hda_bus_type; -/* - * HDA device table - */ -struct hda_device_id { - __u32 vendor_id; - __u32 rev_id; - __u8 api_version; - const char *name; - unsigned long driver_data; -}; - /* * generic arrays */ diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index e70fcd12eeeb..e1a5110bd63b 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -196,5 +196,10 @@ int main(void) DEVID_FIELD(ulpi_device_id, vendor); DEVID_FIELD(ulpi_device_id, product); + DEVID(hda_device_id); + DEVID_FIELD(hda_device_id, vendor_id); + DEVID_FIELD(hda_device_id, rev_id); + DEVID_FIELD(hda_device_id, api_version); + return 0; } diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 5f2088209132..fc51d4bff3f8 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -1250,6 +1250,23 @@ static int do_ulpi_entry(const char *filename, void *symval, } ADD_TO_DEVTABLE("ulpi", ulpi_device_id, do_ulpi_entry); +/* Looks like: hdaudio:vNrNaN */ +static int do_hda_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD(symval, hda_device_id, vendor_id); + DEF_FIELD(symval, hda_device_id, rev_id); + DEF_FIELD(symval, hda_device_id, api_version); + + strcpy(alias, "hdaudio:"); + ADD(alias, "v", vendor_id != 0, vendor_id); + ADD(alias, "r", rev_id != 0, rev_id); + ADD(alias, "a", api_version != 0, api_version); + + add_wildcard(alias); + return 1; +} +ADD_TO_DEVTABLE("hdaudio", hda_device_id, do_hda_entry); + /* Does namelen bytes of name exactly match the symbol? */ static bool sym_is(const char *name, unsigned namelen, const char *symbol) { diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c index 89c2711baaaf..bcb1a79eec38 100644 --- a/sound/hda/hda_bus_type.c +++ b/sound/hda/hda_bus_type.c @@ -4,6 +4,7 @@ #include #include #include +#include #include #include -- cgit v1.2.3 From 4f9e0c38c5e991e2d050d13e28be74b93ab704c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Oct 2015 11:35:49 +0200 Subject: ALSA: hda - Add a common helper to give the codec modalias string This patch provide a new common helper function, snd_hdac_codec_modalias(), to give the codec modalias name string. This function will be used by multiple places in the later patches. Reviewed-by: Vinod Koul Tested-by: Subhransu S Prusty Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ddca48eb02e0..e2b712c90d3f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -109,6 +109,7 @@ void snd_hdac_device_exit(struct hdac_device *dev); int snd_hdac_device_register(struct hdac_device *codec); void snd_hdac_device_unregister(struct hdac_device *codec); int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name); +int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size); int snd_hdac_refresh_widgets(struct hdac_device *codec); int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 4b06b26cee06..bbdb25f5bbb9 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -185,6 +185,21 @@ int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name) } EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name); +/** + * snd_hdac_codec_modalias - give the module alias name + * @codec: HDAC device + * @buf: string buffer to store + * @size: string buffer size + * + * Returns the size of string, like snprintf(), or a negative error code. + */ +int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size) +{ + return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n", + codec->vendor_id, codec->revision_id, codec->type); +} +EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias); + /** * snd_hdac_make_cmd - compose a 32bit command word to be sent to the * HD-audio controller -- cgit v1.2.3 From b6e84c99b121fcba34166842987be96956148bb8 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Mon, 19 Oct 2015 16:58:46 +0530 Subject: ALSA: hdac: Add macro for hda ext devices entry With the new modalias infrastructure support added for hda, create a macro for ext devices similar to legacy to add the device entry. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- include/sound/hdaudio_ext.h | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'include') diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 94210dcdb6ea..a4cadd9c297a 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -40,6 +40,13 @@ void snd_hdac_ext_bus_device_remove(struct hdac_ext_bus *ebus); #define hbus_to_ebus(_bus) \ container_of(_bus, struct hdac_ext_bus, bus) +#define HDA_CODEC_REV_EXT_ENTRY(_vid, _rev, _name, drv_data) \ + { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \ + .api_version = HDA_DEV_ASOC, \ + .driver_data = (unsigned long)(drv_data) } +#define HDA_CODEC_EXT_ENTRY(_vid, _revid, _name, _drv_data) \ + HDA_CODEC_REV_EXT_ENTRY(_vid, _revid, _name, _drv_data) + int snd_hdac_ext_bus_parse_capabilities(struct hdac_ext_bus *sbus); void snd_hdac_ext_bus_ppcap_enable(struct hdac_ext_bus *chip, bool enable); void snd_hdac_ext_bus_ppcap_int_enable(struct hdac_ext_bus *chip, bool enable); -- cgit v1.2.3 From bc1043cdcd84cb441d80cfd79ae4325218f6f9ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 18 Oct 2015 15:39:16 +0200 Subject: ALSA: Add helper function to add single value constraint The recommended and most efficient way to constraint a configuration parameter to a single value is to set the minimum and maximum allowed values to the same value, i.e. calling snd_pcm_hw_constraint_minmax() with the same value for min and max. It is not necessarily obvious though that this is the approach that should be taken and some drivers have come up with other ways of solving this problem, e.g. installing a list constraint with a single item. List constraints are dynamic constraints though and hence less efficient than the static min-max constraint. This patch introduces a new helper function called snd_pcm_hw_constraint_single() which only takes a single value has the same effect as calling snd_pcm_hw_constraint_minmax() with the same values for min and max. But it is hopefully semantically more expressive, making it clear that this is the preferred way of setting a single value constraint. Signed-off-by: Lars-Peter Clausen Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee0a510..04fbcba9290b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1034,6 +1034,22 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, snd_pcm_hw_rule_func_t func, void *private, int dep, ...); +/** + * snd_pcm_hw_constraint_single() - Constrain parameter to a single value + * @runtime: PCM runtime instance + * @var: The hw_params variable to constrain + * @val: The value to constrain to + * + * Return: Positive if the value is changed, zero if it's not changed, or a + * negative error code. + */ +static inline int snd_pcm_hw_constraint_single( + struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, + unsigned int val) +{ + return snd_pcm_hw_constraint_minmax(runtime, var, val, val); +} + int snd_pcm_format_signed(snd_pcm_format_t format); int snd_pcm_format_unsigned(snd_pcm_format_t format); int snd_pcm_format_linear(snd_pcm_format_t format); -- cgit v1.2.3 From 16566e47098211e30b3d8a0bc6a3576871ada8e8 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 21 Oct 2015 09:46:05 +0800 Subject: ASoC: rt5640: Fill up the IN3's support Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/rt5640.txt | 5 ++++- include/sound/rt5640.h | 3 ++- sound/soc/codecs/rt5640.c | 22 +++++++++++++++++++++- 3 files changed, 27 insertions(+), 3 deletions(-) (limited to 'include') diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 5d062a567996..9e62f6eb348f 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -14,7 +14,8 @@ Optional properties: - realtek,in1-differential - realtek,in2-differential - Boolean. Indicate MIC1/2 input are differential, rather than single-ended. +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. @@ -27,6 +28,8 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * IN1N * IN2P * IN2N + * IN3P + * IN3N * HPOL * HPOR * LOUTL diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 59d26dd81e45..e3c84b92ff70 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -12,9 +12,10 @@ #define __LINUX_SND_RT5640_H struct rt5640_platform_data { - /* IN1 & IN2 can optionally be differential */ + /* IN1 & IN2 & IN3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool in3_diff; bool dmic_en; bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e1ceeb885f7d..f2beb1aa5763 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, @@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { RT5640_M_HP_L_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST2_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, RT5640_M_BST4_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, @@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { RT5640_M_HP_R_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST2_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, RT5640_M_BST4_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, @@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1N"), SND_SOC_DAPM_INPUT("IN2P"), SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_BST1_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, RT5640_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2, + RT5640_PWR_BST2_BIT, 0, NULL, 0), /* Input Volume */ SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, RT5640_PWR_IN_L_BIT, 0, NULL, 0), @@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, {"DMIC L1", NULL, "DMIC1"}, {"DMIC R1", NULL, "DMIC1"}, @@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"BST1", NULL, "IN1N"}, {"BST2", NULL, "IN2P"}, {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, {"INL VOL", NULL, "IN2P"}, {"INR VOL", NULL, "IN2N"}, {"RECMIXL", "HPOL Switch", "HPOL"}, {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, {"RECMIXL", "BST2 Switch", "BST2"}, {"RECMIXL", "BST1 Switch", "BST1"}, {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, {"RECMIXR", "HPOR Switch", "HPOR"}, {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, {"RECMIXR", "BST2 Switch", "BST2"}, {"RECMIXR", "BST1 Switch", "BST1"}, {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, @@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.in3_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- cgit v1.2.3 From 26d9ca3462df8f7e83fc372b23c8da5ed2b1c4f3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 18 Oct 2015 17:04:33 +0200 Subject: ASoC: Let snd_soc_limit_volume() take a snd_soc_card snd_soc_limit_volume() operates on a card and the CODEC that is passed in is only used to look up the card. Let it directly take the card instead. This makes it possible to use it when no snd_soc_codec is available. Signed-off-by: Lars-Peter Clausen Acked-by: Jarkko Nikula Tested-by: Jarkko Nikula Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/soc-ops.c | 8 ++++---- 3 files changed, 6 insertions(+), 6 deletions(-) (limited to 'include') diff --git a/include/sound/soc.h b/include/sound/soc.h index 884e728b09d9..71e0c0566b6e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -591,7 +591,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 3bebfb1d3a6f..99538900a253 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -297,7 +297,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) dev_err(card->dev, "Failed to add TPA6130A2 controls\n"); return err; } - snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + snd_soc_limit_volume(card, "TPA6130A2 Headphone Playback Volume", 42); err = omap_mcbsp_st_add_controls(rtd, 2); if (err < 0) { diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 100d92b5b77e..20f702add3f8 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -560,16 +560,16 @@ EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); /** * snd_soc_limit_volume - Set new limit to an existing volume control. * - * @codec: where to look for the control + * @card: where to look for the control * @name: Name of the control * @max: new maximum limit * * Return 0 for success, else error. */ -int snd_soc_limit_volume(struct snd_soc_codec *codec, +int snd_soc_limit_volume(struct snd_soc_card *card, const char *name, int max) { - struct snd_card *card = codec->component.card->snd_card; + struct snd_card *snd_card = card->snd_card; struct snd_kcontrol *kctl; struct soc_mixer_control *mc; int found = 0; @@ -579,7 +579,7 @@ int snd_soc_limit_volume(struct snd_soc_codec *codec, if (unlikely(!name || max <= 0)) return -EINVAL; - list_for_each_entry(kctl, &card->controls, list) { + list_for_each_entry(kctl, &snd_card->controls, list) { if (!strncmp(kctl->id.name, name, sizeof(kctl->id.name))) { found = 1; break; -- cgit v1.2.3 From 6f0c42269f000b1e346c84d9a589f17aa94c96d8 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 13 Oct 2015 23:41:00 +0800 Subject: ASoC: compress: add config item for soc-compress to make it compiled only when needed We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- include/sound/soc.h | 4 +++- sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/soc-compress.c | 12 ++++++++++-- sound/soc/soc-core.c | 4 ++-- 8 files changed, 24 insertions(+), 9 deletions(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1384c7..238200ffba5b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -214,7 +214,7 @@ struct snd_soc_dai_driver { int (*suspend)(struct snd_soc_dai *dai); int (*resume)(struct snd_soc_dai *dai); /* compress dai */ - bool compress_dai; + int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num); /* DAI is also used for the control bus */ bool bus_control; diff --git a/include/sound/soc.h b/include/sound/soc.h index 71e0c0566b6e..a7bc82b08cd4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,9 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 225bfda414e9..3e4f44434600 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + bool + select SND_COMPRESS_OFFLOAD + config SND_SOC_TOPOLOGY bool diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 134aca150a50..280ac8257d3e 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,6 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 05fde5e6e257..221e3bd73adb 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e50116152..4c734f5aeb8b 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -529,7 +529,7 @@ static struct snd_soc_dai_driver sst_platform_dai[] = { }, { .name = "compress-cpu-dai", - .compress_dai = 1, + .compress_new = snd_soc_new_compress, .ops = &sst_compr_dai_ops, .playback = { .stream_name = "Compress Playback", diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 025c38fbe3c0..12a9820feac1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -612,8 +612,15 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .get_codec_caps = soc_compr_get_codec_caps }; -/* create a new compress */ -int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +/** + * snd_soc_new_compress - create a new compress. + * + * @rtd: The runtime for which we will create compress + * @num: the device index number (zero based - shared with normal PCMs) + * + * Return: 0 for success, else error. + */ +int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; @@ -703,3 +710,4 @@ compr_err: kfree(compr); return ret; } +EXPORT_SYMBOL_GPL(snd_soc_new_compress); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15236c3..6e543e18b4fa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1370,9 +1370,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) soc_dpcm_debugfs_add(rtd); #endif - if (cpu_dai->driver->compress_dai) { + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); -- cgit v1.2.3 From 93e39a11520c000c9086215460bf27b35b09c724 Mon Sep 17 00:00:00 2001 From: Mythri P K Date: Tue, 20 Oct 2015 22:30:08 +0530 Subject: ASoC: dapm: Add snd_soc_dapm_kcontrol_widget() Given a kcontrol, we may want to access the parent widget and it's associated data. So export function to return it. Signed-off-by: Mythri P K Signed-off-by: Jeeja KP Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 3 +++ sound/soc/soc-dapm.c | 12 ++++++++++++ 2 files changed, 15 insertions(+) (limited to 'include') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5abba037d245..7855cfe46b69 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -451,6 +451,9 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); +struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget( + struct snd_kcontrol *kcontrol); + int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a28d6a10bad0..38281c2325ff 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -508,6 +508,18 @@ static bool dapm_kcontrol_set_value(const struct snd_kcontrol *kcontrol, return true; } +/** + * snd_soc_dapm_kcontrol_widget() - Returns the widget associated to a + * kcontrol + * @kcontrol: The kcontrol + */ +struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget( + struct snd_kcontrol *kcontrol) +{ + return dapm_kcontrol_get_wlist(kcontrol)->widgets[0]; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_kcontrol_widget); + /** * snd_soc_dapm_kcontrol_dapm() - Returns the dapm context associated to a * kcontrol -- cgit v1.2.3 From 53e597b1d194910bef53ed0632da329fef497904 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 22 Oct 2015 13:11:56 +0200 Subject: ALSA: Remove transfer_ack_{begin,end} callbacks from struct snd_pcm_runtime While there is nothing wrong with the transfer_ack_begin and transfer_ack_end callbacks per-se, the last documented user was part of the alsa-driver 0.5.12a package, which was released 14 years ago and even predates the upstream integration of the ALSA core and has subsequently been superseded by newer alsa-driver releases. This seems to indicate that there is no need for having these callbacks and they are just cruft that can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 19 ++----------------- include/sound/pcm.h | 4 ---- sound/core/pcm_lib.c | 5 ----- 3 files changed, 2 insertions(+), 26 deletions(-) (limited to 'include') diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 84ef6a90131c..a27ab9f53fb6 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -2181,10 +2181,6 @@ struct _snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ @@ -2209,9 +2205,8 @@ struct _snd_pcm_runtime { For the operators (callbacks) of each sound driver, most of these records are supposed to be read-only. Only the PCM middle-layer changes / updates them. The exceptions are - the hardware description (hw), interrupt callbacks - (transfer_ack_xxx), DMA buffer information, and the private - data. Besides, if you use the standard buffer allocation + the hardware description (hw) DMA buffer information and the + private data. Besides, if you use the standard buffer allocation method via snd_pcm_lib_malloc_pages(), you don't need to set the DMA buffer information by yourself. @@ -2538,16 +2533,6 @@ struct _snd_pcm_runtime { -
- Interrupt Callbacks - - The field transfer_ack_begin and - transfer_ack_end are called at - the beginning and at the end of - snd_pcm_period_elapsed(), respectively. - -
-
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2882dddfc91c..3e0ffd21901f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,10 +402,6 @@ struct snd_pcm_runtime { struct snd_pcm_hardware hw; struct snd_pcm_hw_constraints hw_constraints; - /* -- interrupt callbacks -- */ - void (*transfer_ack_begin)(struct snd_pcm_substream *substream); - void (*transfer_ack_end)(struct snd_pcm_substream *substream); - /* -- timer -- */ unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6dc4277937b8..05a3ca93c647 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1875,9 +1875,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) return; runtime = substream->runtime; - if (runtime->transfer_ack_begin) - runtime->transfer_ack_begin(substream); - snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || snd_pcm_update_hw_ptr0(substream, 1) < 0) @@ -1889,8 +1886,6 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) #endif _end: snd_pcm_stream_unlock_irqrestore(substream, flags); - if (runtime->transfer_ack_end) - runtime->transfer_ack_end(substream); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); } -- cgit v1.2.3 From 1d387a3fd86f2acf70803262c5a5a5a89df0e097 Mon Sep 17 00:00:00 2001 From: Anatol Pomozov Date: Thu, 8 Oct 2015 09:37:51 -0700 Subject: ASoC: Document DAI signal polarity Currently there is no clear definition of what FSYNC polarity is. Different drivers use its own definition of what is "normal" and what is "inverted" fsync. This leads to compatibility problems between drivers. For example TegraX1 driver assumes that DSP-A format with frames starting at rising FSYNC edge has "inverted" polarity, while RT5677 assumes it is "normal" polarity. Explicitly specify meaning of BCLK/FSYNC polarity to avoid future compatibility problems. Signed-off-by: Anatol Pomozov Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'include') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1384c7..91e2e61d584c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -48,10 +48,25 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ /* - * DAI hardware signal inversions. + * DAI hardware signal polarity. * * Specifies whether the DAI can also support inverted clocks for the specified * format. + * + * BCLK: + * - "normal" polarity means signal is available at rising edge of BCLK + * - "inverted" polarity means signal is available at falling edge of BCLK + * + * FSYNC "normal" polarity depends on the frame format: + * - I2S: frame consists of left then right channel data. Left channel starts + * with falling FSYNC edge, right channel starts with rising FSYNC edge. + * - Left/Right Justified: frame consists of left then right channel data. + * Left channel starts with rising FSYNC edge, right channel starts with + * falling FSYNC edge. + * - DSP A/B: Frame starts with rising FSYNC edge. + * - AC97: Frame starts with rising FSYNC edge. + * + * "Negative" FSYNC polarity is the one opposite of "normal" polarity. */ #define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ #define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ -- cgit v1.2.3 From e5e113cf0d19392f26c6b63e63ad4680ee4ec5da Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 28 Oct 2015 11:37:53 +0100 Subject: ALSA: Constify ratden/ratnum constraints The ALSA core does not modify the constraints provided by a driver. Most constraint helper functions already take a const pointer to the constraint description, the exception at the moment being the ratden and ratnum constraints. Make those const as well, this allows a driver to declare them as const. Signed-off-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 10 +++++----- sound/core/pcm_lib.c | 17 +++++++++-------- 2 files changed, 14 insertions(+), 13 deletions(-) (limited to 'include') diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e6ad74fd8a2..b0be09279943 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -265,12 +265,12 @@ struct snd_ratden { struct snd_pcm_hw_constraint_ratnums { int nrats; - struct snd_ratnum *rats; + const struct snd_ratnum *rats; }; struct snd_pcm_hw_constraint_ratdens { int nrats; - struct snd_ratden *rats; + const struct snd_ratden *rats; }; struct snd_pcm_hw_constraint_list { @@ -970,7 +970,7 @@ int snd_interval_list(struct snd_interval *i, unsigned int count, int snd_interval_ranges(struct snd_interval *i, unsigned int count, const struct snd_interval *list, unsigned int mask); int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp); void _snd_pcm_hw_params_any(struct snd_pcm_hw_params *params); @@ -1000,11 +1000,11 @@ int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r); + const struct snd_pcm_hw_constraint_ratnums *r); int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r); + const struct snd_pcm_hw_constraint_ratdens *r); int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 05a3ca93c647..6b5a811e01a5 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -801,7 +801,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, * negative error code. */ int snd_interval_ratnum(struct snd_interval *i, - unsigned int rats_count, struct snd_ratnum *rats, + unsigned int rats_count, const struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_den; @@ -920,7 +920,8 @@ EXPORT_SYMBOL(snd_interval_ratnum); * negative error code. */ static int snd_interval_ratden(struct snd_interval *i, - unsigned int rats_count, struct snd_ratden *rats, + unsigned int rats_count, + const struct snd_ratden *rats, unsigned int *nump, unsigned int *denp) { unsigned int best_num, best_diff, best_den; @@ -1339,7 +1340,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges); static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratnums *r = rule->private; + const struct snd_pcm_hw_constraint_ratnums *r = rule->private; unsigned int num = 0, den = 0; int err; err = snd_interval_ratnum(hw_param_interval(params, rule->var), @@ -1363,10 +1364,10 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratnums *r) + const struct snd_pcm_hw_constraint_ratnums *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratnums, r, + snd_pcm_hw_rule_ratnums, (void *)r, var, -1); } @@ -1375,7 +1376,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums); static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { - struct snd_pcm_hw_constraint_ratdens *r = rule->private; + const struct snd_pcm_hw_constraint_ratdens *r = rule->private; unsigned int num = 0, den = 0; int err = snd_interval_ratden(hw_param_interval(params, rule->var), r->nrats, r->rats, &num, &den); @@ -1398,10 +1399,10 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, - struct snd_pcm_hw_constraint_ratdens *r) + const struct snd_pcm_hw_constraint_ratdens *r) { return snd_pcm_hw_rule_add(runtime, cond, var, - snd_pcm_hw_rule_ratdens, r, + snd_pcm_hw_rule_ratdens, (void *)r, var, -1); } -- cgit v1.2.3