From d745f5e7b8b2961f68b0b9093a0f914a8a83c2ae Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 21 Mar 2016 14:41:58 +0100
Subject: ALSA: hda - Add the pin / port mapping on Intel ILK and VLV

Intel IronLake and ValleyView platforms have different HDMI widget pin
and digital port mapping from other newer ones.  The recent ones
(HSW+) have NID 0x05 to 0x07 for port B to port D, while these chips
have NID 0x04 to 0x06.

For adapting this mapping, pass the codec object instead of the bus
object to snd_hdac_sync_audio_rate() and snd_hdac_acomp_get_eld() so
that they can check the codec ID and calculate the mapping properly.

The changes in the HDMI codec driver side will follow in the later
patch.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hda_i915.h | 10 +++++-----
 1 file changed, 5 insertions(+), 5 deletions(-)

(limited to 'include')

diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h
index fa341fcb5829..eed87a7559b7 100644
--- a/include/sound/hda_i915.h
+++ b/include/sound/hda_i915.h
@@ -10,8 +10,8 @@
 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
 int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 int snd_hdac_get_display_clk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
 int snd_hdac_i915_exit(struct hdac_bus *bus);
@@ -29,12 +29,12 @@ static inline int snd_hdac_get_display_clk(struct hdac_bus *bus)
 {
 	return 0;
 }
-static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid,
-					   int rate)
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+					   hda_nid_t nid, int rate)
 {
 	return 0;
 }
-static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 					 bool *audio_enabled, char *buffer,
 					 int max_bytes)
 {
-- 
cgit v1.2.3


From 44fde3b89ba1e154b3cec7d711703fff53852983 Mon Sep 17 00:00:00 2001
From: "Subhransu S. Prusty" <subhransu.s.prusty@intel.com>
Date: Mon, 4 Apr 2016 19:23:54 +0530
Subject: ALSA: hda - Update chmap tlv to report sink's capability

The existing TLV callback implementation copies all of the
cea_channel_speaker_allocation map table to the TLV container
irrespective of what is reported by sink. This is of little use
to the userspace application.

With this patch, it parses the spk_alloc block as queried from
the ELD, and copies only the corresponding mapping channel
allocation entries from the cea channel speaker allocation table.
Thus the user can parse the TLV container to identify sink's
capability and set the channel map accordingly.

It shouldn't impact the behavior in AMD chipset, as this makes
use of already parsed spk alloc block to calculate the channel
map.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hda_chmap.h  |  2 ++
 sound/hda/hdmi_chmap.c     | 44 ++++++++++++++++++++++++++++++++++++++++----
 sound/pci/hda/patch_hdmi.c | 13 +++++++++++++
 3 files changed, 55 insertions(+), 4 deletions(-)

(limited to 'include')

diff --git a/include/sound/hda_chmap.h b/include/sound/hda_chmap.h
index e20d219a0304..babd445c7505 100644
--- a/include/sound/hda_chmap.h
+++ b/include/sound/hda_chmap.h
@@ -36,6 +36,8 @@ struct hdac_chmap_ops {
 	int (*chmap_validate)(struct hdac_chmap *hchmap, int ca,
 			int channels, unsigned char *chmap);
 
+	int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx);
+
 	void (*get_chmap)(struct hdac_device *hdac, int pcm_idx,
 					unsigned char *chmap);
 	void (*set_chmap)(struct hdac_device *hdac, int pcm_idx,
diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c
index d7ec86263828..c6c75e7e0981 100644
--- a/sound/hda/hdmi_chmap.c
+++ b/sound/hda/hdmi_chmap.c
@@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap,
 	WARN_ON(count != channels);
 }
 
+static int spk_mask_from_spk_alloc(int spk_alloc)
+{
+	int i;
+	int spk_mask = eld_speaker_allocation_bits[0];
+
+	for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+		if (spk_alloc & (1 << i))
+			spk_mask |= eld_speaker_allocation_bits[i];
+	}
+
+	return spk_mask;
+}
+
 static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 			      unsigned int size, unsigned int __user *tlv)
 {
 	struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
 	struct hdac_chmap *chmap = info->private_data;
+	int pcm_idx = kcontrol->private_value;
 	unsigned int __user *dst;
 	int chs, count = 0;
+	unsigned long max_chs;
+	int type;
+	int spk_alloc, spk_mask;
 
 	if (size < 8)
 		return -ENOMEM;
@@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 		return -EFAULT;
 	size -= 8;
 	dst = tlv + 2;
-	for (chs = 2; chs <= chmap->channels_max; chs++) {
+
+	spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx);
+	spk_mask = spk_mask_from_spk_alloc(spk_alloc);
+
+	max_chs = hweight_long(spk_mask);
+
+	for (chs = 2; chs <= max_chs; chs++) {
 		int i;
 		struct hdac_cea_channel_speaker_allocation *cap;
 
 		cap = channel_allocations;
 		for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) {
 			int chs_bytes = chs * 4;
-			int type = chmap->ops.chmap_cea_alloc_validate_get_type(
-								chmap, cap, chs);
 			unsigned int tlv_chmap[8];
 
-			if (type < 0)
+			if (cap->channels != chs)
+				continue;
+
+			if (!(cap->spk_mask == (spk_mask & cap->spk_mask)))
 				continue;
+
+			type = chmap->ops.chmap_cea_alloc_validate_get_type(
+							chmap, cap, chs);
+			if (type < 0)
+				return -ENODEV;
 			if (size < 8)
 				return -ENOMEM;
+
 			if (put_user(type, dst) ||
 			    put_user(chs_bytes, dst + 1))
 				return -EFAULT;
+
 			dst += 2;
 			size -= 8;
 			count += 8;
+
 			if (size < chs_bytes)
 				return -ENOMEM;
+
 			size -= chs_bytes;
 			count += chs_bytes;
 			chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap,
 						tlv_chmap, chs);
+
 			if (copy_to_user(dst, tlv_chmap, chs_bytes))
 				return -EFAULT;
 			dst += chs;
 		}
 	}
+
 	if (put_user(count, tlv + 1))
 		return -EFAULT;
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 4833c7bdd1e8..9452384d2000 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1837,6 +1837,18 @@ static const struct hda_pcm_ops generic_ops = {
 	.cleanup = generic_hdmi_playback_pcm_cleanup,
 };
 
+static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx)
+{
+	struct hda_codec *codec = container_of(hdac, struct hda_codec, core);
+	struct hdmi_spec *spec = codec->spec;
+	struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx);
+
+	if (!per_pin)
+		return 0;
+
+	return per_pin->sink_eld.info.spk_alloc;
+}
+
 static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx,
 					unsigned char *chmap)
 {
@@ -2165,6 +2177,7 @@ static int alloc_generic_hdmi(struct hda_codec *codec)
 	spec->chmap.ops.get_chmap = hdmi_get_chmap;
 	spec->chmap.ops.set_chmap = hdmi_set_chmap;
 	spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
+	spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc,
 
 	codec->spec = spec;
 	hdmi_array_init(spec, 4);
-- 
cgit v1.2.3


From 4a4436573a6669516f73bac25016683d396ed4c4 Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Thu, 31 Mar 2016 16:35:58 +0300
Subject: ALSA: pcm: add IEC958 channel status helper for hw_params

Add IEC958 channel status helper that gets the audio properties from
snd_pcm_hw_params instead of snd_pcm_runtime. This is needed to
produce the channel status bits already in audio stream configuration
phase.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/pcm_iec958.h |  2 ++
 sound/core/pcm_iec958.c    | 64 ++++++++++++++++++++++++++++++++++------------
 2 files changed, 49 insertions(+), 17 deletions(-)

(limited to 'include')

diff --git a/include/sound/pcm_iec958.h b/include/sound/pcm_iec958.h
index 0eed397aca8e..36f023acb201 100644
--- a/include/sound/pcm_iec958.h
+++ b/include/sound/pcm_iec958.h
@@ -6,4 +6,6 @@
 int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 	size_t len);
 
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+					     u8 *cs, size_t len);
 #endif
diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c
index 36b2d7aca1bd..e016871a978f 100644
--- a/sound/core/pcm_iec958.c
+++ b/sound/core/pcm_iec958.c
@@ -9,30 +9,18 @@
 #include <linux/types.h>
 #include <sound/asoundef.h>
 #include <sound/pcm.h>
+#include <sound/pcm_params.h>
 #include <sound/pcm_iec958.h>
 
-/**
- * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
- * @runtime: pcm runtime structure with ->rate filled in
- * @cs: channel status buffer, at least four bytes
- * @len: length of channel status buffer
- *
- * Create the consumer format channel status data in @cs of maximum size
- * @len corresponding to the parameters of the PCM runtime @runtime.
- *
- * Drivers may wish to tweak the contents of the buffer after creation.
- *
- * Returns: length of buffer, or negative error code if something failed.
- */
-int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
-	size_t len)
+static int create_iec958_consumer(uint rate, uint sample_width,
+				  u8 *cs, size_t len)
 {
 	unsigned int fs, ws;
 
 	if (len < 4)
 		return -EINVAL;
 
-	switch (runtime->rate) {
+	switch (rate) {
 	case 32000:
 		fs = IEC958_AES3_CON_FS_32000;
 		break;
@@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 	}
 
 	if (len > 4) {
-		switch (snd_pcm_format_width(runtime->format)) {
+		switch (sample_width) {
 		case 16:
 			ws = IEC958_AES4_CON_WORDLEN_20_16;
 			break;
@@ -92,4 +80,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 
 	return len;
 }
+
+/**
+ * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status
+ * @runtime: pcm runtime structure with ->rate filled in
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
+	size_t len)
+{
+	return create_iec958_consumer(runtime->rate,
+				      snd_pcm_format_width(runtime->format),
+				      cs, len);
+}
 EXPORT_SYMBOL(snd_pcm_create_iec958_consumer);
+
+/**
+ * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status
+ * @hw_params: the hw_params instance for extracting rate and sample format
+ * @cs: channel status buffer, at least four bytes
+ * @len: length of channel status buffer
+ *
+ * Create the consumer format channel status data in @cs of maximum size
+ * @len corresponding to the parameters of the PCM runtime @runtime.
+ *
+ * Drivers may wish to tweak the contents of the buffer after creation.
+ *
+ * Returns: length of buffer, or negative error code if something failed.
+ */
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+					     u8 *cs, size_t len)
+{
+	return create_iec958_consumer(params_rate(params), params_width(params),
+				      cs, len);
+}
+EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params);
-- 
cgit v1.2.3


From 09184118a8abae030539469848d475adcc0e5839 Mon Sep 17 00:00:00 2001
From: Jyri Sarha <jsarha@ti.com>
Date: Thu, 31 Mar 2016 16:36:00 +0300
Subject: ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders

The hdmi-codec is a platform device driver to be registered from
drivers of external HDMI encoders with I2S and/or spdif interface. The
driver in turn registers an ASoC codec for the HDMI encoder's audio
functionality.

The structures and definitions in the API header are mostly redundant
copies of similar structures in ASoC headers. This is on purpose to
avoid direct dependencies to ASoC structures in video side driver.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Acked-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/hdmi-codec.h    | 100 +++++++++++
 sound/soc/codecs/Kconfig      |   6 +
 sound/soc/codecs/Makefile     |   2 +
 sound/soc/codecs/hdmi-codec.c | 396 ++++++++++++++++++++++++++++++++++++++++++
 4 files changed, 504 insertions(+)
 create mode 100644 include/sound/hdmi-codec.h
 create mode 100644 sound/soc/codecs/hdmi-codec.c

(limited to 'include')

diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
new file mode 100644
index 000000000000..fc3a481ad91e
--- /dev/null
+++ b/include/sound/hdmi-codec.h
@@ -0,0 +1,100 @@
+/*
+ * hdmi-codec.h - HDMI Codec driver API
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ */
+
+#ifndef __HDMI_CODEC_H__
+#define __HDMI_CODEC_H__
+
+#include <linux/hdmi.h>
+#include <drm/drm_edid.h>
+#include <sound/asoundef.h>
+#include <uapi/sound/asound.h>
+
+/*
+ * Protocol between ASoC cpu-dai and HDMI-encoder
+ */
+struct hdmi_codec_daifmt {
+	enum {
+		HDMI_I2S,
+		HDMI_RIGHT_J,
+		HDMI_LEFT_J,
+		HDMI_DSP_A,
+		HDMI_DSP_B,
+		HDMI_AC97,
+		HDMI_SPDIF,
+	} fmt;
+	int bit_clk_inv:1;
+	int frame_clk_inv:1;
+	int bit_clk_master:1;
+	int frame_clk_master:1;
+};
+
+/*
+ * HDMI audio parameters
+ */
+struct hdmi_codec_params {
+	struct hdmi_audio_infoframe cea;
+	struct snd_aes_iec958 iec;
+	int sample_rate;
+	int sample_width;
+	int channels;
+};
+
+struct hdmi_codec_ops {
+	/*
+	 * Called when ASoC starts an audio stream setup.
+	 * Optional
+	 */
+	int (*audio_startup)(struct device *dev);
+
+	/*
+	 * Configures HDMI-encoder for audio stream.
+	 * Mandatory
+	 */
+	int (*hw_params)(struct device *dev,
+			 struct hdmi_codec_daifmt *fmt,
+			 struct hdmi_codec_params *hparms);
+
+	/*
+	 * Shuts down the audio stream.
+	 * Mandatory
+	 */
+	void (*audio_shutdown)(struct device *dev);
+
+	/*
+	 * Mute/unmute HDMI audio stream.
+	 * Optional
+	 */
+	int (*digital_mute)(struct device *dev, bool enable);
+
+	/*
+	 * Provides EDID-Like-Data from connected HDMI device.
+	 * Optional
+	 */
+	int (*get_eld)(struct device *dev, uint8_t *buf, size_t len);
+};
+
+/* HDMI codec initalization data */
+struct hdmi_codec_pdata {
+	const struct hdmi_codec_ops *ops;
+	uint i2s:1;
+	uint spdif:1;
+	int max_i2s_channels;
+};
+
+#define HDMI_CODEC_DRV_NAME "hdmi-audio-codec"
+
+#endif /* __HDMI_CODEC_H__ */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 649e92a252ae..06d0e0593ec3 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -88,6 +88,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
 	select SND_SOC_NAU8825 if I2C
+	select SND_SOC_HDMI_CODEC
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM179X_I2C if I2C
 	select SND_SOC_PCM179X_SPI if SPI_MASTER
@@ -477,6 +478,11 @@ config SND_SOC_BT_SCO
 config SND_SOC_DMIC
 	tristate
 
+config SND_SOC_HDMI_CODEC
+       tristate
+       select SND_PCM_ELD
+       select SND_PCM_IEC958
+
 config SND_SOC_ES8328
 	tristate "Everest Semi ES8328 CODEC"
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 185a712a7fe7..d7185dda58b8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -81,6 +81,7 @@ snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
 snd-soc-nau8825-objs := nau8825.o
+snd-soc-hdmi-codec-objs := hdmi-codec.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm179x-codec-objs := pcm179x.o
 snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o
@@ -290,6 +291,7 @@ obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
 obj-$(CONFIG_SND_SOC_NAU8825)   += snd-soc-nau8825.o
+obj-$(CONFIG_SND_SOC_HDMI_CODEC)	+= snd-soc-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM179X)	+= snd-soc-pcm179x-codec.o
 obj-$(CONFIG_SND_SOC_PCM179X_I2C)	+= snd-soc-pcm179x-i2c.o
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
new file mode 100644
index 000000000000..b46b8edb9319
--- /dev/null
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -0,0 +1,396 @@
+/*
+ * ALSA SoC codec for HDMI encoder drivers
+ * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.	 See the GNU
+ * General Public License for more details.
+ */
+#include <linux/module.h>
+#include <linux/string.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pcm_drm_eld.h>
+#include <sound/hdmi-codec.h>
+#include <sound/pcm_iec958.h>
+
+#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */
+
+struct hdmi_codec_priv {
+	struct hdmi_codec_pdata hcd;
+	struct snd_soc_dai_driver *daidrv;
+	struct hdmi_codec_daifmt daifmt[2];
+	struct mutex current_stream_lock;
+	struct snd_pcm_substream *current_stream;
+	struct snd_pcm_hw_constraint_list ratec;
+	uint8_t eld[MAX_ELD_BYTES];
+};
+
+static const struct snd_soc_dapm_widget hdmi_widgets[] = {
+	SND_SOC_DAPM_OUTPUT("TX"),
+};
+
+static const struct snd_soc_dapm_route hdmi_routes[] = {
+	{ "TX", NULL, "Playback" },
+};
+
+enum {
+	DAI_ID_I2S = 0,
+	DAI_ID_SPDIF,
+};
+
+static int hdmi_codec_new_stream(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	mutex_lock(&hcp->current_stream_lock);
+	if (!hcp->current_stream) {
+		hcp->current_stream = substream;
+	} else if (hcp->current_stream != substream) {
+		dev_err(dai->dev, "Only one simultaneous stream supported!\n");
+		ret = -EINVAL;
+	}
+	mutex_unlock(&hcp->current_stream_lock);
+
+	return ret;
+}
+
+static int hdmi_codec_startup(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	ret = hdmi_codec_new_stream(substream, dai);
+	if (ret)
+		return ret;
+
+	if (hcp->hcd.ops->audio_startup) {
+		ret = hcp->hcd.ops->audio_startup(dai->dev->parent);
+		if (ret) {
+			mutex_lock(&hcp->current_stream_lock);
+			hcp->current_stream = NULL;
+			mutex_unlock(&hcp->current_stream_lock);
+			return ret;
+		}
+	}
+
+	if (hcp->hcd.ops->get_eld) {
+		ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld,
+					    sizeof(hcp->eld));
+
+		if (!ret) {
+			ret = snd_pcm_hw_constraint_eld(substream->runtime,
+							hcp->eld);
+			if (ret)
+				return ret;
+		}
+	}
+	return 0;
+}
+
+static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	WARN_ON(hcp->current_stream != substream);
+
+	hcp->hcd.ops->audio_shutdown(dai->dev->parent);
+
+	mutex_lock(&hcp->current_stream_lock);
+	hcp->current_stream = NULL;
+	mutex_unlock(&hcp->current_stream_lock);
+}
+
+static int hdmi_codec_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	struct hdmi_codec_params hp = {
+		.iec = {
+			.status = { 0 },
+			.subcode = { 0 },
+			.pad = 0,
+			.dig_subframe = { 0 },
+		}
+	};
+	int ret;
+
+	dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__,
+		params_width(params), params_rate(params),
+		params_channels(params));
+
+	if (params_width(params) > 24)
+		params->msbits = 24;
+
+	ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status,
+						       sizeof(hp.iec.status));
+	if (ret < 0) {
+		dev_err(dai->dev, "Creating IEC958 channel status failed %d\n",
+			ret);
+		return ret;
+	}
+
+	ret = hdmi_codec_new_stream(substream, dai);
+	if (ret)
+		return ret;
+
+	hdmi_audio_infoframe_init(&hp.cea);
+	hp.cea.channels = params_channels(params);
+	hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM;
+	hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM;
+	hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM;
+
+	hp.sample_width = params_width(params);
+	hp.sample_rate = params_rate(params);
+	hp.channels = params_channels(params);
+
+	return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id],
+				       &hp);
+}
+
+static int hdmi_codec_set_fmt(struct snd_soc_dai *dai,
+			      unsigned int fmt)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	struct hdmi_codec_daifmt cf = { 0 };
+	int ret = 0;
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	if (dai->id == DAI_ID_SPDIF) {
+		cf.fmt = HDMI_SPDIF;
+	} else {
+		switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+		case SND_SOC_DAIFMT_CBM_CFM:
+			cf.bit_clk_master = 1;
+			cf.frame_clk_master = 1;
+			break;
+		case SND_SOC_DAIFMT_CBS_CFM:
+			cf.frame_clk_master = 1;
+			break;
+		case SND_SOC_DAIFMT_CBM_CFS:
+			cf.bit_clk_master = 1;
+			break;
+		case SND_SOC_DAIFMT_CBS_CFS:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			cf.frame_clk_inv = 1;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			cf.bit_clk_inv = 1;
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			cf.frame_clk_inv = 1;
+			cf.bit_clk_inv = 1;
+			break;
+		}
+
+		switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+		case SND_SOC_DAIFMT_I2S:
+			cf.fmt = HDMI_I2S;
+			break;
+		case SND_SOC_DAIFMT_DSP_A:
+			cf.fmt = HDMI_DSP_A;
+			break;
+		case SND_SOC_DAIFMT_DSP_B:
+			cf.fmt = HDMI_DSP_B;
+			break;
+		case SND_SOC_DAIFMT_RIGHT_J:
+			cf.fmt = HDMI_RIGHT_J;
+			break;
+		case SND_SOC_DAIFMT_LEFT_J:
+			cf.fmt = HDMI_LEFT_J;
+			break;
+		case SND_SOC_DAIFMT_AC97:
+			cf.fmt = HDMI_AC97;
+			break;
+		default:
+			dev_err(dai->dev, "Invalid DAI interface format\n");
+			return -EINVAL;
+		}
+	}
+
+	hcp->daifmt[dai->id] = cf;
+
+	return ret;
+}
+
+static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+
+	dev_dbg(dai->dev, "%s()\n", __func__);
+
+	if (hcp->hcd.ops->digital_mute)
+		return hcp->hcd.ops->digital_mute(dai->dev->parent, mute);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
+	.startup	= hdmi_codec_startup,
+	.shutdown	= hdmi_codec_shutdown,
+	.hw_params	= hdmi_codec_hw_params,
+	.set_fmt	= hdmi_codec_set_fmt,
+	.digital_mute	= hdmi_codec_digital_mute,
+};
+
+
+#define HDMI_RATES	(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+			 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+			 SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\
+			 SNDRV_PCM_RATE_192000)
+
+#define SPDIF_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE)
+
+/*
+ * This list is only for formats allowed on the I2S bus. So there is
+ * some formats listed that are not supported by HDMI interface. For
+ * instance allowing the 32-bit formats enables 24-precision with CPU
+ * DAIs that do not support 24-bit formats. If the extra formats cause
+ * problems, we should add the video side driver an option to disable
+ * them.
+ */
+#define I2S_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\
+			 SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\
+			 SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\
+			 SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE)
+
+static struct snd_soc_dai_driver hdmi_i2s_dai = {
+	.name = "i2s-hifi",
+	.id = DAI_ID_I2S,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = HDMI_RATES,
+		.formats = I2S_FORMATS,
+		.sig_bits = 24,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static const struct snd_soc_dai_driver hdmi_spdif_dai = {
+	.name = "spdif-hifi",
+	.id = DAI_ID_SPDIF,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = HDMI_RATES,
+		.formats = SPDIF_FORMATS,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static struct snd_soc_codec_driver hdmi_codec = {
+	.dapm_widgets = hdmi_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
+	.dapm_routes = hdmi_routes,
+	.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
+};
+
+static int hdmi_codec_probe(struct platform_device *pdev)
+{
+	struct hdmi_codec_pdata *hcd = pdev->dev.platform_data;
+	struct device *dev = &pdev->dev;
+	struct hdmi_codec_priv *hcp;
+	int dai_count, i = 0;
+	int ret;
+
+	dev_dbg(dev, "%s()\n", __func__);
+
+	if (!hcd) {
+		dev_err(dev, "%s: No plalform data\n", __func__);
+		return -EINVAL;
+	}
+
+	dai_count = hcd->i2s + hcd->spdif;
+	if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params ||
+	    !hcd->ops->audio_shutdown) {
+		dev_err(dev, "%s: Invalid parameters\n", __func__);
+		return -EINVAL;
+	}
+
+	hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL);
+	if (!hcp)
+		return -ENOMEM;
+
+	hcp->hcd = *hcd;
+	mutex_init(&hcp->current_stream_lock);
+
+	hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv),
+				   GFP_KERNEL);
+	if (!hcp->daidrv)
+		return -ENOMEM;
+
+	if (hcd->i2s) {
+		hcp->daidrv[i] = hdmi_i2s_dai;
+		hcp->daidrv[i].playback.channels_max =
+			hcd->max_i2s_channels;
+		i++;
+	}
+
+	if (hcd->spdif)
+		hcp->daidrv[i] = hdmi_spdif_dai;
+
+	ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv,
+				     dai_count);
+	if (ret) {
+		dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n",
+			__func__, ret);
+		return ret;
+	}
+
+	dev_set_drvdata(dev, hcp);
+	return 0;
+}
+
+static int hdmi_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver hdmi_codec_driver = {
+	.driver = {
+		.name = HDMI_CODEC_DRV_NAME,
+	},
+	.probe = hdmi_codec_probe,
+	.remove = hdmi_codec_remove,
+};
+
+module_platform_driver(hdmi_codec_driver);
+
+MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>");
+MODULE_DESCRIPTION("HDMI Audio Codec Driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME);
-- 
cgit v1.2.3


From 2f0ad49104cbb19db24442af736614659363d2ab Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@linux.intel.com>
Date: Tue, 19 Apr 2016 13:12:35 +0800
Subject: ASoC: Change DAI link's be_id to a generic id

The generic ID can be used by topology:
- Toplogy can create FE links and set their ID, machine drivers will
  be notified and check this ID for machine-specific init.
- Toplogy can use the ID to find existing BE & CC links and further
  configure them.

Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h                             |  2 +-
 sound/soc/intel/boards/broadwell.c              |  2 +-
 sound/soc/intel/boards/bytcr_rt5640.c           |  2 +-
 sound/soc/intel/boards/bytcr_rt5651.c           |  2 +-
 sound/soc/intel/boards/cht_bsw_max98090_ti.c    |  2 +-
 sound/soc/intel/boards/cht_bsw_rt5645.c         |  2 +-
 sound/soc/intel/boards/cht_bsw_rt5672.c         |  2 +-
 sound/soc/intel/boards/haswell.c                |  2 +-
 sound/soc/intel/boards/skl_nau88l25_max98357a.c | 12 ++++++------
 sound/soc/intel/boards/skl_nau88l25_ssm4567.c   | 12 ++++++------
 sound/soc/intel/boards/skl_rt286.c              | 10 +++++-----
 11 files changed, 25 insertions(+), 25 deletions(-)

(limited to 'include')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 02b4a215fd75..ef25e86d51ee 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1002,7 +1002,7 @@ struct snd_soc_dai_link {
 	 */
 	const char *platform_name;
 	struct device_node *platform_of_node;
-	int be_id;	/* optional ID for machine driver BE identification */
+	int id;	/* optional ID for machine driver link identification */
 
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 3f8a1e10bed0..7486a0022fde 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "snd-soc-dummy-dai",
 		.platform_name = "snd-soc-dummy",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 032a2e753f0b..88efb62439ba 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
 		/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c
index 1c95ccc886c4..35f591eab3c9 100644
--- a/sound/soc/intel/boards/bytcr_rt5651.c
+++ b/sound/soc/intel/boards/bytcr_rt5651.c
@@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
index e609f089593a..6260df6bd49c 100644
--- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c
+++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c
@@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
index 2a6f80843bc9..0618a7f1025b 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5645.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	/* back ends */
 	{
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
index 2e5347f8f96c..df9d254baa18 100644
--- a/sound/soc/intel/boards/cht_bsw_rt5672.c
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = {
 	{
 		/* SSP2 - Codec */
 		.name = "SSP2-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "ssp2-port",
 		.platform_name = "sst-mfld-platform",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 22558572cb9c..863f1d5e2a2c 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "snd-soc-dummy-dai",
 		.platform_name = "snd-soc-dummy",
 		.no_pcm = 1,
diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
index 72176b79a18d..9cc9240ed717 100644
--- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c
+++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c
@@ -456,7 +456,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -472,7 +472,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP1 - Codec */
 		.name = "SSP1-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "SSP1 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -489,7 +489,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -501,7 +501,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -512,7 +512,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 5,
+		.id = 5,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
index 5f1ca99ae9b0..53380b2cb1a8 100644
--- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
+++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c
@@ -505,7 +505,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -523,7 +523,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	{
 		/* SSP1 - Codec */
 		.name = "SSP1-Codec",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "SSP1 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -540,7 +540,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -552,7 +552,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -563,7 +563,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -574,7 +574,7 @@ static struct snd_soc_dai_link skylake_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 5,
+		.id = 5,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c
index 2016397a8e75..9e39fc1b89d3 100644
--- a/sound/soc/intel/boards/skl_rt286.c
+++ b/sound/soc/intel/boards/skl_rt286.c
@@ -375,7 +375,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	{
 		/* SSP0 - Codec */
 		.name = "SSP0-Codec",
-		.be_id = 0,
+		.id = 0,
 		.cpu_dai_name = "SSP0 Pin",
 		.platform_name = "0000:00:1f.3",
 		.no_pcm = 1,
@@ -393,7 +393,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "dmic01",
-		.be_id = 1,
+		.id = 1,
 		.cpu_dai_name = "DMIC01 Pin",
 		.codec_name = "dmic-codec",
 		.codec_dai_name = "dmic-hifi",
@@ -405,7 +405,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp1",
-		.be_id = 2,
+		.id = 2,
 		.cpu_dai_name = "iDisp1 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi1",
@@ -416,7 +416,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp2",
-		.be_id = 3,
+		.id = 3,
 		.cpu_dai_name = "iDisp2 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi2",
@@ -427,7 +427,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = {
 	},
 	{
 		.name = "iDisp3",
-		.be_id = 4,
+		.id = 4,
 		.cpu_dai_name = "iDisp3 Pin",
 		.codec_name = "ehdaudio0D2",
 		.codec_dai_name = "intel-hdmi-hifi3",
-- 
cgit v1.2.3


From 305e9020f09d28560373c0112682e6fd11e909f6 Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@linux.intel.com>
Date: Tue, 19 Apr 2016 13:12:25 +0800
Subject: ASoC: Export snd_soc_find_dai()

This API can be used by topology to find an existing BE dai by name
and further configure it.

Topology will also check DAI ID to avoid wrong match.

Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc.h  | 3 +++
 sound/soc/soc-core.c | 3 ++-
 2 files changed, 5 insertions(+), 1 deletion(-)

(limited to 'include')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 02b4a215fd75..7687e2d4b0e4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 int snd_soc_register_dai(struct snd_soc_component *component,
 	struct snd_soc_dai_driver *dai_drv);
 
+struct snd_soc_dai *snd_soc_find_dai(
+	const struct snd_soc_dai_link_component *dlc);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index d2e62b159610..07663def2db6 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -930,7 +930,7 @@ static struct snd_soc_component *soc_find_component(
 	return NULL;
 }
 
-static struct snd_soc_dai *snd_soc_find_dai(
+struct snd_soc_dai *snd_soc_find_dai(
 	const struct snd_soc_dai_link_component *dlc)
 {
 	struct snd_soc_component *component;
@@ -959,6 +959,7 @@ static struct snd_soc_dai *snd_soc_find_dai(
 
 	return NULL;
 }
+EXPORT_SYMBOL_GPL(snd_soc_find_dai);
 
 static bool soc_is_dai_link_bound(struct snd_soc_card *card,
 		struct snd_soc_dai_link *dai_link)
-- 
cgit v1.2.3


From 34ce71a96dcba24c71b07f1b087bd179b885784d Mon Sep 17 00:00:00 2001
From: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Date: Sat, 23 Apr 2016 02:58:05 +0200
Subject: ALSA: timer: remove legacy rtctimer

There are no users of rtctimer left. Remove its code as this is the
in-kernel user of the legacy PC RTC driver that will hopefully be removed
at some point.

Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/uapi/sound/asound.h |   2 +-
 sound/core/Kconfig          |  29 -------
 sound/core/Makefile         |   1 -
 sound/core/rtctimer.c       | 187 --------------------------------------------
 sound/core/seq/seq.c        |   2 -
 sound/core/timer.c          |   2 -
 6 files changed, 1 insertion(+), 222 deletions(-)
 delete mode 100644 sound/core/rtctimer.c

(limited to 'include')

diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 67bf49d8c944..609cadb8739d 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -672,7 +672,7 @@ enum {
 
 /* global timers (device member) */
 #define SNDRV_TIMER_GLOBAL_SYSTEM	0
-#define SNDRV_TIMER_GLOBAL_RTC		1
+#define SNDRV_TIMER_GLOBAL_RTC		1	/* unused */
 #define SNDRV_TIMER_GLOBAL_HPET		2
 #define SNDRV_TIMER_GLOBAL_HRTIMER	3
 
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6d12ca9bcb80..9749f9e8b45c 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT
 	  Say Y here to use the HR-timer backend as the default sequencer
 	  timer.
 
-config SND_RTCTIMER
-	tristate "RTC Timer support"
-	depends on RTC
-	select SND_TIMER
-	help
-	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
-	  the RTC timer as a precise timing source and maps the RTC
-	  timer to ALSA's timer interface.  The ALSA sequencer code also
-	  can use this timing source.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-rtctimer.
-
-	  Note that this option is exclusive with the new RTC drivers
-	  (CONFIG_RTC_CLASS) since this requires the old API.
-
-config SND_SEQ_RTCTIMER_DEFAULT
-	bool "Use RTC as default sequencer timer"
-	depends on SND_RTCTIMER && SND_SEQUENCER
-	depends on !SND_SEQ_HRTIMER_DEFAULT
-	default y
-	help
-	  Say Y here to use the RTC timer as the default sequencer
-	  timer.  This is strongly recommended because it ensures
-	  precise MIDI timing even when the system timer runs at less
-	  than 1000 Hz.
-
-	  If in doubt, say Y.
-
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
 	help
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 48ab4b8f8279..e85d9dd12c2d 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -37,7 +37,6 @@ obj-$(CONFIG_SND) 		+= snd.o
 obj-$(CONFIG_SND_HWDEP)		+= snd-hwdep.o
 obj-$(CONFIG_SND_TIMER)		+= snd-timer.o
 obj-$(CONFIG_SND_HRTIMER)	+= snd-hrtimer.o
-obj-$(CONFIG_SND_RTCTIMER)	+= snd-rtctimer.o
 obj-$(CONFIG_SND_PCM)		+= snd-pcm.o
 obj-$(CONFIG_SND_DMAENGINE_PCM)	+= snd-pcm-dmaengine.o
 obj-$(CONFIG_SND_RAWMIDI)	+= snd-rawmidi.o
diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
deleted file mode 100644
index f3420d11a12f..000000000000
--- a/sound/core/rtctimer.c
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  RTC based high-frequency timer
- *
- *  Copyright (C) 2000 Takashi Iwai
- *	based on rtctimer.c by Steve Ratcliffe
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License as published by
- *   the Free Software Foundation; either version 2 of the License, or
- *   (at your option) any later version.
- *
- *   This program is distributed in the hope that it will be useful,
- *   but WITHOUT ANY WARRANTY; without even the implied warranty of
- *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- *   GNU General Public License for more details.
- *
- *   You should have received a copy of the GNU General Public License
- *   along with this program; if not, write to the Free Software
- *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
- *
- */
-
-#include <linux/init.h>
-#include <linux/interrupt.h>
-#include <linux/module.h>
-#include <linux/log2.h>
-#include <sound/core.h>
-#include <sound/timer.h>
-
-#if IS_ENABLED(CONFIG_RTC)
-
-#include <linux/mc146818rtc.h>
-
-#define RTC_FREQ	1024		/* default frequency */
-#define NANO_SEC	1000000000L	/* 10^9 in sec */
-
-/*
- * prototypes
- */
-static int rtctimer_open(struct snd_timer *t);
-static int rtctimer_close(struct snd_timer *t);
-static int rtctimer_start(struct snd_timer *t);
-static int rtctimer_stop(struct snd_timer *t);
-
-
-/*
- * The hardware dependent description for this timer.
- */
-static struct snd_timer_hardware rtc_hw = {
-	.flags =	SNDRV_TIMER_HW_AUTO |
-			SNDRV_TIMER_HW_FIRST |
-			SNDRV_TIMER_HW_TASKLET,
-	.ticks =	100000000L,		/* FIXME: XXX */
-	.open =		rtctimer_open,
-	.close =	rtctimer_close,
-	.start =	rtctimer_start,
-	.stop =		rtctimer_stop,
-};
-
-static int rtctimer_freq = RTC_FREQ;		/* frequency */
-static struct snd_timer *rtctimer;
-static struct tasklet_struct rtc_tasklet;
-static rtc_task_t rtc_task;
-
-
-static int
-rtctimer_open(struct snd_timer *t)
-{
-	int err;
-
-	err = rtc_register(&rtc_task);
-	if (err < 0)
-		return err;
-	t->private_data = &rtc_task;
-	return 0;
-}
-
-static int
-rtctimer_close(struct snd_timer *t)
-{
-	rtc_task_t *rtc = t->private_data;
-	if (rtc) {
-		rtc_unregister(rtc);
-		tasklet_kill(&rtc_tasklet);
-		t->private_data = NULL;
-	}
-	return 0;
-}
-
-static int
-rtctimer_start(struct snd_timer *timer)
-{
-	rtc_task_t *rtc = timer->private_data;
-	if (snd_BUG_ON(!rtc))
-		return -EINVAL;
-	rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq);
-	rtc_control(rtc, RTC_PIE_ON, 0);
-	return 0;
-}
-
-static int
-rtctimer_stop(struct snd_timer *timer)
-{
-	rtc_task_t *rtc = timer->private_data;
-	if (snd_BUG_ON(!rtc))
-		return -EINVAL;
-	rtc_control(rtc, RTC_PIE_OFF, 0);
-	return 0;
-}
-
-static void rtctimer_tasklet(unsigned long data)
-{
-	snd_timer_interrupt((struct snd_timer *)data, 1);
-}
-
-/*
- * interrupt
- */
-static void rtctimer_interrupt(void *private_data)
-{
-	tasklet_schedule(private_data);
-}
-
-
-/*
- *  ENTRY functions
- */
-static int __init rtctimer_init(void)
-{
-	int err;
-	struct snd_timer *timer;
-
-	if (rtctimer_freq < 2 || rtctimer_freq > 8192 ||
-	    !is_power_of_2(rtctimer_freq)) {
-		pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq);
-		return -EINVAL;
-	}
-
-	/* Create a new timer and set up the fields */
-	err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer);
-	if (err < 0)
-		return err;
-
-	timer->module = THIS_MODULE;
-	strcpy(timer->name, "RTC timer");
-	timer->hw = rtc_hw;
-	timer->hw.resolution = NANO_SEC / rtctimer_freq;
-
-	tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer);
-
-	/* set up RTC callback */
-	rtc_task.func = rtctimer_interrupt;
-	rtc_task.private_data = &rtc_tasklet;
-
-	err = snd_timer_global_register(timer);
-	if (err < 0) {
-		snd_timer_global_free(timer);
-		return err;
-	}
-	rtctimer = timer; /* remember this */
-
-	return 0;
-}
-
-static void __exit rtctimer_exit(void)
-{
-	if (rtctimer) {
-		snd_timer_global_free(rtctimer);
-		rtctimer = NULL;
-	}
-}
-
-
-/*
- * exported stuff
- */
-module_init(rtctimer_init)
-module_exit(rtctimer_exit)
-
-module_param(rtctimer_freq, int, 0444);
-MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz");
-
-MODULE_LICENSE("GPL");
-
-MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC));
-
-#endif /* IS_ENABLED(CONFIG_RTC) */
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 7e0aabb808a6..639544b4fb04 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -47,8 +47,6 @@ int seq_default_timer_card = -1;
 int seq_default_timer_device =
 #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT
 	SNDRV_TIMER_GLOBAL_HRTIMER
-#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT)
-	SNDRV_TIMER_GLOBAL_RTC
 #else
 	SNDRV_TIMER_GLOBAL_SYSTEM
 #endif
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 6469bedda2f3..0cfc028c1193 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -37,8 +37,6 @@
 
 #if IS_ENABLED(CONFIG_SND_HRTIMER)
 #define DEFAULT_TIMER_LIMIT 4
-#elif IS_ENABLED(CONFIG_SND_RTCTIMER)
-#define DEFAULT_TIMER_LIMIT 2
 #else
 #define DEFAULT_TIMER_LIMIT 1
 #endif
-- 
cgit v1.2.3


From 73fe01cfb3babff01748a9fbc95cc3ea2079cc7f Mon Sep 17 00:00:00 2001
From: Matthias Reichl <hias@horus.com>
Date: Wed, 27 Apr 2016 15:26:51 +0200
Subject: ASoC: dmaengine_pcm: Add support for packed transfers

dmaengine_pcm currently only supports setups where FIFO reads/writes
correspond to exactly one sample, eg 16-bit sample data is transferred
via 16-bit FIFO accesses, 32-bit data via 32-bit accesses.

This patch adds support for setups with fixed width FIFOs where
multiple samples are packed into a larger word.

For example setups with a 32-bit wide FIFO register that expect
16-bit sample transfers to be done with the left+right sample data
packed into a 32-bit word.

Support for packed transfers is controlled via the
SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags

If this flag is set dmaengine_pcm doesn't put any restriction on the
supported formats and sets the DMA transfer width to undefined.

This means control over the constraints is now transferred to the DAI
driver and it's responsible to provide proper configuration and
check for possible corner cases that aren't handled by the ALSA core.

Signed-off-by: Matthias Reichl <hias@horus.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Martin Sperl <kernel@martin.sperl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/dmaengine_pcm.h         | 12 ++++++++
 sound/core/pcm_dmaengine.c            | 11 +++++--
 sound/soc/soc-generic-dmaengine-pcm.c | 57 +++++++++++++++++++++--------------
 3 files changed, 55 insertions(+), 25 deletions(-)

(limited to 'include')

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index f86ef5ea9b01..67be2445941a 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
 	void *filter_data);
 struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
 
+/*
+ * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word.
+ * If this flag is set the dmaengine driver won't put any restriction on
+ * the supported sample formats and set the DMA transfer size to undefined.
+ * The DAI driver is responsible to disable any unsupported formats in it's
+ * configuration and catch corner cases that are not already handled in
+ * the ALSA core.
+ */
+#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0)
+
 /**
  * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
  * @addr: Address of the DAI data source or destination register.
@@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * requesting the DMA channel.
  * @chan_name: Custom channel name to use when requesting DMA channel.
  * @fifo_size: FIFO size of the DAI controller in bytes
+ * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now
  */
 struct snd_dmaengine_dai_dma_data {
 	dma_addr_t addr;
@@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data {
 	void *filter_data;
 	const char *chan_name;
 	unsigned int fifo_size;
+	unsigned int flags;
 };
 
 void snd_dmaengine_pcm_set_config_from_dai_data(
diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 697c166acf05..8eb58c709b14 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/sound/core/pcm_dmaengine.c
@@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config);
  * direction of the substream. If the substream is a playback stream the dst
  * fields will be initialized, if it is a capture stream the src fields will be
  * initialized. The {dst,src}_addr_width field will only be initialized if the
- * addr_width field of the DAI DMA data struct is not equal to
- * DMA_SLAVE_BUSWIDTH_UNDEFINED.
+ * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of
+ * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If
+ * both conditions are met the latter takes priority.
  */
 void snd_dmaengine_pcm_set_config_from_dai_data(
 	const struct snd_pcm_substream *substream,
@@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		slave_config->dst_addr = dma_data->addr;
 		slave_config->dst_maxburst = dma_data->maxburst;
+		if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+			slave_config->dst_addr_width =
+				DMA_SLAVE_BUSWIDTH_UNDEFINED;
 		if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
 			slave_config->dst_addr_width = dma_data->addr_width;
 	} else {
 		slave_config->src_addr = dma_data->addr;
 		slave_config->src_maxburst = dma_data->maxburst;
+		if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)
+			slave_config->src_addr_width =
+				DMA_SLAVE_BUSWIDTH_UNDEFINED;
 		if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED)
 			slave_config->src_addr_width = dma_data->addr_width;
 	}
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index 6fd1906af387..6cef3977507a 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/sound/soc/soc-generic-dmaengine-pcm.c
@@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea
 	}
 
 	/*
-	 * Prepare formats mask for valid/allowed sample types. If the dma does
-	 * not have support for the given physical word size, it needs to be
-	 * masked out so user space can not use the format which produces
-	 * corrupted audio.
-	 * In case the dma driver does not implement the slave_caps the default
-	 * assumption is that it supports 1, 2 and 4 bytes widths.
+	 * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
+	 * hw.formats set to 0, meaning no restrictions are in place.
+	 * In this case it's the responsibility of the DAI driver to
+	 * provide the supported format information.
 	 */
-	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
-		int bits = snd_pcm_format_physical_width(i);
-
-		/* Enable only samples with DMA supported physical widths */
-		switch (bits) {
-		case 8:
-		case 16:
-		case 24:
-		case 32:
-		case 64:
-			if (addr_widths & (1 << (bits / 8)))
-				hw.formats |= (1LL << i);
-			break;
-		default:
-			/* Unsupported types */
-			break;
+	if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
+		/*
+		 * Prepare formats mask for valid/allowed sample types. If the
+		 * dma does not have support for the given physical word size,
+		 * it needs to be masked out so user space can not use the
+		 * format which produces corrupted audio.
+		 * In case the dma driver does not implement the slave_caps the
+		 * default assumption is that it supports 1, 2 and 4 bytes
+		 * widths.
+		 */
+		for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+			int bits = snd_pcm_format_physical_width(i);
+
+			/*
+			 * Enable only samples with DMA supported physical
+			 * widths
+			 */
+			switch (bits) {
+			case 8:
+			case 16:
+			case 24:
+			case 32:
+			case 64:
+				if (addr_widths & (1 << (bits / 8)))
+					hw.formats |= (1LL << i);
+				break;
+			default:
+				/* Unsupported types */
+				break;
+			}
 		}
-	}
 
 	return snd_soc_set_runtime_hwparams(substream, &hw);
 }
-- 
cgit v1.2.3


From 4d2458507d0b465c62ae80f3e81b8c008ec96b05 Mon Sep 17 00:00:00 2001
From: Fabio Estevam <fabio.estevam@nxp.com>
Date: Wed, 4 May 2016 19:33:59 -0300
Subject: ASoC: fsl_sai: Allow setting the SAI MCLK direction

On mx6ul the General Purpose Register 1 (GPR1) contains the following
bits for configuring the direction of the SAI MCLKs:
SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR

Introduce  the "fsl,sai-mclk-direction-output" optional property to allow
configuring the SAI_MCLK outputs.

Tested on a imx6ul-evk board.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/fsl-sai.txt |  5 +++++
 include/linux/mfd/syscon/imx6q-iomuxc-gpr.h         |  6 ++++++
 sound/soc/fsl/fsl_sai.c                             | 20 ++++++++++++++++++++
 3 files changed, 31 insertions(+)

(limited to 'include')

diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 777b941d6cbe..740b467adf7d 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -48,6 +48,11 @@ Required properties:
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
+Optional properties (for mx6ul):
+
+  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+			 indicates that SAI will output the SAI MCLK clock.
+
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
   default synchronous mode (sync Rx with Tx) will be used, which means both
diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
index 238c8db953eb..68353822afce 100644
--- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
+++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
@@ -447,5 +447,11 @@
 #define IMX6UL_GPR1_ENET2_CLK_OUTPUT		(0x1 << 18)
 #define IMX6UL_GPR1_ENET_CLK_DIR		(0x3 << 17)
 #define IMX6UL_GPR1_ENET_CLK_OUTPUT		(0x3 << 17)
+#define IMX6UL_GPR1_SAI1_MCLK_DIR		(0x1 << 19)
+#define IMX6UL_GPR1_SAI2_MCLK_DIR		(0x1 << 20)
+#define IMX6UL_GPR1_SAI3_MCLK_DIR		(0x1 << 21)
+#define IMX6UL_GPR1_SAI_MCLK_MASK		(0x7 << 19)
+#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \
+		     IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR)
 
 #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index d8b673f7c577..2147994ab46f 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -21,6 +21,8 @@
 #include <sound/core.h>
 #include <sound/dmaengine_pcm.h>
 #include <sound/pcm_params.h>
+#include <linux/mfd/syscon.h>
+#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h>
 
 #include "fsl_sai.h"
 #include "imx-pcm.h"
@@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev)
 {
 	struct device_node *np = pdev->dev.of_node;
 	struct fsl_sai *sai;
+	struct regmap *gpr;
 	struct resource *res;
 	void __iomem *base;
 	char tmp[8];
 	int irq, ret, i;
+	int index;
 
 	sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL);
 	if (!sai)
@@ -878,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev)
 		fsl_sai_dai.symmetric_samplebits = 0;
 	}
 
+	if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) &&
+	    of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) {
+		gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr");
+		if (IS_ERR(gpr)) {
+			dev_err(&pdev->dev, "cannot find iomuxc registers\n");
+			return PTR_ERR(gpr);
+		}
+
+		index = of_alias_get_id(np, "sai");
+		if (index < 0)
+			return index;
+
+		regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index),
+				   MCLK_DIR(index));
+	}
+
 	sai->dma_params_rx.addr = res->start + FSL_SAI_RDR;
 	sai->dma_params_tx.addr = res->start + FSL_SAI_TDR;
 	sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX;
-- 
cgit v1.2.3


From 32902177f7f6ae70e1d5e71d935aa1bfcae7f01c Mon Sep 17 00:00:00 2001
From: John Keeping <john@metanate.com>
Date: Thu, 12 May 2016 13:55:53 +0100
Subject: ASoC: dapm: deprecate MICBIAS widget type

Commit 086d7f804e26 ("ASoC: Convert WM8962 MICBIAS to a supply widget",
2011-09-23) says:

	A supply widget is generally clearer than a MICBIAS widget and a
	mic bias is just a type of supply so use a supply widget for the
	MICBIAS. This also avoids confusion with the routing when
	connected to multiple inputs.

but this has never been documented as a policy.  Add some comments to
make it clear.

Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/soc-dapm.h | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'include')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 97069466c38d..3101d53468aa 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -100,6 +100,7 @@ struct device;
 {       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_micbias, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
@@ -473,7 +474,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_out_drv,			/* output driver */
 	snd_soc_dapm_adc,			/* analog to digital converter */
 	snd_soc_dapm_dac,			/* digital to analog converter */
-	snd_soc_dapm_micbias,		/* microphone bias (power) */
+	snd_soc_dapm_micbias,		/* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */
 	snd_soc_dapm_mic,			/* microphone */
 	snd_soc_dapm_hp,			/* headphones */
 	snd_soc_dapm_spk,			/* speaker */
-- 
cgit v1.2.3


From 4446085d21e75dd6c0c45577f12db0bd7c7bf35f Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Thu, 12 May 2016 08:58:53 +0530
Subject: ALSA: hdac: add link pm and ref counting

The HDA links can be switched off when not is use, similarly
command DMA can be stopped as well. This calls for a reference
counting mechanism on the link by it's users to manage the link
power. The DMA can be turned off when all links are off

For this we add two APIs
	snd_hdac_ext_bus_link_get
	snd_hdac_ext_bus_link_put

They help users to turn up/down link and manage the DMA as well

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 include/sound/hdaudio_ext.h         | 13 ++++++++
 sound/hda/ext/hdac_ext_bus.c        |  3 ++
 sound/hda/ext/hdac_ext_controller.c | 66 +++++++++++++++++++++++++++++++++++++
 3 files changed, 82 insertions(+)

(limited to 'include')

diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index 07fa59237feb..b9593b201599 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -14,6 +14,8 @@
  * @gtscap: gts capabilities pointer
  * @drsmcap: dma resume capabilities pointer
  * @hlink_list: link list of HDA links
+ * @lock: lock for link mgmt
+ * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
  */
 struct hdac_ext_bus {
 	struct hdac_bus bus;
@@ -27,6 +29,9 @@ struct hdac_ext_bus {
 	void __iomem *drsmcap;
 
 	struct list_head hlink_list;
+
+	struct mutex lock;
+	bool cmd_dma_state;
 };
 
 int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev,
@@ -142,6 +147,9 @@ struct hdac_ext_link {
 	void __iomem *ml_addr; /* link output stream reg pointer */
 	u32 lcaps;   /* link capablities */
 	u16 lsdiid;  /* link sdi identifier */
+
+	int ref_count;
+
 	struct list_head list;
 };
 
@@ -154,6 +162,11 @@ void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link,
 				 int stream);
 
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link);
+
 /* update register macro */
 #define snd_hdac_updatel(addr, reg, mask, val)		\
 	writel(((readl(addr + reg) & ~(mask)) | (val)), \
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index 2433f7c81472..3b7ae24900fd 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev,
 	INIT_LIST_HEAD(&ebus->hlink_list);
 	ebus->idx = idx++;
 
+	mutex_init(&ebus->lock);
+	ebus->cmd_dma_state = true;
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init);
diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c
index 548cc1e4114b..860f8cad6602 100644
--- a/sound/hda/ext/hdac_ext_controller.c
+++ b/sound/hda/ext/hdac_ext_controller.c
@@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus)
 		hlink->lcaps  = readl(hlink->ml_addr + AZX_REG_ML_LCAP);
 		hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID);
 
+		/* since link in On, update the ref */
+		hlink->ref_count = 1;
+
 		list_add_tail(&hlink->list, &ebus->hlink_list);
 	}
 
@@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus)
 	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all);
+
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link)
+{
+	int ret = 0;
+
+	mutex_lock(&ebus->lock);
+
+	/*
+	 * if we move from 0 to 1, count will be 1 so power up this link
+	 * as well, also check the dma status and trigger that
+	 */
+	if (++link->ref_count == 1) {
+		if (!ebus->cmd_dma_state) {
+			snd_hdac_bus_init_cmd_io(&ebus->bus);
+			ebus->cmd_dma_state = true;
+		}
+
+		ret = snd_hdac_ext_bus_link_power_up(link);
+	}
+
+	mutex_unlock(&ebus->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get);
+
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link)
+{
+	int ret = 0;
+	struct hdac_ext_link *hlink;
+	bool link_up = false;
+
+	mutex_lock(&ebus->lock);
+
+	/*
+	 * if we move from 1 to 0, count will be 0
+	 * so power down this link as well
+	 */
+	if (--link->ref_count == 0) {
+		ret = snd_hdac_ext_bus_link_power_down(link);
+
+		/*
+		 * now check if all links are off, if so turn off
+		 * cmd dma as well
+		 */
+		list_for_each_entry(hlink, &ebus->hlink_list, list) {
+			if (hlink->ref_count) {
+				link_up = true;
+				break;
+			}
+		}
+
+		if (!link_up) {
+			snd_hdac_bus_stop_cmd_io(&ebus->bus);
+			ebus->cmd_dma_state = false;
+		}
+	}
+
+	mutex_unlock(&ebus->lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put);
-- 
cgit v1.2.3