From 73a2cd9193ae50d4cc7c447f8929e13010b589be Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:04 -0700
Subject: ASoC: fsl_esai: Add indentation for binding doc to increase
 readability

This patch simply adds indentations for DT binding doc to increase readability
without changing any contents.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/fsl,esai.txt         | 44 +++++++++++-----------
 1 file changed, 23 insertions(+), 21 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/fsl,esai.txt b/Documentation/devicetree/bindings/sound/fsl,esai.txt
index 52f5b6bf3e8e..d3b6b5f48010 100644
--- a/Documentation/devicetree/bindings/sound/fsl,esai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,esai.txt
@@ -7,37 +7,39 @@ other DSPs. It has up to six transmitters and four receivers.
 
 Required properties:
 
-  - compatible : Compatible list, must contain "fsl,imx35-esai" or
-		 "fsl,vf610-esai"
+  - compatible		: Compatible list, must contain "fsl,imx35-esai" or
+			  "fsl,vf610-esai"
 
-  - reg : Offset and length of the register set for the device.
+  - reg			: Offset and length of the register set for the device.
 
-  - interrupts : Contains the spdif interrupt.
+  - interrupts		: Contains the spdif interrupt.
 
-  - dmas : Generic dma devicetree binding as described in
-  Documentation/devicetree/bindings/dma/dma.txt.
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
 
-  - dma-names : Two dmas have to be defined, "tx" and "rx".
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
 
-  - clocks: Contains an entry for each entry in clock-names.
+  - clocks		: Contains an entry for each entry in clock-names.
 
-  - clock-names : Includes the following entries:
-	"core"		The core clock used to access registers
-	"extal"		The esai baud clock for esai controller used to derive
-			HCK, SCK and FS.
-	"fsys"		The system clock derived from ahb clock used to derive
-			HCK, SCK and FS.
+  - clock-names		: Includes the following entries:
+	"core"		  The core clock used to access registers
+	"extal"		  The esai baud clock for esai controller used to
+			  derive HCK, SCK and FS.
+	"fsys"		  The system clock derived from ahb clock used to
+			  derive HCK, SCK and FS.
 
-  - fsl,fifo-depth: The number of elements in the transmit and receive FIFOs.
-    This number is the maximum allowed value for TFCR[TFWM] or RFCR[RFWM].
+  - fsl,fifo-depth	: The number of elements in the transmit and receive
+			  FIFOs. This number is the maximum allowed value for
+			  TFCR[TFWM] or RFCR[RFWM].
 
   - fsl,esai-synchronous: This is a boolean property. If present, indicating
-    that ESAI would work in the synchronous mode, which means all the settings
-    for Receiving would be duplicated from Transmition related registers.
+			  that ESAI would work in the synchronous mode, which
+			  means all the settings for Receiving would be
+			  duplicated from Transmition related registers.
 
-  - big-endian : If this property is absent, the native endian mode will
-    be in use as default, or the big endian mode will be in use for all the
-    device registers.
+  - big-endian		: If this property is absent, the native endian mode
+			  will be in use as default, or the big endian mode
+			  will be in use for all the device registers.
 
 Example:
 
-- 
cgit v1.2.3-59-g8ed1b


From 9c4c1045343836b848c3dd546277c955d145d20a Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:05 -0700
Subject: ASoC: fsl_spdif: Add indentation for binding doc to increase
 readability

This patch simply adds indentations for DT binding doc to increase readability
without changing any contents.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/fsl,spdif.txt        | 37 +++++++++++-----------
 1 file changed, 18 insertions(+), 19 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
index 3e9e82c8eab3..b5ee32ee3706 100644
--- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt
+++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt
@@ -6,32 +6,31 @@ a fibre cable.
 
 Required properties:
 
-  - compatible : Compatible list, must contain "fsl,imx35-spdif".
+  - compatible		: Compatible list, must contain "fsl,imx35-spdif".
 
-  - reg : Offset and length of the register set for the device.
+  - reg			: Offset and length of the register set for the device.
 
-  - interrupts : Contains the spdif interrupt.
+  - interrupts		: Contains the spdif interrupt.
 
-  - dmas : Generic dma devicetree binding as described in
-  Documentation/devicetree/bindings/dma/dma.txt.
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
 
-  - dma-names : Two dmas have to be defined, "tx" and "rx".
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
 
-  - clocks : Contains an entry for each entry in clock-names.
+  - clocks		: Contains an entry for each entry in clock-names.
 
-  - clock-names : Includes the following entries:
-	"core"		The core clock of spdif controller
-	"rxtx<0-7>"	Clock source list for tx and rx clock.
-			This clock list should be identical to
-			the source list connecting to the spdif
-			clock mux in "SPDIF Transceiver Clock
-			Diagram" of SoC reference manual. It
-			can also be referred to TxClk_Source
-			bit of register SPDIF_STC.
+  - clock-names		: Includes the following entries:
+	"core"		  The core clock of spdif controller.
+	"rxtx<0-7>"	  Clock source list for tx and rx clock.
+			  This clock list should be identical to the source
+			  list connecting to the spdif clock mux in "SPDIF
+			  Transceiver Clock Diagram" of SoC reference manual.
+			  It can also be referred to TxClk_Source bit of
+			  register SPDIF_STC.
 
-   - big-endian : If this property is absent, the native endian mode will
-   be in use as default, or the big endian mode will be in use for all the
-   device registers.
+   - big-endian		: If this property is absent, the native endian mode
+			  will be in use as default, or the big endian mode
+			  will be in use for all the device registers.
 
 Example:
 
-- 
cgit v1.2.3-59-g8ed1b


From 0b9938b264d1a76458cf06aab6de7b9cec68efca Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:06 -0700
Subject: ASoC: fsl_sai: Add indentation for binding doc to increase
 readability

This patch refines the DT binding doc for more readability by adding
extra blank lines and indentations.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/fsl-sai.txt          | 66 ++++++++++++++--------
 1 file changed, 41 insertions(+), 25 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 4956b14d4b06..044e5d76e2dd 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -5,32 +5,48 @@ which provides a synchronous audio interface that supports fullduplex
 serial interfaces with frame synchronization such as I2S, AC97, TDM, and
 codec/DSP interfaces.
 
-
 Required properties:
-- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai".
-- reg: Offset and length of the register set for the device.
-- clocks: Must contain an entry for each entry in clock-names.
-- clock-names : Must include the "bus" for register access and "mclk1" "mclk2"
-  "mclk3" for bit clock and frame clock providing.
-- dmas : Generic dma devicetree binding as described in
-  Documentation/devicetree/bindings/dma/dma.txt.
-- dma-names : Two dmas have to be defined, "tx" and "rx".
-- pinctrl-names: Must contain a "default" entry.
-- pinctrl-NNN: One property must exist for each entry in pinctrl-names.
-  See ../pinctrl/pinctrl-bindings.txt for details of the property values.
-- big-endian: Boolean property, required if all the FTM_PWM registers
-  are big-endian rather than little-endian.
-- lsb-first: Configures whether the LSB or the MSB is transmitted first for
-  the fifo data. If this property is absent, the MSB is transmitted first as
-  default, or the LSB is transmitted first.
-- fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
-  that SAI will work in the synchronous mode (sync Tx with Rx) which means
-  both the transimitter and receiver will send and receive data by following
-  receiver's bit clocks and frame sync clocks.
-- fsl,sai-asynchronous: This is a boolean property. If present, indicating
-  that SAI will work in the asynchronous mode, which means both transimitter
-  and receiver will send and receive data by following their own bit clocks
-  and frame sync clocks separately.
+
+  - compatible		: Compatible list, contains "fsl,vf610-sai" or
+			  "fsl,imx6sx-sai".
+
+  - reg			: Offset and length of the register set for the device.
+
+  - clocks		: Must contain an entry for each entry in clock-names.
+
+  - clock-names		: Must include the "bus" for register access and
+			  "mclk1", "mclk2", "mclk3" for bit clock and frame
+			  clock providing.
+  - dmas		: Generic dma devicetree binding as described in
+			  Documentation/devicetree/bindings/dma/dma.txt.
+
+  - dma-names		: Two dmas have to be defined, "tx" and "rx".
+
+  - pinctrl-names	: Must contain a "default" entry.
+
+  - pinctrl-NNN		: One property must exist for each entry in
+			  pinctrl-names. See ../pinctrl/pinctrl-bindings.txt
+			  for details of the property values.
+
+  - big-endian		: Boolean property, required if all the FTM_PWM
+			  registers are big-endian rather than little-endian.
+
+  - lsb-first		: Configures whether the LSB or the MSB is transmitted
+			  first for the fifo data. If this property is absent,
+			  the MSB is transmitted first as default, or the LSB
+			  is transmitted first.
+
+  - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating
+			  that SAI will work in the synchronous mode (sync Tx
+			  with Rx) which means both the transimitter and the
+			  receiver will send and receive data by following
+			  receiver's bit clocks and frame sync clocks.
+
+  - fsl,sai-asynchronous: This is a boolean property. If present, indicating
+			  that SAI will work in the asynchronous mode, which
+			  means both transimitter and receiver will send and
+			  receive data by following their own bit clocks and
+			  frame sync clocks separately.
 
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
-- 
cgit v1.2.3-59-g8ed1b


From d29ae41edde680c00f9f74d448b7d59c91ca0474 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:07 -0700
Subject: ASoC: eukrea-tlv320: Add indentation for binding doc to increase
 readability

This patch refines the DT binding doc for more readability by adding
extra blank lines and indentations.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt | 15 ++++++++++-----
 1 file changed, 10 insertions(+), 5 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
index 0d7985c864af..6dfa88c4dc1e 100644
--- a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
+++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt
@@ -1,11 +1,16 @@
 Audio complex for Eukrea boards with tlv320aic23 codec.
 
 Required properties:
-- compatible : "eukrea,asoc-tlv320"
-- eukrea,model : The user-visible name of this sound complex.
-- ssi-controller : The phandle of the SSI controller.
-- fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX).
-- fsl,mux-ext-port : The external port of the i.MX audio muxer.
+
+  - compatible		: "eukrea,asoc-tlv320"
+
+  - eukrea,model	: The user-visible name of this sound complex.
+
+  - ssi-controller	: The phandle of the SSI controller.
+
+  - fsl,mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX).
+
+  - fsl,mux-ext-port	: The external port of the i.MX audio muxer.
 
 Note: The AUDMUX port numbering should start at 1, which is consistent with
 hardware manual.
-- 
cgit v1.2.3-59-g8ed1b


From 5463c709ddeeacc73ae6844fd8aebc0a217b98d4 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:08 -0700
Subject: ASoC: imx-audmux: Add indentation for binding doc to increase
 readability

This patch refines the DT binding doc for more readability by adding
extra blank lines and indentations.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/imx-audmux.txt       | 22 ++++++++++++++--------
 1 file changed, 14 insertions(+), 8 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt
index f88a00e54c63..b30a737e209e 100644
--- a/Documentation/devicetree/bindings/sound/imx-audmux.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audmux.txt
@@ -1,18 +1,24 @@
 Freescale Digital Audio Mux (AUDMUX) device
 
 Required properties:
-- compatible : "fsl,imx21-audmux" for AUDMUX version firstly used on i.MX21,
-  or "fsl,imx31-audmux" for the version firstly used on i.MX31.
-- reg : Should contain AUDMUX registers location and length
+
+  - compatible		: "fsl,imx21-audmux" for AUDMUX version firstly used
+			  on i.MX21, or "fsl,imx31-audmux" for the version
+			  firstly used on i.MX31.
+
+  - reg			: Should contain AUDMUX registers location and length.
 
 An initial configuration can be setup using child nodes.
 
 Required properties of optional child nodes:
-- fsl,audmux-port : Integer of the audmux port that is configured by this
-  child node.
-- fsl,port-config : List of configuration options for the specific port. For
-  imx31-audmux and above, it is a list of tuples <ptcr pdcr>. For
-  imx21-audmux it is a list of pcr values.
+
+  - fsl,audmux-port	: Integer of the audmux port that is configured by this
+			  child node.
+
+  - fsl,port-config	: List of configuration options for the specific port.
+			  For imx31-audmux and above, it is a list of tuples
+			  <ptcr pdcr>. For imx21-audmux it is a list of pcr
+			  values.
 
 Example:
 
-- 
cgit v1.2.3-59-g8ed1b


From afa1fde676592b9c40e893f4f652f2c842e31683 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:09 -0700
Subject: ASoC: imx-sgtl5000: Add indentation for binding doc to increase
 readability

This patch refines the DT binding doc for more readability by adding
extra blank lines and indentations.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../bindings/sound/imx-audio-sgtl5000.txt          | 61 ++++++++++++----------
 1 file changed, 34 insertions(+), 27 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
index e4acdd891e49..2f89db88fd57 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt
@@ -1,33 +1,40 @@
 Freescale i.MX audio complex with SGTL5000 codec
 
 Required properties:
-- compatible : "fsl,imx-audio-sgtl5000"
-- model : The user-visible name of this sound complex
-- ssi-controller : The phandle of the i.MX SSI controller
-- audio-codec : The phandle of the SGTL5000 audio codec
-- audio-routing : A list of the connections between audio components.
-  Each entry is a pair of strings, the first being the connection's sink,
-  the second being the connection's source. Valid names could be power
-  supplies, SGTL5000 pins, and the jacks on the board:
-
-  Power supplies:
-   * Mic Bias
-
-  SGTL5000 pins:
-   * MIC_IN
-   * LINE_IN
-   * HP_OUT
-   * LINE_OUT
-
-  Board connectors:
-   * Mic Jack
-   * Line In Jack
-   * Headphone Jack
-   * Line Out Jack
-   * Ext Spk
-
-- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
-- mux-ext-port : The external port of the i.MX audio muxer
+
+  - compatible		: "fsl,imx-audio-sgtl5000"
+
+  - model		: The user-visible name of this sound complex
+
+  - ssi-controller	: The phandle of the i.MX SSI controller
+
+  - audio-codec		: The phandle of the SGTL5000 audio codec
+
+  - audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. Valid names could be power supplies, SGTL5000
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  SGTL5000 pins:
+			   * MIC_IN
+			   * LINE_IN
+			   * HP_OUT
+			   * LINE_OUT
+
+			  Board connectors:
+			   * Mic Jack
+			   * Line In Jack
+			   * Headphone Jack
+			   * Line Out Jack
+			   * Ext Spk
+
+  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
+
+  - mux-ext-port	: The external port of the i.MX audio muxer
 
 Note: The AUDMUX port numbering should start at 1, which is consistent with
 hardware manual.
-- 
cgit v1.2.3-59-g8ed1b


From 6219b082b3099668c32be99bd31216379fc3d97a Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:10 -0700
Subject: ASoC: imx-spdif: Add indentation for binding doc to increase
 readability

This patch simply adds indentations for DT binding doc to increase readability
without changing any contents.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/imx-audio-spdif.txt  | 22 ++++++++++++----------
 1 file changed, 12 insertions(+), 10 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
index 7d13479f9c3c..da84a442ccea 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -2,23 +2,25 @@ Freescale i.MX audio complex with S/PDIF transceiver
 
 Required properties:
 
-  - compatible : "fsl,imx-audio-spdif"
+  - compatible		: "fsl,imx-audio-spdif"
 
-  - model : The user-visible name of this sound complex
+  - model		: The user-visible name of this sound complex
 
-  - spdif-controller : The phandle of the i.MX S/PDIF controller
+  - spdif-controller	: The phandle of the i.MX S/PDIF controller
 
 
 Optional properties:
 
-  - spdif-out : This is a boolean property. If present, the transmitting
-    function of S/PDIF will be enabled, indicating there's a physical
-    S/PDIF out connector/jack on the board or it's connecting to some
-    other IP block, such as an HDMI encoder/display-controller.
+  - spdif-out		: This is a boolean property. If present, the
+			  transmitting function of S/PDIF will be enabled,
+			  indicating there's a physical S/PDIF out connector
+			  or jack on the board or it's connecting to some
+			  other IP block, such as an HDMI encoder or
+			  display-controller.
 
-  - spdif-in : This is a boolean property. If present, the receiving
-    function of S/PDIF will be enabled, indicating there's a physical
-    S/PDIF in connector/jack on the board.
+  - spdif-in		: This is a boolean property. If present, the receiving
+			  function of S/PDIF will be enabled, indicating there
+			  is a physical S/PDIF in connector/jack on the board.
 
 * Note: At least one of these two properties should be set in the DT binding.
 
-- 
cgit v1.2.3-59-g8ed1b


From 6a6dec83e5abbc5003dac234970b51afa142defd Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Tue, 7 Oct 2014 12:29:11 -0700
Subject: ASoC: imx-wm8962: Add indentation for binding doc to increase
 readability

This patch simply adds indentations for DT binding doc to increase readability
without changing any contents.

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/imx-audio-wm8962.txt | 45 +++++++++++++---------
 1 file changed, 26 insertions(+), 19 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
index f49450a87890..acea71bee34f 100644
--- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt
@@ -1,25 +1,32 @@
 Freescale i.MX audio complex with WM8962 codec
 
 Required properties:
-- compatible : "fsl,imx-audio-wm8962"
-- model : The user-visible name of this sound complex
-- ssi-controller : The phandle of the i.MX SSI controller
-- audio-codec : The phandle of the WM8962 audio codec
-- audio-routing : A list of the connections between audio components.
-  Each entry is a pair of strings, the first being the connection's sink,
-  the second being the connection's source. Valid names could be power
-  supplies, WM8962 pins, and the jacks on the board:
-
-  Power supplies:
-   * Mic Bias
-
-  Board connectors:
-   * Mic Jack
-   * Headphone Jack
-   * Ext Spk
-
-- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX)
-- mux-ext-port : The external port of the i.MX audio muxer
+
+  - compatible		: "fsl,imx-audio-wm8962"
+
+  - model		: The user-visible name of this sound complex
+
+  - ssi-controller	: The phandle of the i.MX SSI controller
+
+  - audio-codec		: The phandle of the WM8962 audio codec
+
+  - audio-routing	: A list of the connections between audio components.
+			  Each entry is a pair of strings, the first being the
+			  connection's sink, the second being the connection's
+			  source. Valid names could be power supplies, WM8962
+			  pins, and the jacks on the board:
+
+			  Power supplies:
+			   * Mic Bias
+
+			  Board connectors:
+			   * Mic Jack
+			   * Headphone Jack
+			   * Ext Spk
+
+  - mux-int-port	: The internal port of the i.MX audio muxer (AUDMUX)
+
+  - mux-ext-port	: The external port of the i.MX audio muxer
 
 Note: The AUDMUX port numbering should start at 1, which is consistent with
 hardware manual.
-- 
cgit v1.2.3-59-g8ed1b


From 40eb90a18e93fbd4fd0e6892b31241356c3c8e43 Mon Sep 17 00:00:00 2001
From: Anatol Pomozov <anatol.pomozov@gmail.com>
Date: Fri, 10 Oct 2014 20:46:36 -0700
Subject: ASoC: rt5677: Add option to configure gpio as
 floating/pullup/pulldown

gpio_config is array of 6 elements that allows to set GPIO as
floating, pullup, pulldown.

Sponsored: Google ChromeOS
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt5677.txt |  7 ++++
 include/sound/rt5677.h                             |  3 ++
 sound/soc/codecs/rt5677.c                          | 39 ++++++++++++++++++++++
 3 files changed, 49 insertions(+)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
index 0701b834fc73..f82f0e906cd9 100644
--- a/Documentation/devicetree/bindings/sound/rt5677.txt
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -27,6 +27,12 @@ Optional properties:
   Boolean. Indicate MIC1/2 input and LOUT1/2/3 outputs are differential,
   rather than single-ended.
 
+- realtek,gpio-config
+  Array of six 8bit elements that configures GPIO.
+    0 - floating (reset value)
+    1 - pull down
+    2 - pull up
+
 Pins on the device (for linking into audio routes):
 
   * IN1P
@@ -56,4 +62,5 @@ rt5677 {
 	realtek,pow-ldo2-gpio =
 		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
 	realtek,in1-differential = "true";
+	realtek,gpio-config = /bits/ 8  <0 0 0 0 0 2>;   /* pull up GPIO6 */
 };
diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h
index 082670e3a353..a56b429a1dbc 100644
--- a/include/sound/rt5677.h
+++ b/include/sound/rt5677.h
@@ -27,6 +27,9 @@ struct rt5677_platform_data {
 	bool lout3_diff;
 	/* DMIC2 clock source selection */
 	enum rt5677_dmic2_clk dmic2_clk_pin;
+
+	/* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */
+	u8 gpio_config[6];
 };
 
 #endif
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 16aa4d99a713..a454df39b7a5 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3309,6 +3309,38 @@ static int rt5677_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 	return 0;
 }
 
+/** Configures the gpio as
+ *   0 - floating
+ *   1 - pull down
+ *   2 - pull up
+ */
+static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset,
+		int value)
+{
+	int shift;
+
+	switch (offset) {
+	case RT5677_GPIO1 ... RT5677_GPIO2:
+		shift = 2 * (1 - offset);
+		regmap_update_bits(rt5677->regmap,
+			RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL2,
+			0x3 << shift,
+			(value & 0x3) << shift);
+		break;
+
+	case RT5677_GPIO3 ... RT5677_GPIO6:
+		shift = 2 * (9 - offset);
+		regmap_update_bits(rt5677->regmap,
+			RT5677_PR_BASE + RT5677_DIG_IN_PIN_ST_CTRL3,
+			0x3 << shift,
+			(value & 0x3) << shift);
+		break;
+
+	default:
+		break;
+	}
+}
+
 static struct gpio_chip rt5677_template_chip = {
 	.label			= "rt5677",
 	.owner			= THIS_MODULE,
@@ -3353,6 +3385,7 @@ static void rt5677_free_gpio(struct i2c_client *i2c)
 static int rt5677_probe(struct snd_soc_codec *codec)
 {
 	struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec);
+	int i;
 
 	rt5677->codec = codec;
 
@@ -3371,6 +3404,9 @@ static int rt5677_probe(struct snd_soc_codec *codec)
 	regmap_write(rt5677->regmap, RT5677_DIG_MISC, 0x0020);
 	regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x0c00);
 
+	for (i = 0; i < RT5677_GPIO_NUM; i++)
+		rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]);
+
 	return 0;
 }
 
@@ -3590,6 +3626,9 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np)
 			(rt5677->pow_ldo2 != -ENOENT))
 		return rt5677->pow_ldo2;
 
+	of_property_read_u8_array(np, "realtek,gpio-config",
+		rt5677->pdata.gpio_config, RT5677_GPIO_NUM);
+
 	return 0;
 }
 
-- 
cgit v1.2.3-59-g8ed1b


From bd0593f5f6add279257334b4a76aecd3ee8d31dc Mon Sep 17 00:00:00 2001
From: Jean-Michel Hautbois <jean-michel.hautbois@vodalys.com>
Date: Tue, 14 Oct 2014 08:43:11 +0200
Subject: ASoC: sgtl5000: Add MicBias resistor support in DT

Some systems may require a different resistor than the default one (4K).
This adds a property in sgtl5000 codec.
It keeps the default of 4K when nothing is specified so it does not break
existing code.

Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@vodalys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/sgtl5000.txt         |  8 ++++
 sound/soc/codecs/sgtl5000.c                        | 55 ++++++++++++++++++++--
 2 files changed, 59 insertions(+), 4 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index 955df60a118c..d6ec92707d81 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -7,10 +7,18 @@ Required properties:
 
 - clocks : the clock provider of SYS_MCLK
 
+- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs
+	The resistor can take values of 2k, 4k or 8k.
+	If set to 0 it will be off.
+	If this node is not mentioned or if the value is unknown, then
+	micbias	resistor is set to 4K.
+
+
 Example:
 
 codec: sgtl5000@0a {
 	compatible = "fsl,sgtl5000";
 	reg = <0x0a>;
 	clocks = <&clks 150>;
+	sgtl5000-micbias-resistor = <1>;
 };
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index 10160e7a9277..c417b4ad0492 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -16,6 +16,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/clk.h>
+#include <linux/log2.h>
 #include <linux/regmap.h>
 #include <linux/regulator/driver.h>
 #include <linux/regulator/machine.h>
@@ -121,6 +122,13 @@ struct ldo_regulator {
 	bool enabled;
 };
 
+enum sgtl5000_micbias_resistor {
+	SGTL5000_MICBIAS_OFF = 0,
+	SGTL5000_MICBIAS_2K = 2,
+	SGTL5000_MICBIAS_4K = 4,
+	SGTL5000_MICBIAS_8K = 8,
+};
+
 /* sgtl5000 private structure in codec */
 struct sgtl5000_priv {
 	int sysclk;	/* sysclk rate */
@@ -131,6 +139,7 @@ struct sgtl5000_priv {
 	struct regmap *regmap;
 	struct clk *mclk;
 	int revision;
+	u8 micbias_resistor;
 };
 
 /*
@@ -145,12 +154,14 @@ struct sgtl5000_priv {
 static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(w->codec);
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		/* change mic bias resistor to 4Kohm */
+		/* change mic bias resistor */
 		snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
-				SGTL5000_BIAS_R_MASK,
-				SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT);
+			SGTL5000_BIAS_R_MASK,
+			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
@@ -1327,7 +1338,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			SGTL5000_HP_ZCD_EN |
 			SGTL5000_ADC_ZCD_EN);
 
-	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2);
+	snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
+			SGTL5000_BIAS_R_MASK,
+			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 
 	/*
 	 * disable DAP
@@ -1419,6 +1432,8 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 	struct sgtl5000_priv *sgtl5000;
 	int ret, reg, rev;
 	unsigned int mclk;
+	struct device_node *np = client->dev.of_node;
+	u32 value;
 
 	sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv),
 								GFP_KERNEL);
@@ -1471,6 +1486,38 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 	dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev);
 	sgtl5000->revision = rev;
 
+	if (np) {
+		if (!of_property_read_u32(np,
+			"micbias-resistor-k-ohms", &value)) {
+			switch (value) {
+			case SGTL5000_MICBIAS_OFF:
+				sgtl5000->micbias_resistor = 0;
+				break;
+			case SGTL5000_MICBIAS_2K:
+				sgtl5000->micbias_resistor = 1;
+				break;
+			case SGTL5000_MICBIAS_4K:
+				sgtl5000->micbias_resistor = 2;
+				break;
+			case SGTL5000_MICBIAS_8K:
+				sgtl5000->micbias_resistor = 3;
+				break;
+			default:
+				sgtl5000->micbias_resistor = 2;
+				dev_err(&client->dev,
+					"Unsuitable MicBias resistor\n");
+			}
+		} else {
+			/* default is 4Kohms */
+			sgtl5000->micbias_resistor = 2;
+		}
+				dev_err(&client->dev,
+					"Unsuitable MicBias resistor\n");
+			}
+		} else {
+		}
+	}
+
 	i2c_set_clientdata(client, sgtl5000);
 
 	/* Ensure sgtl5000 will start with sane register values */
-- 
cgit v1.2.3-59-g8ed1b


From 8735779774b8bbe14456c9e6ba4525eefc67a228 Mon Sep 17 00:00:00 2001
From: Jean-Michel Hautbois <jean-michel.hautbois@vodalys.com>
Date: Tue, 14 Oct 2014 08:43:12 +0200
Subject: ASoC: sgtl5000: Add MicBias voltage support

Some systems may require to specify a bias different than default (1.25V).
This adds a property in sgtl5000 codec.
The property is specified in milli-volts so that it is coherent with datasheet.

Signed-off-by: Jean-Michel Hautbois <jean-michel.hautbois@vodalys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/sgtl5000.txt |  7 ++++++-
 sound/soc/codecs/sgtl5000.c                          | 13 +++++++++++++
 2 files changed, 19 insertions(+), 1 deletion(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt
index d6ec92707d81..1aab40339617 100644
--- a/Documentation/devicetree/bindings/sound/sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt
@@ -13,6 +13,10 @@ Required properties:
 	If this node is not mentioned or if the value is unknown, then
 	micbias	resistor is set to 4K.
 
+- micbias-voltage-m-volts : the bias voltage to be used in mVolts
+	The voltage can take values from 1.25V to 3V by 250mV steps
+	If this node is not mentionned or the value is unknown, then
+	the value is set to 1.25V.
 
 Example:
 
@@ -20,5 +24,6 @@ codec: sgtl5000@0a {
 	compatible = "fsl,sgtl5000";
 	reg = <0x0a>;
 	clocks = <&clks 150>;
-	sgtl5000-micbias-resistor = <1>;
+	micbias-resistor-k-ohms = <2>;
+	micbias-voltage-m-volts = <2250>;
 };
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index c417b4ad0492..59336f6aba80 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -140,6 +140,7 @@ struct sgtl5000_priv {
 	struct clk *mclk;
 	int revision;
 	u8 micbias_resistor;
+	u8 micbias_voltage;
 };
 
 /*
@@ -1342,6 +1343,9 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			SGTL5000_BIAS_R_MASK,
 			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 
+	snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
+			SGTL5000_BIAS_R_MASK,
+			sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
 	/*
 	 * disable DAP
 	 * TODO:
@@ -1511,10 +1515,19 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 			/* default is 4Kohms */
 			sgtl5000->micbias_resistor = 2;
 		}
+		if (!of_property_read_u32(np,
+			"micbias-voltage-m-volts", &value)) {
+			/* 1250mV => 0 */
+			/* steps of 250mV */
+			if ((value >= 1250) && (value <= 3000))
+				sgtl5000->micbias_voltage = (value / 250) - 5;
+			else {
+				sgtl5000->micbias_voltage = 0;
 				dev_err(&client->dev,
 					"Unsuitable MicBias resistor\n");
 			}
 		} else {
+			sgtl5000->micbias_voltage = 0;
 		}
 	}
 
-- 
cgit v1.2.3-59-g8ed1b


From 5e3363ad1b7b2e1f197a3f56b01e21cb155ad454 Mon Sep 17 00:00:00 2001
From: Oder Chiou <oder_chiou@realtek.com>
Date: Thu, 16 Oct 2014 11:24:26 -0700
Subject: ASoC: rt5677: add GPIO IRQ support

This allows to enable Mic Jack detection feature

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Modified-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt5677.txt |  10 ++
 include/sound/rt5677.h                             |   7 ++
 sound/soc/codecs/rt5677.c                          | 134 +++++++++++++++++++++
 sound/soc/codecs/rt5677.h                          |  49 ++++++++
 4 files changed, 200 insertions(+)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt
index f82f0e906cd9..740ff771aa8b 100644
--- a/Documentation/devicetree/bindings/sound/rt5677.txt
+++ b/Documentation/devicetree/bindings/sound/rt5677.txt
@@ -33,6 +33,15 @@ Optional properties:
     1 - pull down
     2 - pull up
 
+- realtek,jd1-gpio
+  Configures GPIO Mic Jack detection 1.
+  Select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively.
+
+- realtek,jd2-gpio
+- realtek,jd3-gpio
+  Configures GPIO Mic Jack detection 2 and 3.
+  Select 0 ~ 3 as OFF, GPIO4, GPIO5 and GPIO6 respectively.
+
 Pins on the device (for linking into audio routes):
 
   * IN1P
@@ -63,4 +72,5 @@ rt5677 {
 		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
 	realtek,in1-differential = "true";
 	realtek,gpio-config = /bits/ 8  <0 0 0 0 0 2>;   /* pull up GPIO6 */
+	realtek,jd2-gpio = <3>;  /* Enables Jack detection for GPIO6 */
 };
diff --git a/include/sound/rt5677.h b/include/sound/rt5677.h
index a56b429a1dbc..d9eb7d861cd0 100644
--- a/include/sound/rt5677.h
+++ b/include/sound/rt5677.h
@@ -30,6 +30,13 @@ struct rt5677_platform_data {
 
 	/* configures GPIO, 0 - floating, 1 - pulldown, 2 - pullup */
 	u8 gpio_config[6];
+
+	/* jd1 can select 0 ~ 3 as OFF, GPIO1, GPIO2 and GPIO3 respectively */
+	unsigned int jd1_gpio;
+	/* jd2 and jd3 can select 0 ~ 3 as
+		OFF, GPIO4, GPIO5 and GPIO6 respectively */
+	unsigned int jd2_gpio;
+	unsigned int jd3_gpio;
 };
 
 #endif
diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index d17d079fdcf3..6c73dfd22a0c 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -3614,6 +3614,46 @@ static void rt5677_gpio_config(struct rt5677_priv *rt5677, unsigned offset,
 	}
 }
 
+static int rt5677_to_irq(struct gpio_chip *chip, unsigned offset)
+{
+	struct rt5677_priv *rt5677 = gpio_to_rt5677(chip);
+	struct regmap_irq_chip_data *data = rt5677->irq_data;
+	int irq;
+
+	if (offset >= RT5677_GPIO1 && offset <= RT5677_GPIO3) {
+		if ((rt5677->pdata.jd1_gpio == 1 && offset == RT5677_GPIO1) ||
+			(rt5677->pdata.jd1_gpio == 2 &&
+				offset == RT5677_GPIO2) ||
+			(rt5677->pdata.jd1_gpio == 3 &&
+				offset == RT5677_GPIO3)) {
+			irq = RT5677_IRQ_JD1;
+		} else {
+			return -ENXIO;
+		}
+	}
+
+	if (offset >= RT5677_GPIO4 && offset <= RT5677_GPIO6) {
+		if ((rt5677->pdata.jd2_gpio == 1 && offset == RT5677_GPIO4) ||
+			(rt5677->pdata.jd2_gpio == 2 &&
+				offset == RT5677_GPIO5) ||
+			(rt5677->pdata.jd2_gpio == 3 &&
+				offset == RT5677_GPIO6)) {
+			irq = RT5677_IRQ_JD2;
+		} else if ((rt5677->pdata.jd3_gpio == 1 &&
+				offset == RT5677_GPIO4) ||
+			(rt5677->pdata.jd3_gpio == 2 &&
+				offset == RT5677_GPIO5) ||
+			(rt5677->pdata.jd3_gpio == 3 &&
+				offset == RT5677_GPIO6)) {
+			irq = RT5677_IRQ_JD3;
+		} else {
+			return -ENXIO;
+		}
+	}
+
+	return regmap_irq_get_virq(data, irq);
+}
+
 static struct gpio_chip rt5677_template_chip = {
 	.label			= "rt5677",
 	.owner			= THIS_MODULE,
@@ -3621,6 +3661,7 @@ static struct gpio_chip rt5677_template_chip = {
 	.set			= rt5677_gpio_set,
 	.direction_input	= rt5677_gpio_direction_in,
 	.get			= rt5677_gpio_get,
+	.to_irq			= rt5677_to_irq,
 	.can_sleep		= 1,
 };
 
@@ -3685,6 +3726,31 @@ static int rt5677_probe(struct snd_soc_codec *codec)
 	for (i = 0; i < RT5677_GPIO_NUM; i++)
 		rt5677_gpio_config(rt5677, i, rt5677->pdata.gpio_config[i]);
 
+	if (rt5677->irq_data) {
+		regmap_update_bits(rt5677->regmap, RT5677_GPIO_CTRL1, 0x8000,
+			0x8000);
+		regmap_update_bits(rt5677->regmap, RT5677_DIG_MISC, 0x0018,
+			0x0008);
+
+		if (rt5677->pdata.jd1_gpio)
+			regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+				RT5677_SEL_GPIO_JD1_MASK,
+				rt5677->pdata.jd1_gpio <<
+				RT5677_SEL_GPIO_JD1_SFT);
+
+		if (rt5677->pdata.jd2_gpio)
+			regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+				RT5677_SEL_GPIO_JD2_MASK,
+				rt5677->pdata.jd2_gpio <<
+				RT5677_SEL_GPIO_JD2_SFT);
+
+		if (rt5677->pdata.jd3_gpio)
+			regmap_update_bits(rt5677->regmap, RT5677_JD_CTRL1,
+				RT5677_SEL_GPIO_JD3_MASK,
+				rt5677->pdata.jd3_gpio <<
+				RT5677_SEL_GPIO_JD3_SFT);
+	}
+
 	mutex_init(&rt5677->dsp_cmd_lock);
 
 	return 0;
@@ -3915,9 +3981,74 @@ static int rt5677_parse_dt(struct rt5677_priv *rt5677, struct device_node *np)
 	of_property_read_u8_array(np, "realtek,gpio-config",
 		rt5677->pdata.gpio_config, RT5677_GPIO_NUM);
 
+	of_property_read_u32(np, "realtek,jd1-gpio", &rt5677->pdata.jd1_gpio);
+	of_property_read_u32(np, "realtek,jd2-gpio", &rt5677->pdata.jd2_gpio);
+	of_property_read_u32(np, "realtek,jd3-gpio", &rt5677->pdata.jd3_gpio);
+
 	return 0;
 }
 
+static struct regmap_irq rt5677_irqs[] = {
+	[RT5677_IRQ_JD1] = {
+		.reg_offset = 0,
+		.mask = RT5677_EN_IRQ_GPIO_JD1,
+	},
+	[RT5677_IRQ_JD2] = {
+		.reg_offset = 0,
+		.mask = RT5677_EN_IRQ_GPIO_JD2,
+	},
+	[RT5677_IRQ_JD3] = {
+		.reg_offset = 0,
+		.mask = RT5677_EN_IRQ_GPIO_JD3,
+	},
+};
+
+static struct regmap_irq_chip rt5677_irq_chip = {
+	.name = "rt5677",
+	.irqs = rt5677_irqs,
+	.num_irqs = ARRAY_SIZE(rt5677_irqs),
+
+	.num_regs = 1,
+	.status_base = RT5677_IRQ_CTRL1,
+	.mask_base = RT5677_IRQ_CTRL1,
+	.mask_invert = 1,
+};
+
+int rt5677_irq_init(struct i2c_client *i2c)
+{
+	int ret;
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+
+	if (!rt5677->pdata.jd1_gpio &&
+		!rt5677->pdata.jd2_gpio &&
+		!rt5677->pdata.jd3_gpio)
+		return 0;
+
+	if (!i2c->irq) {
+		dev_err(&i2c->dev, "No interrupt specified\n");
+		return -EINVAL;
+	}
+
+	ret = regmap_add_irq_chip(rt5677->regmap, i2c->irq,
+		IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, 0,
+		&rt5677_irq_chip, &rt5677->irq_data);
+
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register IRQ chip: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+void rt5677_irq_exit(struct i2c_client *i2c)
+{
+	struct rt5677_priv *rt5677 = i2c_get_clientdata(i2c);
+
+	if (rt5677->irq_data)
+		regmap_del_irq_chip(i2c->irq, rt5677->irq_data);
+}
+
 static int rt5677_i2c_probe(struct i2c_client *i2c,
 		    const struct i2c_device_id *id)
 {
@@ -4015,6 +4146,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 	}
 
 	rt5677_init_gpio(i2c);
+	rt5677_irq_init(i2c);
 
 	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5677,
 				      rt5677_dai, ARRAY_SIZE(rt5677_dai));
@@ -4022,6 +4154,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c,
 
 static int rt5677_i2c_remove(struct i2c_client *i2c)
 {
+	rt5677_irq_exit(i2c);
+
 	snd_soc_unregister_codec(&i2c->dev);
 	rt5677_free_gpio(i2c);
 
diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h
index 20efa4a4c82c..d2c743c255a1 100644
--- a/sound/soc/codecs/rt5677.h
+++ b/sound/soc/codecs/rt5677.h
@@ -1368,6 +1368,48 @@
 #define RT5677_SEL_SRC_IB01			(0x1 << 0)
 #define RT5677_SEL_SRC_IB01_SFT			0
 
+/* Jack Detect Control 1 (0xb5) */
+#define RT5677_SEL_GPIO_JD1_MASK		(0x3 << 14)
+#define RT5677_SEL_GPIO_JD1_SFT			14
+#define RT5677_SEL_GPIO_JD2_MASK		(0x3 << 12)
+#define RT5677_SEL_GPIO_JD2_SFT			12
+#define RT5677_SEL_GPIO_JD3_MASK		(0x3 << 10)
+#define RT5677_SEL_GPIO_JD3_SFT			10
+
+/* IRQ Control 1 (0xbd) */
+#define RT5677_STA_GPIO_JD1			(0x1 << 15)
+#define RT5677_STA_GPIO_JD1_SFT			15
+#define RT5677_EN_IRQ_GPIO_JD1			(0x1 << 14)
+#define RT5677_EN_IRQ_GPIO_JD1_SFT		14
+#define RT5677_EN_GPIO_JD1_STICKY		(0x1 << 13)
+#define RT5677_EN_GPIO_JD1_STICKY_SFT		13
+#define RT5677_INV_GPIO_JD1			(0x1 << 12)
+#define RT5677_INV_GPIO_JD1_SFT			12
+#define RT5677_STA_GPIO_JD2			(0x1 << 11)
+#define RT5677_STA_GPIO_JD2_SFT			11
+#define RT5677_EN_IRQ_GPIO_JD2			(0x1 << 10)
+#define RT5677_EN_IRQ_GPIO_JD2_SFT		10
+#define RT5677_EN_GPIO_JD2_STICKY		(0x1 << 9)
+#define RT5677_EN_GPIO_JD2_STICKY_SFT		9
+#define RT5677_INV_GPIO_JD2			(0x1 << 8)
+#define RT5677_INV_GPIO_JD2_SFT			8
+#define RT5677_STA_MICBIAS1_OVCD		(0x1 << 7)
+#define RT5677_STA_MICBIAS1_OVCD_SFT		7
+#define RT5677_EN_IRQ_MICBIAS1_OVCD		(0x1 << 6)
+#define RT5677_EN_IRQ_MICBIAS1_OVCD_SFT		6
+#define RT5677_EN_MICBIAS1_OVCD_STICKY		(0x1 << 5)
+#define RT5677_EN_MICBIAS1_OVCD_STICKY_SFT	5
+#define RT5677_INV_MICBIAS1_OVCD		(0x1 << 4)
+#define RT5677_INV_MICBIAS1_OVCD_SFT		4
+#define RT5677_STA_GPIO_JD3			(0x1 << 3)
+#define RT5677_STA_GPIO_JD3_SFT			3
+#define RT5677_EN_IRQ_GPIO_JD3			(0x1 << 2)
+#define RT5677_EN_IRQ_GPIO_JD3_SFT		2
+#define RT5677_EN_GPIO_JD3_STICKY		(0x1 << 1)
+#define RT5677_EN_GPIO_JD3_STICKY_SFT		1
+#define RT5677_INV_GPIO_JD3			(0x1 << 0)
+#define RT5677_INV_GPIO_JD3_SFT			0
+
 /* GPIO status (0xbf) */
 #define RT5677_GPIO6_STATUS_MASK		(0x1 << 5)
 #define RT5677_GPIO6_STATUS_SFT			5
@@ -1545,6 +1587,12 @@ enum {
 	RT5677_GPIO_NUM,
 };
 
+enum {
+	RT5677_IRQ_JD1,
+	RT5677_IRQ_JD2,
+	RT5677_IRQ_JD3,
+};
+
 struct rt5677_priv {
 	struct snd_soc_codec *codec;
 	struct rt5677_platform_data pdata;
@@ -1565,6 +1613,7 @@ struct rt5677_priv {
 	struct gpio_chip gpio_chip;
 #endif
 	bool dsp_vad_en;
+	struct regmap_irq_chip_data *irq_data;
 };
 
 #endif /* __RT5677_H__ */
-- 
cgit v1.2.3-59-g8ed1b


From defcd98b16461e123cb4a6cb6ef24a1d0085c1b2 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Mon, 3 Nov 2014 10:28:57 -0800
Subject: ASoC: max98090: Different comp tables for different pclks

In addtion expand the table to handle other values of sysclk.  Instead
of making the table 3D, expand it to a more descriptive struct.  The
divisors are specified in Table 19 of the 98090 data sheet version
0p94.

The dmic frequency was previously assumed.  Instead make it explicit
and configurable through device tree.  This now handles independently
set pclk and dmic frequency.

Based on downstream work by Ralph Birt.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/max98090.txt         |   2 +
 sound/soc/codecs/max98090.c                        | 189 +++++++++++++++++----
 sound/soc/codecs/max98090.h                        |   8 +
 3 files changed, 163 insertions(+), 36 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt
index c454e67f54bb..aa802a274520 100644
--- a/Documentation/devicetree/bindings/sound/max98090.txt
+++ b/Documentation/devicetree/bindings/sound/max98090.txt
@@ -16,6 +16,8 @@ Optional properties:
 
 - clock-names: Should be "mclk"
 
+- maxim,dmic-freq: Frequency at which to clock DMIC
+
 Pins on the device (for linking into audio routes):
 
   * MIC1
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 1229554f1464..a65861cf0a44 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1826,27 +1826,155 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-static const int comp_pclk_rates[] = {
-	11289600, 12288000, 12000000, 13000000, 19200000
-};
-
-static const int dmic_micclk[] = {
-	2, 2, 2, 2, 4, 2
-};
+static const int dmic_divisors[] = { 2, 3, 4, 5, 6, 8 };
 
 static const int comp_lrclk_rates[] = {
 	8000, 16000, 32000, 44100, 48000, 96000
 };
 
-static const int dmic_comp[6][6] = {
-	{7, 8, 3, 3, 3, 3},
-	{7, 8, 3, 3, 3, 3},
-	{7, 8, 3, 3, 3, 3},
-	{7, 8, 3, 1, 1, 1},
-	{7, 8, 3, 1, 2, 2},
-	{7, 8, 3, 3, 3, 3}
+struct dmic_table {
+	int pclk;
+	struct {
+		int freq;
+		int comp[6]; /* One each for 8, 16, 32, 44.1, 48, and 96 kHz */
+	} settings[6]; /* One for each dmic divisor. */
 };
 
+static const struct dmic_table dmic_table[] = { /* One for each pclk freq. */
+	{
+		.pclk = 11289600,
+		.settings = {
+			{ .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+		},
+	},
+	{
+		.pclk = 12000000,
+		.settings = {
+			{ .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+		}
+	},
+	{
+		.pclk = 12288000,
+		.settings = {
+			{ .freq = 2, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 1, .comp = { 7, 8, 2, 2, 2, 2 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 6, 6, 6, 6 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 3, 3, 3, 3 } },
+		}
+	},
+	{
+		.pclk = 13000000,
+		.settings = {
+			{ .freq = 2, .comp = { 7, 8, 1, 1, 1, 1 } },
+			{ .freq = 1, .comp = { 7, 8, 0, 0, 0, 0 } },
+			{ .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+			{ .freq = 0, .comp = { 7, 8, 4, 4, 5, 5 } },
+			{ .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+			{ .freq = 0, .comp = { 7, 8, 1, 1, 1, 1 } },
+		}
+	},
+	{
+		.pclk = 19200000,
+		.settings = {
+			{ .freq = 2, .comp = { 0, 0, 0, 0, 0, 0 } },
+			{ .freq = 1, .comp = { 7, 8, 1, 1, 1, 1 } },
+			{ .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+			{ .freq = 0, .comp = { 7, 8, 2, 2, 3, 3 } },
+			{ .freq = 0, .comp = { 7, 8, 1, 1, 2, 2 } },
+			{ .freq = 0, .comp = { 7, 8, 5, 5, 6, 6 } },
+		}
+	},
+};
+
+static int max98090_find_divisor(int target_freq, int pclk)
+{
+	int current_diff = INT_MAX;
+	int test_diff = INT_MAX;
+	int divisor_index = 0;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(dmic_divisors); i++) {
+		test_diff = abs(target_freq - (pclk / dmic_divisors[i]));
+		if (test_diff < current_diff) {
+			current_diff = test_diff;
+			divisor_index = i;
+		}
+	}
+
+	return divisor_index;
+}
+
+static int max98090_find_closest_pclk(int pclk)
+{
+	int m1;
+	int m2;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(dmic_table); i++) {
+		if (pclk == dmic_table[i].pclk)
+			return i;
+		if (pclk < dmic_table[i].pclk) {
+			if (i == 0)
+				return i;
+			m1 = pclk - dmic_table[i-1].pclk;
+			m2 = dmic_table[i].pclk - pclk;
+			if (m1 < m2)
+				return i - 1;
+			else
+				return i;
+		}
+	}
+
+	return -EINVAL;
+}
+
+static int max98090_configure_dmic(struct max98090_priv *max98090,
+				   int target_dmic_clk, int pclk, int fs)
+{
+	int micclk_index;
+	int pclk_index;
+	int dmic_freq;
+	int dmic_comp;
+	int i;
+
+	pclk_index = max98090_find_closest_pclk(pclk);
+	if (pclk_index < 0)
+		return pclk_index;
+
+	micclk_index = max98090_find_divisor(target_dmic_clk, pclk);
+
+	for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) {
+		if (fs <= (comp_lrclk_rates[i] + comp_lrclk_rates[i+1]) / 2)
+			break;
+	}
+
+	dmic_freq = dmic_table[pclk_index].settings[micclk_index].freq;
+	dmic_comp = dmic_table[pclk_index].settings[micclk_index].comp[i];
+
+	regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_ENABLE,
+			   M98090_MICCLK_MASK,
+			   micclk_index << M98090_MICCLK_SHIFT);
+
+	regmap_update_bits(max98090->regmap, M98090_REG_DIGITAL_MIC_CONFIG,
+			   M98090_DMIC_COMP_MASK | M98090_DMIC_FREQ_MASK,
+			   dmic_comp << M98090_DMIC_COMP_SHIFT |
+			   dmic_freq << M98090_DMIC_FREQ_SHIFT);
+
+	return 0;
+}
+
 static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params,
 				   struct snd_soc_dai *dai)
@@ -1854,7 +1982,6 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
 	struct max98090_cdata *cdata;
-	int i, j;
 
 	cdata = &max98090->dai[0];
 	max98090->bclk = snd_soc_params_to_bclk(params);
@@ -1893,27 +2020,8 @@ static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 		snd_soc_update_bits(codec, M98090_REG_FILTER_CONFIG,
 			M98090_DHF_MASK, M98090_DHF_MASK);
 
-	/* Check for supported PCLK to LRCLK ratios */
-	for (j = 0; j < ARRAY_SIZE(comp_pclk_rates); j++) {
-		if (comp_pclk_rates[j] == max98090->sysclk) {
-			break;
-		}
-	}
-
-	for (i = 0; i < ARRAY_SIZE(comp_lrclk_rates) - 1; i++) {
-		if (max98090->lrclk <= (comp_lrclk_rates[i] +
-			comp_lrclk_rates[i + 1]) / 2) {
-			break;
-		}
-	}
-
-	snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_ENABLE,
-			M98090_MICCLK_MASK,
-			dmic_micclk[j] << M98090_MICCLK_SHIFT);
-
-	snd_soc_update_bits(codec, M98090_REG_DIGITAL_MIC_CONFIG,
-			M98090_DMIC_COMP_MASK,
-			dmic_comp[j][i] << M98090_DMIC_COMP_SHIFT);
+	max98090_configure_dmic(max98090, max98090->dmic_freq, max98090->pclk,
+				max98090->lrclk);
 
 	return 0;
 }
@@ -1944,12 +2052,15 @@ static int max98090_dai_set_sysclk(struct snd_soc_dai *dai,
 	if ((freq >= 10000000) && (freq <= 20000000)) {
 		snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
 			M98090_PSCLK_DIV1);
+		max98090->pclk = freq;
 	} else if ((freq > 20000000) && (freq <= 40000000)) {
 		snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
 			M98090_PSCLK_DIV2);
+		max98090->pclk = freq >> 1;
 	} else if ((freq > 40000000) && (freq <= 60000000)) {
 		snd_soc_write(codec, M98090_REG_SYSTEM_CLOCK,
 			M98090_PSCLK_DIV4);
+		max98090->pclk = freq >> 2;
 	} else {
 		dev_err(codec->dev, "Invalid master clock frequency\n");
 		return -EINVAL;
@@ -2324,6 +2435,7 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	/* Initialize private data */
 
 	max98090->sysclk = (unsigned)-1;
+	max98090->pclk = (unsigned)-1;
 	max98090->master = false;
 
 	cdata = &max98090->dai[0];
@@ -2463,6 +2575,11 @@ static int max98090_i2c_probe(struct i2c_client *i2c,
 	i2c_set_clientdata(i2c, max98090);
 	max98090->pdata = i2c->dev.platform_data;
 
+	ret = of_property_read_u32(i2c->dev.of_node, "maxim,dmic-freq",
+				   &max98090->dmic_freq);
+	if (ret < 0)
+		max98090->dmic_freq = MAX98090_DEFAULT_DMIC_FREQ;
+
 	max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap);
 	if (IS_ERR(max98090->regmap)) {
 		ret = PTR_ERR(max98090->regmap);
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index a5f6bada06da..21ff743f5af2 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -11,6 +11,12 @@
 #ifndef _MAX98090_H
 #define _MAX98090_H
 
+/*
+ * The default operating frequency for a DMIC attached to the codec.
+ * This can be overridden by a device tree property.
+ */
+#define MAX98090_DEFAULT_DMIC_FREQ		2500000
+
 /*
  * MAX98090 Register Definitions
  */
@@ -1518,8 +1524,10 @@ struct max98090_priv {
 	struct max98090_pdata *pdata;
 	struct clk *mclk;
 	unsigned int sysclk;
+	unsigned int pclk;
 	unsigned int bclk;
 	unsigned int lrclk;
+	u32 dmic_freq;
 	struct max98090_cdata dai[1];
 	int jack_state;
 	struct delayed_work jack_work;
-- 
cgit v1.2.3-59-g8ed1b


From a5a56871f804edac93a53b5e871c0e9818fb9033 Mon Sep 17 00:00:00 2001
From: Padmavathi Venna <padma.v@samsung.com>
Date: Fri, 7 Nov 2014 12:24:40 +0530
Subject: ASoC: samsung: add support for exynos7 I2S controller

Exynos7 I2S controller has no internal dma, supports more
no. of root clock sampling frequencies and has more no.of Rx
fifos to support 7.1CH recording in TDM mode. Due to more no.
of root clock frequency values some of the bit offsets got
shifted up by one. Also I2S1 on previous Samsung platforms
uses v3 dai type but on Exynos7 it is upgraded to v5 with
slightly modified register offsets for supporting more no.of
RFS values. Due to the above changes, the driver has to be
modified to handle all versions of I2S controller. For this
I introduced a new structure to hold modified bit offsets and
masks which is passed as dai data.

Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/samsung-i2s.txt      |  15 +-
 sound/soc/samsung/Kconfig                          |   2 +-
 sound/soc/samsung/i2s-regs.h                       |  10 +-
 sound/soc/samsung/i2s.c                            | 218 +++++++++++++++------
 4 files changed, 174 insertions(+), 71 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
index 7386d444ada1..d188296bb6ec 100644
--- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt
+++ b/Documentation/devicetree/bindings/sound/samsung-i2s.txt
@@ -6,10 +6,17 @@ Required SoC Specific Properties:
    - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S.
    - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with
      secondary fifo, s/w reset control and internal mux for root clk src.
-   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(7.1) I2S with
-     secondary fifo, s/w reset control, internal mux for root clk src and
-     TDM support. TDM (Time division multiplexing) is to allow transfer of
-     multiple channel audio data on single data line.
+   - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for
+     playback, sterio channel capture, secondary fifo using internal
+     or external dma, s/w reset control, internal mux for root clk src
+     and 7.1 channel TDM support for playback. TDM (Time division multiplexing)
+     is to allow transfer of multiple channel audio data on single data line.
+   - samsung,exynos7-i2s: with all the available features of exynos5 i2s,
+     exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo
+     with only external dma and more no.of root clk sampling frequencies.
+   - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports
+     stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with
+     slightly modified bit offsets.
 
 - reg: physical base address of the controller and length of memory mapped
   region.
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 55a38697443d..e0e737faadd9 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -1,6 +1,6 @@
 config SND_SOC_SAMSUNG
 	tristate "ASoC support for Samsung"
-	depends on PLAT_SAMSUNG
+	depends on (PLAT_SAMSUNG || ARCH_EXYNOS)
 	depends on S3C64XX_PL080 || !ARCH_S3C64XX
 	depends on S3C24XX_DMAC || !ARCH_S3C24XX
 	select SND_SOC_GENERIC_DMAENGINE_PCM
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
index 821a50231002..9170c311d66e 100644
--- a/sound/soc/samsung/i2s-regs.h
+++ b/sound/soc/samsung/i2s-regs.h
@@ -33,8 +33,9 @@
 #define I2SLVL3ADDR	0x3c
 #define I2SSTR1		0x40
 #define I2SVER		0x44
-#define I2SFIC2		0x48
+#define I2SFIC1		0x48
 #define I2STDM		0x4c
+#define I2SFSTA		0x50
 
 #define CON_RSTCLR		(1 << 31)
 #define CON_FRXOFSTATUS		(1 << 26)
@@ -93,8 +94,6 @@
 #define MOD_BLC_24BIT		(2 << 13)
 #define MOD_BLC_MASK		(3 << 13)
 
-#define MOD_IMS_SYSMUX		(1 << 10)
-#define MOD_SLAVE		(1 << 11)
 #define MOD_TXONLY		(0 << 8)
 #define MOD_RXONLY		(1 << 8)
 #define MOD_TXRX		(2 << 8)
@@ -132,7 +131,10 @@
 #define EXYNOS5420_MOD_BCLK_256FS	8
 #define EXYNOS5420_MOD_BCLK_MASK	0xf
 
-#define MOD_CDCLKCON		(1 << 12)
+#define EXYNOS7_MOD_RCLK_64FS	4
+#define EXYNOS7_MOD_RCLK_128FS	5
+#define EXYNOS7_MOD_RCLK_96FS	6
+#define EXYNOS7_MOD_RCLK_192FS	7
 
 #define PSR_PSREN		(1 << 15)
 
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 38b9a524cc9f..947352d00ddf 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -36,9 +36,24 @@ enum samsung_dai_type {
 	TYPE_SEC,
 };
 
+struct samsung_i2s_variant_regs {
+	unsigned int	bfs_off;
+	unsigned int	rfs_off;
+	unsigned int	sdf_off;
+	unsigned int	txr_off;
+	unsigned int	rclksrc_off;
+	unsigned int	mss_off;
+	unsigned int	cdclkcon_off;
+	unsigned int	lrp_off;
+	unsigned int	bfs_mask;
+	unsigned int	rfs_mask;
+	unsigned int	ftx0cnt_off;
+};
+
 struct samsung_i2s_dai_data {
 	int dai_type;
 	u32 quirks;
+	const struct samsung_i2s_variant_regs *i2s_variant_regs;
 };
 
 struct i2s_dai {
@@ -81,6 +96,7 @@ struct i2s_dai {
 	u32	suspend_i2scon;
 	u32	suspend_i2spsr;
 	unsigned long gpios[7];	/* i2s gpio line numbers */
+	const struct samsung_i2s_variant_regs *variant_regs;
 };
 
 /* Lock for cross i/f checks */
@@ -95,7 +111,8 @@ static inline bool is_secondary(struct i2s_dai *i2s)
 /* If operating in SoC-Slave mode */
 static inline bool is_slave(struct i2s_dai *i2s)
 {
-	return (readl(i2s->addr + I2SMOD) & MOD_SLAVE) ? true : false;
+	u32 mod = readl(i2s->addr + I2SMOD);
+	return (mod & (1 << i2s->variant_regs->mss_off)) ? true : false;
 }
 
 /* If this interface of the controller is transmitting data */
@@ -200,14 +217,14 @@ static inline bool is_manager(struct i2s_dai *i2s)
 static inline unsigned get_rfs(struct i2s_dai *i2s)
 {
 	u32 rfs;
-
-	if (i2s->quirks & QUIRK_SUPPORTS_TDM)
-		rfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_RCLK_SHIFT;
-	else
-		rfs = (readl(i2s->addr + I2SMOD) >> MOD_RCLK_SHIFT);
-	rfs &= MOD_RCLK_MASK;
+	rfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->rfs_off;
+	rfs &= i2s->variant_regs->rfs_mask;
 
 	switch (rfs) {
+	case 7: return 192;
+	case 6: return 96;
+	case 5: return 128;
+	case 4: return 64;
 	case 3:	return 768;
 	case 2: return 384;
 	case 1:	return 512;
@@ -219,15 +236,23 @@ static inline unsigned get_rfs(struct i2s_dai *i2s)
 static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs)
 {
 	u32 mod = readl(i2s->addr + I2SMOD);
-	int rfs_shift;
+	int rfs_shift = i2s->variant_regs->rfs_off;
 
-	if (i2s->quirks & QUIRK_SUPPORTS_TDM)
-		rfs_shift = EXYNOS5420_MOD_RCLK_SHIFT;
-	else
-		rfs_shift = MOD_RCLK_SHIFT;
-	mod &= ~(MOD_RCLK_MASK << rfs_shift);
+	mod &= ~(i2s->variant_regs->rfs_mask << rfs_shift);
 
 	switch (rfs) {
+	case 192:
+		mod |= (EXYNOS7_MOD_RCLK_192FS << rfs_shift);
+		break;
+	case 96:
+		mod |= (EXYNOS7_MOD_RCLK_96FS << rfs_shift);
+		break;
+	case 128:
+		mod |= (EXYNOS7_MOD_RCLK_128FS << rfs_shift);
+		break;
+	case 64:
+		mod |= (EXYNOS7_MOD_RCLK_64FS << rfs_shift);
+		break;
 	case 768:
 		mod |= (MOD_RCLK_768FS << rfs_shift);
 		break;
@@ -249,14 +274,8 @@ static inline void set_rfs(struct i2s_dai *i2s, unsigned rfs)
 static inline unsigned get_bfs(struct i2s_dai *i2s)
 {
 	u32 bfs;
-
-	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
-		bfs = readl(i2s->addr + I2SMOD) >> EXYNOS5420_MOD_BCLK_SHIFT;
-		bfs &= EXYNOS5420_MOD_BCLK_MASK;
-	} else {
-		bfs =  readl(i2s->addr + I2SMOD) >> MOD_BCLK_SHIFT;
-		bfs &= MOD_BCLK_MASK;
-	}
+	bfs = readl(i2s->addr + I2SMOD) >> i2s->variant_regs->bfs_off;
+	bfs &= i2s->variant_regs->bfs_mask;
 
 	switch (bfs) {
 	case 8: return 256;
@@ -275,16 +294,8 @@ static inline unsigned get_bfs(struct i2s_dai *i2s)
 static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs)
 {
 	u32 mod = readl(i2s->addr + I2SMOD);
-	int bfs_shift;
 	int tdm = i2s->quirks & QUIRK_SUPPORTS_TDM;
-
-	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
-		bfs_shift = EXYNOS5420_MOD_BCLK_SHIFT;
-		mod &= ~(EXYNOS5420_MOD_BCLK_MASK << bfs_shift);
-	} else {
-		bfs_shift = MOD_BCLK_SHIFT;
-		mod &= ~(MOD_BCLK_MASK << bfs_shift);
-	}
+	int bfs_shift = i2s->variant_regs->bfs_off;
 
 	/* Non-TDM I2S controllers do not support BCLK > 48 * FS */
 	if (!tdm && bfs > 48) {
@@ -292,6 +303,8 @@ static inline void set_bfs(struct i2s_dai *i2s, unsigned bfs)
 		return;
 	}
 
+	mod &= ~(i2s->variant_regs->bfs_mask << bfs_shift);
+
 	switch (bfs) {
 	case 48:
 		mod |= (MOD_BCLK_48FS << bfs_shift);
@@ -346,8 +359,9 @@ static inline int get_blc(struct i2s_dai *i2s)
 static void i2s_txctrl(struct i2s_dai *i2s, int on)
 {
 	void __iomem *addr = i2s->addr;
+	int txr_off = i2s->variant_regs->txr_off;
 	u32 con = readl(addr + I2SCON);
-	u32 mod = readl(addr + I2SMOD) & ~MOD_MASK;
+	u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off);
 
 	if (on) {
 		con |= CON_ACTIVE;
@@ -362,9 +376,9 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on)
 		}
 
 		if (any_rx_active(i2s))
-			mod |= MOD_TXRX;
+			mod |= 2 << txr_off;
 		else
-			mod |= MOD_TXONLY;
+			mod |= 0 << txr_off;
 	} else {
 		if (is_secondary(i2s)) {
 			con |=  CON_TXSDMA_PAUSE;
@@ -382,7 +396,7 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on)
 		con |=  CON_TXCH_PAUSE;
 
 		if (any_rx_active(i2s))
-			mod |= MOD_RXONLY;
+			mod |= 1 << txr_off;
 		else
 			con &= ~CON_ACTIVE;
 	}
@@ -395,23 +409,24 @@ static void i2s_txctrl(struct i2s_dai *i2s, int on)
 static void i2s_rxctrl(struct i2s_dai *i2s, int on)
 {
 	void __iomem *addr = i2s->addr;
+	int txr_off = i2s->variant_regs->txr_off;
 	u32 con = readl(addr + I2SCON);
-	u32 mod = readl(addr + I2SMOD) & ~MOD_MASK;
+	u32 mod = readl(addr + I2SMOD) & ~(3 << txr_off);
 
 	if (on) {
 		con |= CON_RXDMA_ACTIVE | CON_ACTIVE;
 		con &= ~(CON_RXDMA_PAUSE | CON_RXCH_PAUSE);
 
 		if (any_tx_active(i2s))
-			mod |= MOD_TXRX;
+			mod |= 2 << txr_off;
 		else
-			mod |= MOD_RXONLY;
+			mod |= 1 << txr_off;
 	} else {
 		con |=  CON_RXDMA_PAUSE | CON_RXCH_PAUSE;
 		con &= ~CON_RXDMA_ACTIVE;
 
 		if (any_tx_active(i2s))
-			mod |= MOD_TXONLY;
+			mod |= 0 << txr_off;
 		else
 			con &= ~CON_ACTIVE;
 	}
@@ -451,6 +466,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 	struct i2s_dai *i2s = to_info(dai);
 	struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai;
 	u32 mod = readl(i2s->addr + I2SMOD);
+	const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs;
+	unsigned int cdcon_mask = 1 << i2s_regs->cdclkcon_off;
+	unsigned int rsrc_mask = 1 << i2s_regs->rclksrc_off;
 
 	switch (clk_id) {
 	case SAMSUNG_I2S_OPCLK:
@@ -465,18 +483,18 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 		if ((rfs && other && other->rfs && (other->rfs != rfs)) ||
 				(any_active(i2s) &&
 				(((dir == SND_SOC_CLOCK_IN)
-					&& !(mod & MOD_CDCLKCON)) ||
+					&& !(mod & cdcon_mask)) ||
 				((dir == SND_SOC_CLOCK_OUT)
-					&& (mod & MOD_CDCLKCON))))) {
+					&& (mod & cdcon_mask))))) {
 			dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
 			return -EAGAIN;
 		}
 
 		if (dir == SND_SOC_CLOCK_IN)
-			mod |= MOD_CDCLKCON;
+			mod |= 1 << i2s_regs->cdclkcon_off;
 		else
-			mod &= ~MOD_CDCLKCON;
+			mod &= 0 << i2s_regs->cdclkcon_off;
 
 		i2s->rfs = rfs;
 		break;
@@ -491,8 +509,8 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 
 		if (!any_active(i2s)) {
 			if (i2s->op_clk && !IS_ERR(i2s->op_clk)) {
-				if ((clk_id && !(mod & MOD_IMS_SYSMUX)) ||
-					(!clk_id && (mod & MOD_IMS_SYSMUX))) {
+				if ((clk_id && !(mod & rsrc_mask)) ||
+					(!clk_id && (mod & rsrc_mask))) {
 					clk_disable_unprepare(i2s->op_clk);
 					clk_put(i2s->op_clk);
 				} else {
@@ -520,8 +538,8 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 				other->op_clk = i2s->op_clk;
 				other->rclk_srcrate = i2s->rclk_srcrate;
 			}
-		} else if ((!clk_id && (mod & MOD_IMS_SYSMUX))
-				|| (clk_id && !(mod & MOD_IMS_SYSMUX))) {
+		} else if ((!clk_id && (mod & rsrc_mask))
+				|| (clk_id && !(mod & rsrc_mask))) {
 			dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
 			return -EAGAIN;
@@ -533,10 +551,9 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai,
 		}
 
 		if (clk_id == 0)
-			mod &= ~MOD_IMS_SYSMUX;
+			mod &= 0 << i2s_regs->rclksrc_off;
 		else
-			mod |= MOD_IMS_SYSMUX;
-		break;
+			mod |= 1 << i2s_regs->rclksrc_off;
 
 	default:
 		dev_err(&i2s->pdev->dev, "We don't serve that!\n");
@@ -553,16 +570,12 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 {
 	struct i2s_dai *i2s = to_info(dai);
 	u32 mod = readl(i2s->addr + I2SMOD);
-	int lrp_shift, sdf_shift, sdf_mask, lrp_rlow;
+	int lrp_shift, sdf_shift, sdf_mask, lrp_rlow, mod_slave;
 	u32 tmp = 0;
 
-	if (i2s->quirks & QUIRK_SUPPORTS_TDM) {
-		lrp_shift = EXYNOS5420_MOD_LRP_SHIFT;
-		sdf_shift = EXYNOS5420_MOD_SDF_SHIFT;
-	} else {
-		lrp_shift = MOD_LRP_SHIFT;
-		sdf_shift = MOD_SDF_SHIFT;
-	}
+	lrp_shift = i2s->variant_regs->lrp_off;
+	sdf_shift = i2s->variant_regs->sdf_off;
+	mod_slave = 1 << i2s->variant_regs->mss_off;
 
 	sdf_mask = MOD_SDF_MASK << sdf_shift;
 	lrp_rlow = MOD_LR_RLOW << lrp_shift;
@@ -605,7 +618,7 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
-		tmp |= MOD_SLAVE;
+		tmp |= mod_slave;
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		/* Set default source clock in Master mode */
@@ -623,13 +636,13 @@ static int i2s_set_fmt(struct snd_soc_dai *dai,
 	 * channel.
 	 */
 	if (any_active(i2s) &&
-		((mod & (sdf_mask | lrp_rlow | MOD_SLAVE)) != tmp)) {
+		((mod & (sdf_mask | lrp_rlow | mod_slave)) != tmp)) {
 		dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
 		return -EAGAIN;
 	}
 
-	mod &= ~(sdf_mask | lrp_rlow | MOD_SLAVE);
+	mod &= ~(sdf_mask | lrp_rlow | mod_slave);
 	mod |= tmp;
 	writel(mod, i2s->addr + I2SMOD);
 
@@ -751,6 +764,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
 	struct i2s_dai *i2s = to_info(dai);
 	struct i2s_dai *other = i2s->pri_dai ? : i2s->sec_dai;
 	unsigned long flags;
+	const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs;
 
 	spin_lock_irqsave(&lock, flags);
 
@@ -761,7 +775,7 @@ static void i2s_shutdown(struct snd_pcm_substream *substream,
 		other->mode |= DAI_MANAGER;
 	} else {
 		u32 mod = readl(i2s->addr + I2SMOD);
-		i2s->cdclk_out = !(mod & MOD_CDCLKCON);
+		i2s->cdclk_out = !(mod & (1 << i2s_regs->cdclkcon_off));
 		if (other)
 			other->cdclk_out = i2s->cdclk_out;
 	}
@@ -914,13 +928,14 @@ i2s_delay(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
 	struct i2s_dai *i2s = to_info(dai);
 	u32 reg = readl(i2s->addr + I2SFIC);
 	snd_pcm_sframes_t delay;
+	const struct samsung_i2s_variant_regs *i2s_regs = i2s->variant_regs;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
 		delay = FIC_RXCOUNT(reg);
 	else if (is_secondary(i2s))
 		delay = FICS_TXCOUNT(readl(i2s->addr + I2SFICS));
 	else
-		delay = FIC_TXCOUNT(reg);
+		delay = (reg >> i2s_regs->ftx0cnt_off) & 0x7f;
 
 	return delay;
 }
@@ -1227,6 +1242,7 @@ static int samsung_i2s_probe(struct platform_device *pdev)
 	pri_dai->dma_capture.dma_size = 4;
 	pri_dai->base = regs_base;
 	pri_dai->quirks = quirks;
+	pri_dai->variant_regs = i2s_dai_data->i2s_variant_regs;
 
 	if (quirks & QUIRK_PRI_6CHAN)
 		pri_dai->i2s_dai_drv.playback.channels_max = 6;
@@ -1301,21 +1317,93 @@ static int samsung_i2s_remove(struct platform_device *pdev)
 	return 0;
 }
 
+static const struct samsung_i2s_variant_regs i2sv3_regs = {
+	.bfs_off = 1,
+	.rfs_off = 3,
+	.sdf_off = 5,
+	.txr_off = 8,
+	.rclksrc_off = 10,
+	.mss_off = 11,
+	.cdclkcon_off = 12,
+	.lrp_off = 7,
+	.bfs_mask = 0x3,
+	.rfs_mask = 0x3,
+	.ftx0cnt_off = 8,
+};
+
+static const struct samsung_i2s_variant_regs i2sv6_regs = {
+	.bfs_off = 0,
+	.rfs_off = 4,
+	.sdf_off = 6,
+	.txr_off = 8,
+	.rclksrc_off = 10,
+	.mss_off = 11,
+	.cdclkcon_off = 12,
+	.lrp_off = 15,
+	.bfs_mask = 0xf,
+	.rfs_mask = 0x3,
+	.ftx0cnt_off = 8,
+};
+
+static const struct samsung_i2s_variant_regs i2sv7_regs = {
+	.bfs_off = 0,
+	.rfs_off = 4,
+	.sdf_off = 7,
+	.txr_off = 9,
+	.rclksrc_off = 11,
+	.mss_off = 12,
+	.cdclkcon_off = 22,
+	.lrp_off = 15,
+	.bfs_mask = 0xf,
+	.rfs_mask = 0x7,
+	.ftx0cnt_off = 0,
+};
+
+static const struct samsung_i2s_variant_regs i2sv5_i2s1_regs = {
+	.bfs_off = 0,
+	.rfs_off = 3,
+	.sdf_off = 6,
+	.txr_off = 8,
+	.rclksrc_off = 10,
+	.mss_off = 11,
+	.cdclkcon_off = 12,
+	.lrp_off = 15,
+	.bfs_mask = 0x7,
+	.rfs_mask = 0x7,
+	.ftx0cnt_off = 8,
+};
+
 static const struct samsung_i2s_dai_data i2sv3_dai_type = {
 	.dai_type = TYPE_PRI,
 	.quirks = QUIRK_NO_MUXPSR,
+	.i2s_variant_regs = &i2sv3_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv5_dai_type = {
 	.dai_type = TYPE_PRI,
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
 			QUIRK_SUPPORTS_IDMA,
+	.i2s_variant_regs = &i2sv3_regs,
 };
 
 static const struct samsung_i2s_dai_data i2sv6_dai_type = {
 	.dai_type = TYPE_PRI,
 	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
 			QUIRK_SUPPORTS_TDM | QUIRK_SUPPORTS_IDMA,
+	.i2s_variant_regs = &i2sv6_regs,
+};
+
+static const struct samsung_i2s_dai_data i2sv7_dai_type = {
+	.dai_type = TYPE_PRI,
+	.quirks = QUIRK_PRI_6CHAN | QUIRK_SEC_DAI | QUIRK_NEED_RSTCLR |
+			QUIRK_SUPPORTS_TDM,
+	.i2s_variant_regs = &i2sv7_regs,
+};
+
+static const struct samsung_i2s_dai_data i2sv5_dai_type_i2s1 = {
+	.dai_type = TYPE_PRI,
+	.quirks = QUIRK_PRI_6CHAN | QUIRK_NEED_RSTCLR,
+	.i2s_variant_regs = &i2sv5_i2s1_regs,
 };
 
 static const struct samsung_i2s_dai_data samsung_dai_type_pri = {
@@ -1349,6 +1437,12 @@ static const struct of_device_id exynos_i2s_match[] = {
 	}, {
 		.compatible = "samsung,exynos5420-i2s",
 		.data = &i2sv6_dai_type,
+	}, {
+		.compatible = "samsung,exynos7-i2s",
+		.data = &i2sv7_dai_type,
+	}, {
+		.compatible = "samsung,exynos7-i2s1",
+		.data = &i2sv5_dai_type_i2s1,
 	},
 	{},
 };
-- 
cgit v1.2.3-59-g8ed1b


From a7a3324a602cd7ebabfb7f5990006ec4f3d6449f Mon Sep 17 00:00:00 2001
From: Misael Lopez Cruz <misael.lopez@ti.com>
Date: Wed, 12 Nov 2014 16:38:05 +0200
Subject: ASoC: davinci-mcasp: Add overrun/underrun event handling

An underrun (playback) event occurs when the serializer transfer
data from the XRBUF buffer to the XRSR shift register, but the
XRBUF hasn't been filled. Similarly, the overrun (capture) event
occurs when data from the XRSR shift register is transferred to
the XRBUF but it hasn't been read yet.

These events are handled as XRUN events that cause the pcm to stop.
The stream has to be explicitly restarted by the userspace which
ensures that after stopping/starting McASP the data transfer is
aligned with DMA. The other possibility was to internally stop and
start McASP without DMA even knowing about it.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../bindings/sound/davinci-mcasp-audio.txt         |   2 +-
 sound/soc/davinci/davinci-mcasp.c                  | 124 +++++++++++++++++++++
 sound/soc/davinci/davinci-mcasp.h                  |  11 ++
 3 files changed, 136 insertions(+), 1 deletion(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
index 60ca07996458..46bc9829c71a 100644
--- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
+++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt
@@ -32,7 +32,7 @@ Optional properties:
 - rx-num-evt : FIFO levels.
 - sram-size-playback : size of sram to be allocated during playback
 - sram-size-capture  : size of sram to be allocated during capture
-- interrupts : Interrupt numbers for McASP, currently not used by the driver
+- interrupts : Interrupt numbers for McASP
 - interrupt-names : Known interrupt names are "tx" and "rx"
 - pinctrl-0: Should specify pin control group used for this controller.
 - pinctrl-names: Should contain only one value - "default", for more details
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index a9822c7bbb0b..e460f97c514e 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -70,6 +70,7 @@ struct davinci_mcasp {
 	void __iomem *base;
 	u32 fifo_base;
 	struct device *dev;
+	struct snd_pcm_substream *substreams[2];
 
 	/* McASP specific data */
 	int	tdm_slots;
@@ -80,6 +81,7 @@ struct davinci_mcasp {
 	u8	bclk_div;
 	u16	bclk_lrclk_ratio;
 	int	streams;
+	u32	irq_request[2];
 
 	int	sysclk_freq;
 	bool	bclk_master;
@@ -185,6 +187,10 @@ static void mcasp_start_rx(struct davinci_mcasp *mcasp)
 	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
 	if (mcasp_is_synchronous(mcasp))
 		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+	/* enable receive IRQs */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
 }
 
 static void mcasp_start_tx(struct davinci_mcasp *mcasp)
@@ -214,6 +220,10 @@ static void mcasp_start_tx(struct davinci_mcasp *mcasp)
 	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
 	/* Release Frame Sync generator */
 	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+	/* enable transmit IRQs */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
 }
 
 static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
@@ -228,6 +238,10 @@ static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
 
 static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
 {
+	/* disable IRQ sources */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
+
 	/*
 	 * In synchronous mode stop the TX clocks if no other stream is
 	 * running
@@ -249,6 +263,10 @@ static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
 {
 	u32 val = 0;
 
+	/* disable IRQ sources */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
+
 	/*
 	 * In synchronous mode keep TX clocks running if the capture stream is
 	 * still running.
@@ -276,6 +294,76 @@ static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
 		mcasp_stop_rx(mcasp);
 }
 
+static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
+{
+	struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+	struct snd_pcm_substream *substream;
+	u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK];
+	u32 handled_mask = 0;
+	u32 stat;
+
+	stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG);
+	if (stat & XUNDRN & irq_mask) {
+		dev_warn(mcasp->dev, "Transmit buffer underflow\n");
+		handled_mask |= XUNDRN;
+
+		substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
+		if (substream) {
+			snd_pcm_stream_lock_irq(substream);
+			if (snd_pcm_running(substream))
+				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+			snd_pcm_stream_unlock_irq(substream);
+		}
+	}
+
+	if (!handled_mask)
+		dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n",
+			 stat);
+
+	if (stat & XRERR)
+		handled_mask |= XRERR;
+
+	/* Ack the handled event only */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask);
+
+	return IRQ_RETVAL(handled_mask);
+}
+
+static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
+{
+	struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+	struct snd_pcm_substream *substream;
+	u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE];
+	u32 handled_mask = 0;
+	u32 stat;
+
+	stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG);
+	if (stat & ROVRN & irq_mask) {
+		dev_warn(mcasp->dev, "Receive buffer overflow\n");
+		handled_mask |= ROVRN;
+
+		substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
+		if (substream) {
+			snd_pcm_stream_lock_irq(substream);
+			if (snd_pcm_running(substream))
+				snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
+			snd_pcm_stream_unlock_irq(substream);
+		}
+	}
+
+	if (!handled_mask)
+		dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n",
+			 stat);
+
+	if (stat & XRERR)
+		handled_mask |= XRERR;
+
+	/* Ack the handled event only */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask);
+
+	return IRQ_RETVAL(handled_mask);
+}
+
 static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 					 unsigned int fmt)
 {
@@ -869,6 +957,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	u32 max_channels = 0;
 	int i, dir;
 
+	mcasp->substreams[substream->stream] = substream;
+
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return 0;
 
@@ -907,6 +997,8 @@ static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
 {
 	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
 
+	mcasp->substreams[substream->stream] = NULL;
+
 	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
 		return;
 
@@ -1256,6 +1348,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	struct resource *mem, *ioarea, *res, *dat;
 	struct davinci_mcasp_pdata *pdata;
 	struct davinci_mcasp *mcasp;
+	char *irq_name;
+	int irq;
 	int ret;
 
 	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -1336,6 +1430,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	mcasp->dev = &pdev->dev;
 
+	irq = platform_get_irq_byname(pdev, "rx");
+	if (irq >= 0) {
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+					  dev_name(&pdev->dev));
+		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+						davinci_mcasp_rx_irq_handler,
+						IRQF_ONESHOT, irq_name, mcasp);
+		if (ret) {
+			dev_err(&pdev->dev, "RX IRQ request failed\n");
+			goto err;
+		}
+
+		mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN;
+	}
+
+	irq = platform_get_irq_byname(pdev, "tx");
+	if (irq >= 0) {
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+					  dev_name(&pdev->dev));
+		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+						davinci_mcasp_tx_irq_handler,
+						IRQF_ONESHOT, irq_name, mcasp);
+		if (ret) {
+			dev_err(&pdev->dev, "TX IRQ request failed\n");
+			goto err;
+		}
+
+		mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN;
+	}
+
 	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
 	if (dat)
 		mcasp->dat_port = true;
diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h
index 9737108f0305..79dc511180bf 100644
--- a/sound/soc/davinci/davinci-mcasp.h
+++ b/sound/soc/davinci/davinci-mcasp.h
@@ -256,6 +256,7 @@
  * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits
  * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits
  */
+#define XRERR		BIT(8) /* Transmit/Receive error */
 #define XRDATA		BIT(5) /* Transmit/Receive data ready */
 
 /*
@@ -284,6 +285,16 @@
  */
 #define TXDATADMADIS	BIT(0)
 
+/*
+ * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits
+ */
+#define ROVRN		BIT(0)
+
+/*
+ * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits
+ */
+#define XUNDRN		BIT(0)
+
 /*
  * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
  */
-- 
cgit v1.2.3-59-g8ed1b


From caaeb6a96f35af14f615838b924f54c9d0f33ab3 Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert+renesas@glider.be>
Date: Mon, 10 Nov 2014 20:00:40 +0100
Subject: ASoC: sh: fsi: Document SoC-specific bindings

The documentation only mentioned the generic fallback compatible
property.
Add the missing SoC-specific compatible properties, some of which are
already in use.

Also fix a small typo, while we're at it.

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/renesas,fsi.txt | 9 +++++++--
 1 file changed, 7 insertions(+), 2 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.txt b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
index c5be003f413e..0d0ab51105b0 100644
--- a/Documentation/devicetree/bindings/sound/renesas,fsi.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,fsi.txt
@@ -1,11 +1,16 @@
 Renesas FSI
 
 Required properties:
-- compatible			: "renesas,sh_fsi2" or "renesas,sh_fsi"
+- compatible			: "renesas,fsi2-<soctype>",
+				  "renesas,sh_fsi2" or "renesas,sh_fsi" as
+				  fallback.
+				  Examples with soctypes are:
+				    - "renesas,fsi2-r8a7740" (R-Mobile A1)
+				    - "renesas,fsi2-sh73a0" (SH-Mobile AG5)
 - reg				: Should contain the register physical address and length
 - interrupts			: Should contain FSI interrupt
 
-- fsia,spdif-connection		: FSI is connected by S/PDFI
+- fsia,spdif-connection		: FSI is connected by S/PDIF
 - fsia,stream-mode-support	: FSI supports 16bit stream mode.
 - fsia,use-internal-clock	: FSI uses internal clock when master mode.
 
-- 
cgit v1.2.3-59-g8ed1b


From 56ba98acc398883324c0e70dc8aee1dc53eb2331 Mon Sep 17 00:00:00 2001
From: Geert Uytterhoeven <geert+renesas@glider.be>
Date: Mon, 10 Nov 2014 20:00:42 +0100
Subject: ASoC: rsnd: Document SoC-specific bindings

The documentation only mentioned the generic fallback compatible
property.
Add the missing SoC-specific compatible properties, which are already in
use.

Also drop a bogus 0x unit-address prefix while we're at it.

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 10 +++++++---
 1 file changed, 7 insertions(+), 3 deletions(-)

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
index aa697abf337e..2dd690bc19cc 100644
--- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
+++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt
@@ -1,8 +1,12 @@
 Renesas R-Car sound
 
 Required properties:
-- compatible			: "renesas,rcar_sound-gen1" if generation1
+- compatible			: "renesas,rcar_sound-<soctype>", fallbacks
+				  "renesas,rcar_sound-gen1" if generation1, and
 				  "renesas,rcar_sound-gen2" if generation2
+				  Examples with soctypes are:
+				    - "renesas,rcar_sound-r8a7790" (R-Car H2)
+				    - "renesas,rcar_sound-r8a7791" (R-Car M2-W)
 - reg				: Should contain the register physical address.
 				  required register is
 				   SRU/ADG/SSI      if generation1
@@ -35,9 +39,9 @@ DAI subnode properties:
 
 Example:
 
-rcar_sound: rcar_sound@0xffd90000 {
+rcar_sound: rcar_sound@ec500000 {
 	#sound-dai-cells = <1>;
-	compatible = "renesas,rcar_sound-gen2";
+	compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2";
 	reg =	<0 0xec500000 0 0x1000>, /* SCU */
 		<0 0xec5a0000 0 0x100>,  /* ADG */
 		<0 0xec540000 0 0x1000>, /* SSIU */
-- 
cgit v1.2.3-59-g8ed1b


From 9e6280cd44c1cceaaac921567ee8c5731b7cc72b Mon Sep 17 00:00:00 2001
From: Krishna Mohan Dani <krishna.md@samsung.com>
Date: Thu, 13 Nov 2014 17:44:21 +0530
Subject: ASoC: rt5631: Add device tree binding documentation

Document the device tree binding for the ALC5631 codec and update vendor
specific prefix for the Realtek.

Signed-off-by: Krishna Mohan Dani <krishna.md@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/rt5631.txt | 48 ++++++++++++++++++++++
 1 file changed, 48 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/rt5631.txt

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/rt5631.txt b/Documentation/devicetree/bindings/sound/rt5631.txt
new file mode 100644
index 000000000000..92b986ca337b
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rt5631.txt
@@ -0,0 +1,48 @@
+ALC5631/RT5631 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "realtek,alc5631" or "realtek,rt5631"
+
+  - reg : the I2C address of the device.
+
+Pins on the device (for linking into audio routes):
+
+  * SPK_OUT_R_P
+  * SPK_OUT_R_N
+  * SPK_OUT_L_P
+  * SPK_OUT_L_N
+  * HP_OUT_L
+  * HP_OUT_R
+  * AUX_OUT2_LP
+  * AUX_OUT2_RN
+  * AUX_OUT1_LP
+  * AUX_OUT1_RN
+  * AUX_IN_L_JD
+  * AUX_IN_R_JD
+  * MONO_IN_P
+  * MONO_IN_N
+  * MIC1_P
+  * MIC1_N
+  * MIC2_P
+  * MIC2_N
+  * MONO_OUT_P
+  * MONO_OUT_N
+  * MICBIAS1
+  * MICBIAS2
+
+Example:
+
+alc5631: alc5631@1a {
+	compatible = "realtek,alc5631";
+	reg = <0x1a>;
+};
+
+or
+
+rt5631: rt5631@1a {
+	compatible = "realtek,rt5631";
+	reg = <0x1a>;
+};
-- 
cgit v1.2.3-59-g8ed1b


From 2880fc877971d6c14b0c76ac09744e3ff5b126d5 Mon Sep 17 00:00:00 2001
From: Dylan Reid <dgreid@chromium.org>
Date: Thu, 13 Nov 2014 11:18:29 -0800
Subject: ASoC: add TI ts3a227e headset chip driver

The TS3A227E is an autonomous audio accessory detection and
configuration switch that detects 3-pole or 4-pole audio accessories
and configures internal switches to route the signals accordingly.

This chip also has built-in support for the new button standard
described in the Android "Wired audio headset specification" v1.0.
These buttons will be reported on the jack as buttons 0-3 mapped to
KEY_MEDIA, KEY_VOLUMEUP, KEY_VOLUMEDOWN, and KEY_VOICE_COMMAND.

This will be added as an aux_dev and have the jack passed in from the
machine driver.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/ts3a227e.txt         |  26 ++
 sound/soc/codecs/Kconfig                           |   5 +
 sound/soc/codecs/Makefile                          |   2 +
 sound/soc/codecs/ts3a227e.c                        | 314 +++++++++++++++++++++
 sound/soc/codecs/ts3a227e.h                        |  17 ++
 5 files changed, 364 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/ts3a227e.txt
 create mode 100644 sound/soc/codecs/ts3a227e.c
 create mode 100644 sound/soc/codecs/ts3a227e.h

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt
new file mode 100644
index 000000000000..e8bf23eb1803
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt
@@ -0,0 +1,26 @@
+Texas Instruments TS3A227E
+Autonomous Audio Accessory Detection and Configuration Switch
+
+The TS3A227E detect headsets of 3-ring and 4-ring standards and
+switches automatically to route the microphone correctly.  It also
+handles key press detection in accordance with the Android audio
+headset specification v1.0.
+
+Required properties:
+
+ - compatible:		Should contain "ti,ts3a227e".
+ - reg:			The i2c address. Should contain <0x3b>.
+ - interrupt-parent:	The parent interrupt controller
+ - interrupts:		Interrupt number for /INT pin from the 227e
+
+
+Examples:
+
+	i2c {
+		ts3a227e@3b {
+			compatible = "ti,ts3a227e";
+			reg = <0x3b>;
+			interrupt-parent = <&gpio>;
+			interrupts = <3 IRQ_TYPE_LEVEL_LOW>;
+		};
+	};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index a68d1731a8fd..243ec862c426 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -109,6 +109,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
+	select SND_SOC_TS3A227E if I2C
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_TWL6040 if TWL6040_CORE
 	select SND_SOC_UDA134X
@@ -607,6 +608,10 @@ config SND_SOC_TLV320AIC3X
 config SND_SOC_TLV320DAC33
 	tristate
 
+config SND_SOC_TS3A227E
+	tristate "TI Headset/Mic detect and keypress chip"
+	depends on I2C
+
 config SND_SOC_TWL4030
 	select MFD_TWL4030_AUDIO
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5dce451661e4..a1eb7ef3b90e 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -109,6 +109,7 @@ snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o
 snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
+snd-soc-ts3a227e-objs := ts3a227e.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
 snd-soc-uda134x-objs := uda134x.o
@@ -282,6 +283,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC31XX)     += snd-soc-tlv320aic31xx.o
 obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
+obj-$(CONFIG_SND_SOC_TS3A227E)	+= snd-soc-ts3a227e.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c
new file mode 100644
index 000000000000..1d1205702d23
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.c
@@ -0,0 +1,314 @@
+/*
+ * TS3A227E Autonomous Audio Accessory Detection and Configuration Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/init.h>
+#include <linux/input.h>
+#include <linux/module.h>
+#include <linux/of_gpio.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/soc.h>
+
+struct ts3a227e {
+	struct regmap *regmap;
+	struct snd_soc_jack *jack;
+	bool plugged;
+	bool mic_present;
+	unsigned int buttons_held;
+};
+
+/* Button values to be reported on the jack */
+static const int ts3a227e_buttons[] = {
+	SND_JACK_BTN_0,
+	SND_JACK_BTN_1,
+	SND_JACK_BTN_2,
+	SND_JACK_BTN_3,
+};
+
+#define TS3A227E_NUM_BUTTONS 4
+#define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \
+			    SND_JACK_MICROPHONE | \
+			    SND_JACK_BTN_0 | \
+			    SND_JACK_BTN_1 | \
+			    SND_JACK_BTN_2 | \
+			    SND_JACK_BTN_3)
+
+/* TS3A227E registers */
+#define TS3A227E_REG_DEVICE_ID		0x00
+#define TS3A227E_REG_INTERRUPT		0x01
+#define TS3A227E_REG_KP_INTERRUPT	0x02
+#define TS3A227E_REG_INTERRUPT_DISABLE	0x03
+#define TS3A227E_REG_SETTING_1		0x04
+#define TS3A227E_REG_SETTING_2		0x05
+#define TS3A227E_REG_SETTING_3		0x06
+#define TS3A227E_REG_SWITCH_CONTROL_1	0x07
+#define TS3A227E_REG_SWITCH_CONTROL_2	0x08
+#define TS3A227E_REG_SWITCH_STATUS_1	0x09
+#define TS3A227E_REG_SWITCH_STATUS_2	0x0a
+#define TS3A227E_REG_ACCESSORY_STATUS	0x0b
+#define TS3A227E_REG_ADC_OUTPUT		0x0c
+#define TS3A227E_REG_KP_THRESHOLD_1	0x0d
+#define TS3A227E_REG_KP_THRESHOLD_2	0x0e
+#define TS3A227E_REG_KP_THRESHOLD_3	0x0f
+
+/* TS3A227E_REG_INTERRUPT 0x01 */
+#define INS_REM_EVENT 0x01
+#define DETECTION_COMPLETE_EVENT 0x02
+
+/* TS3A227E_REG_KP_INTERRUPT 0x02 */
+#define PRESS_MASK(idx) (0x01 << (2 * (idx)))
+#define RELEASE_MASK(idx) (0x02 << (2 * (idx)))
+
+/* TS3A227E_REG_INTERRUPT_DISABLE 0x03 */
+#define INS_REM_INT_DISABLE 0x01
+#define DETECTION_COMPLETE_INT_DISABLE 0x02
+#define ADC_COMPLETE_INT_DISABLE 0x04
+#define INTB_DISABLE 0x08
+
+/* TS3A227E_REG_SETTING_2 0x05 */
+#define KP_ENABLE 0x04
+
+/* TS3A227E_REG_ACCESSORY_STATUS  0x0b */
+#define TYPE_3_POLE 0x01
+#define TYPE_4_POLE_OMTP 0x02
+#define TYPE_4_POLE_STANDARD 0x04
+#define JACK_INSERTED 0x08
+#define EITHER_MIC_MASK (TYPE_4_POLE_OMTP | TYPE_4_POLE_STANDARD)
+
+static const struct reg_default ts3a227e_reg_defaults[] = {
+	{ TS3A227E_REG_DEVICE_ID, 0x10 },
+	{ TS3A227E_REG_INTERRUPT, 0x00 },
+	{ TS3A227E_REG_KP_INTERRUPT, 0x00 },
+	{ TS3A227E_REG_INTERRUPT_DISABLE, 0x08 },
+	{ TS3A227E_REG_SETTING_1, 0x23 },
+	{ TS3A227E_REG_SETTING_2, 0x00 },
+	{ TS3A227E_REG_SETTING_3, 0x0e },
+	{ TS3A227E_REG_SWITCH_CONTROL_1, 0x00 },
+	{ TS3A227E_REG_SWITCH_CONTROL_2, 0x00 },
+	{ TS3A227E_REG_SWITCH_STATUS_1, 0x0c },
+	{ TS3A227E_REG_SWITCH_STATUS_2, 0x00 },
+	{ TS3A227E_REG_ACCESSORY_STATUS, 0x00 },
+	{ TS3A227E_REG_ADC_OUTPUT, 0x00 },
+	{ TS3A227E_REG_KP_THRESHOLD_1, 0x20 },
+	{ TS3A227E_REG_KP_THRESHOLD_2, 0x40 },
+	{ TS3A227E_REG_KP_THRESHOLD_3, 0x68 },
+};
+
+static bool ts3a227e_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_DEVICE_ID ... TS3A227E_REG_KP_THRESHOLD_3:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool ts3a227e_writeable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_INTERRUPT_DISABLE ... TS3A227E_REG_SWITCH_CONTROL_2:
+	case TS3A227E_REG_KP_THRESHOLD_1 ... TS3A227E_REG_KP_THRESHOLD_3:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE:
+	case TS3A227E_REG_SETTING_2:
+	case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT:
+		return true;
+	default:
+		return false;
+	}
+}
+
+static void ts3a227e_jack_report(struct ts3a227e *ts3a227e)
+{
+	unsigned int i;
+	int report = 0;
+
+	if (!ts3a227e->jack)
+		return;
+
+	if (ts3a227e->plugged)
+		report = SND_JACK_HEADPHONE;
+	if (ts3a227e->mic_present)
+		report |= SND_JACK_MICROPHONE;
+	for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+		if (ts3a227e->buttons_held & (1 << i))
+			report |= ts3a227e_buttons[i];
+	}
+	snd_soc_jack_report(ts3a227e->jack, report, TS3A227E_JACK_MASK);
+}
+
+static void ts3a227e_new_jack_state(struct ts3a227e *ts3a227e, unsigned acc_reg)
+{
+	bool plugged, mic_present;
+
+	plugged = !!(acc_reg & JACK_INSERTED);
+	mic_present = plugged && !!(acc_reg & EITHER_MIC_MASK);
+
+	ts3a227e->plugged = plugged;
+
+	if (mic_present != ts3a227e->mic_present) {
+		ts3a227e->mic_present = mic_present;
+		ts3a227e->buttons_held = 0;
+		if (mic_present) {
+			/* Enable key press detection. */
+			regmap_update_bits(ts3a227e->regmap,
+					   TS3A227E_REG_SETTING_2,
+					   KP_ENABLE, KP_ENABLE);
+		}
+	}
+}
+
+static irqreturn_t ts3a227e_interrupt(int irq, void *data)
+{
+	struct ts3a227e *ts3a227e = (struct ts3a227e *)data;
+	struct regmap *regmap = ts3a227e->regmap;
+	unsigned int int_reg, kp_int_reg, acc_reg, i;
+
+	/* Check for plug/unplug. */
+	regmap_read(regmap, TS3A227E_REG_INTERRUPT, &int_reg);
+	if (int_reg & (DETECTION_COMPLETE_EVENT | INS_REM_EVENT)) {
+		regmap_read(regmap, TS3A227E_REG_ACCESSORY_STATUS, &acc_reg);
+		ts3a227e_new_jack_state(ts3a227e, acc_reg);
+	}
+
+	/* Report any key events. */
+	regmap_read(regmap, TS3A227E_REG_KP_INTERRUPT, &kp_int_reg);
+	for (i = 0; i < TS3A227E_NUM_BUTTONS; i++) {
+		if (kp_int_reg & PRESS_MASK(i))
+			ts3a227e->buttons_held |= (1 << i);
+		if (kp_int_reg & RELEASE_MASK(i))
+			ts3a227e->buttons_held &= ~(1 << i);
+	}
+
+	ts3a227e_jack_report(ts3a227e);
+
+	return IRQ_HANDLED;
+}
+
+/**
+ * ts3a227e_enable_jack_detect - Specify a jack for event reporting
+ *
+ * @component:  component to register the jack with
+ * @jack: jack to use to report headset and button events on
+ *
+ * After this function has been called the headset insert/remove and button
+ * events 0-3 will be routed to the given jack.  Jack can be null to stop
+ * reporting.
+ */
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+				struct snd_soc_jack *jack)
+{
+	struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component);
+
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
+	snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+
+	ts3a227e->jack = jack;
+	ts3a227e_jack_report(ts3a227e);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect);
+
+static struct snd_soc_component_driver ts3a227e_soc_driver;
+
+static const struct regmap_config ts3a227e_regmap_config = {
+	.val_bits = 8,
+	.reg_bits = 8,
+
+	.max_register = TS3A227E_REG_KP_THRESHOLD_3,
+	.readable_reg = ts3a227e_readable_reg,
+	.writeable_reg = ts3a227e_writeable_reg,
+	.volatile_reg = ts3a227e_volatile_reg,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = ts3a227e_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults),
+};
+
+static int ts3a227e_i2c_probe(struct i2c_client *i2c,
+			      const struct i2c_device_id *id)
+{
+	struct ts3a227e *ts3a227e;
+	struct device *dev = &i2c->dev;
+	int ret;
+
+	ts3a227e = devm_kzalloc(&i2c->dev, sizeof(*ts3a227e), GFP_KERNEL);
+	if (ts3a227e == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, ts3a227e);
+
+	ts3a227e->regmap = devm_regmap_init_i2c(i2c, &ts3a227e_regmap_config);
+	if (IS_ERR(ts3a227e->regmap))
+		return PTR_ERR(ts3a227e->regmap);
+
+	ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt,
+					IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+					"TS3A227E", ts3a227e);
+	if (ret) {
+		dev_err(dev, "Cannot request irq %d (%d)\n", i2c->irq, ret);
+		return ret;
+	}
+
+	ret = devm_snd_soc_register_component(&i2c->dev, &ts3a227e_soc_driver,
+					      NULL, 0);
+	if (ret)
+		return ret;
+
+	/* Enable interrupts except for ADC complete. */
+	regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_INTERRUPT_DISABLE,
+			   INTB_DISABLE | ADC_COMPLETE_INT_DISABLE,
+			   ADC_COMPLETE_INT_DISABLE);
+
+	return 0;
+}
+
+static const struct i2c_device_id ts3a227e_i2c_ids[] = {
+	{ "ts3a227e", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ts3a227e_i2c_ids);
+
+static const struct of_device_id ts3a227e_of_match[] = {
+	{ .compatible = "ti,ts3a227e", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ts3a227e_of_match);
+
+static struct i2c_driver ts3a227e_driver = {
+	.driver = {
+		.name = "ts3a227e",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(ts3a227e_of_match),
+	},
+	.probe = ts3a227e_i2c_probe,
+	.id_table = ts3a227e_i2c_ids,
+};
+module_i2c_driver(ts3a227e_driver);
+
+MODULE_DESCRIPTION("ASoC ts3a227e driver");
+MODULE_AUTHOR("Dylan Reid <dgreid@chromium.org>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ts3a227e.h b/sound/soc/codecs/ts3a227e.h
new file mode 100644
index 000000000000..e2acf9c5bebe
--- /dev/null
+++ b/sound/soc/codecs/ts3a227e.h
@@ -0,0 +1,17 @@
+/*
+ * TS3A227E Autonous Audio Accessory Detection and Configureation Switch
+ *
+ * Copyright (C) 2014 Google, Inc.
+ *
+ * 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 _TS3A227E_H
+#define _TS3A227E_H
+
+int ts3a227e_enable_jack_detect(struct snd_soc_component *component,
+				struct snd_soc_jack *jack);
+
+#endif
-- 
cgit v1.2.3-59-g8ed1b


From e2280c9040d8bc5039617af35ccf7b8ac4abb428 Mon Sep 17 00:00:00 2001
From: Zidan Wang <b50113@freescale.com>
Date: Thu, 20 Nov 2014 19:07:48 +0800
Subject: ASoC: wm8960: Add device tree support

Document the device tree binding for the WM8960 codec, and modify the
driver to extract the platform data from device tree, if present.

Signed-off-by: Zidan Wang <b50113@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 Documentation/devicetree/bindings/sound/wm8960.txt | 31 ++++++++++++++++
 sound/soc/codecs/wm8960.c                          | 41 ++++++++++++++++------
 2 files changed, 62 insertions(+), 10 deletions(-)
 create mode 100644 Documentation/devicetree/bindings/sound/wm8960.txt

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt
new file mode 100644
index 000000000000..2deb8a3da9c5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/wm8960.txt
@@ -0,0 +1,31 @@
+WM8960 audio CODEC
+
+This device supports I2C only.
+
+Required properties:
+
+  - compatible : "wlf,wm8960"
+
+  - reg : the I2C address of the device.
+
+Optional properties:
+  - wlf,shared-lrclk: This is a boolean property. If present, the LRCM bit of
+	R24 (Additional control 2) gets set, indicating that ADCLRC and DACLRC pins
+	will be disabled only when ADC (Left and Right) and DAC (Left and Right)
+	are disabled.
+	When wm8960 works on synchronize mode and DACLRC pin is used to supply
+	frame clock, it will no frame clock for captrue unless enable DAC to enable
+	DACLRC pin. If shared-lrclk is present, no need to enable DAC for captrue.
+
+  - wlf,capless: This is a boolean property. If present, OUT3 pin will be
+	enabled and disabled together with HP_L and HP_R pins in response to jack
+	detect events.
+
+Example:
+
+codec: wm8960@1a {
+	compatible = "wlf,wm8960";
+	reg = <0x1a>;
+
+	wlf,shared-lrclk;
+};
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 4dc4e85116cd..99d6457c87ba 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -125,6 +125,7 @@ struct wm8960_priv {
 	struct snd_soc_dapm_widget *out3;
 	bool deemph;
 	int playback_fs;
+	struct wm8960_data pdata;
 };
 
 #define wm8960_reset(c)	snd_soc_write(c, WM8960_RESET, 0)
@@ -440,8 +441,8 @@ static const struct snd_soc_dapm_route audio_paths_capless[] = {
 
 static int wm8960_add_widgets(struct snd_soc_codec *codec)
 {
-	struct wm8960_data *pdata = codec->dev->platform_data;
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
+	struct wm8960_data *pdata = &wm8960->pdata;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_dapm_widget *w;
 
@@ -961,17 +962,13 @@ static int wm8960_resume(struct snd_soc_codec *codec)
 static int wm8960_probe(struct snd_soc_codec *codec)
 {
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-	struct wm8960_data *pdata = dev_get_platdata(codec->dev);
+	struct wm8960_data *pdata = &wm8960->pdata;
 	int ret;
 
-	wm8960->set_bias_level = wm8960_set_bias_level_out3;
-
-	if (!pdata) {
-		dev_warn(codec->dev, "No platform data supplied\n");
-	} else {
-		if (pdata->capless)
-			wm8960->set_bias_level = wm8960_set_bias_level_capless;
-	}
+	if (pdata->capless)
+		wm8960->set_bias_level = wm8960_set_bias_level_capless;
+	else
+		wm8960->set_bias_level = wm8960_set_bias_level_out3;
 
 	ret = wm8960_reset(codec);
 	if (ret < 0) {
@@ -1029,6 +1026,18 @@ static const struct regmap_config wm8960_regmap = {
 	.volatile_reg = wm8960_volatile,
 };
 
+static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
+				struct wm8960_data *pdata)
+{
+	const struct device_node *np = i2c->dev.of_node;
+
+	if (of_property_read_bool(np, "wlf,capless"))
+		pdata->capless = true;
+
+	if (of_property_read_bool(np, "wlf,shared-lrclk"))
+		pdata->shared_lrclk = true;
+}
+
 static int wm8960_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
@@ -1045,6 +1054,11 @@ static int wm8960_i2c_probe(struct i2c_client *i2c,
 	if (IS_ERR(wm8960->regmap))
 		return PTR_ERR(wm8960->regmap);
 
+	if (pdata)
+		memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data));
+	else if (i2c->dev.of_node)
+		wm8960_set_pdata_from_of(i2c, &wm8960->pdata);
+
 	if (pdata && pdata->shared_lrclk) {
 		ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2,
 					 0x4, 0x4);
@@ -1075,10 +1089,17 @@ static const struct i2c_device_id wm8960_i2c_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
 
+static const struct of_device_id wm8960_of_match[] = {
+       { .compatible = "wlf,wm8960", },
+       { }
+};
+MODULE_DEVICE_TABLE(of, wm8960_of_match);
+
 static struct i2c_driver wm8960_i2c_driver = {
 	.driver = {
 		.name = "wm8960",
 		.owner = THIS_MODULE,
+		.of_match_table = wm8960_of_match,
 	},
 	.probe =    wm8960_i2c_probe,
 	.remove =   wm8960_i2c_remove,
-- 
cgit v1.2.3-59-g8ed1b


From d683d0b690c13437d752ccce47963ac64119b07a Mon Sep 17 00:00:00 2001
From: Krishna Mohan Dani <krishna.md@samsung.com>
Date: Wed, 26 Nov 2014 14:53:04 +0530
Subject: ASoC: Samsung: Add arndale_rt5631 machine driver and binding

Adding machine driver to instantiate I2S based realtek's ALC5631
sound card on Arndale board.

There are other variants of Audio Daughter Cards for Arndale
Board for which support already exists but there is no support for
Realtek's alc5631 codec hence support for ALC5631 based machine
driver is being added.
This patch also documents the device tree binding for the Arndale
board based machine driver.

Signed-off-by: Claude Youn <claude.youn@gmail.com>
Signed-off-by: Krishna Mohan Dani <krishna.md@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
 .../devicetree/bindings/sound/arndale.txt          |  24 ++++
 sound/soc/samsung/Kconfig                          |   6 +
 sound/soc/samsung/Makefile                         |   2 +
 sound/soc/samsung/arndale_rt5631.c                 | 150 +++++++++++++++++++++
 4 files changed, 182 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/arndale.txt
 create mode 100644 sound/soc/samsung/arndale_rt5631.c

(limited to 'Documentation/devicetree/bindings')

diff --git a/Documentation/devicetree/bindings/sound/arndale.txt b/Documentation/devicetree/bindings/sound/arndale.txt
new file mode 100644
index 000000000000..0e76946385ae
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/arndale.txt
@@ -0,0 +1,24 @@
+Audio Binding for Arndale boards
+
+Required properties:
+- compatible : Can be the following,
+			"samsung,arndale-rt5631"
+
+- samsung,audio-cpu: The phandle of the Samsung I2S controller
+- samsung,audio-codec: The phandle of the audio codec
+
+Optional:
+- samsung,model: The name of the sound-card
+
+Arndale Boards has many audio daughter cards, one of them is
+rt5631/alc5631. Below example shows audio bindings for rt5631/
+alc5631 based codec.
+
+Example:
+
+sound {
+		compatible = "samsung,arndale-rt5631";
+
+		samsung,audio-cpu = <&i2s0>
+		samsung,audio-codec = <&rt5631>;
+};
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index e0e737faadd9..fc67f97f19f6 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -239,3 +239,9 @@ config SND_SOC_ODROIDX2
 	select SND_SAMSUNG_I2S
 	help
 	  Say Y here to enable audio support for the Odroid-X2/U3.
+
+config SND_SOC_ARNDALE_RT5631_ALC5631
+        tristate "Audio support for RT5631(ALC5631) on Arndale Board"
+        depends on SND_SOC_SAMSUNG
+        select SND_SAMSUNG_I2S
+        select SND_SOC_RT5631
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 91505ddaaf95..31e3dba7e3b5 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -45,6 +45,7 @@ snd-soc-lowland-objs := lowland.o
 snd-soc-littlemill-objs := littlemill.o
 snd-soc-bells-objs := bells.o
 snd-soc-odroidx2-max98090-objs := odroidx2_max98090.o
+snd-soc-arndale-rt5631-objs := arndale_rt5631.o
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
@@ -71,3 +72,4 @@ obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o
 obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o
 obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o
 obj-$(CONFIG_SND_SOC_ODROIDX2) += snd-soc-odroidx2-max98090.o
+obj-$(CONFIG_SND_SOC_ARNDALE_RT5631_ALC5631) += snd-soc-arndale-rt5631.o
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
new file mode 100644
index 000000000000..1e2b61ca8db2
--- /dev/null
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -0,0 +1,150 @@
+/*
+ *  arndale_rt5631.c
+ *
+ *  Copyright (c) 2014, Insignal Co., Ltd.
+ *
+ *  Author: Claude <claude@insginal.co.kr>
+ *
+ *  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 <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "i2s.h"
+
+static int arndale_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int rfs, ret;
+	unsigned long rclk;
+
+	rfs = 256;
+
+	rclk = params_rate(params) * rfs;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK,
+					0, SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0,
+					0, SND_SOC_CLOCK_OUT);
+
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, rclk, SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops arndale_ops = {
+	.hw_params = arndale_hw_params,
+};
+
+static struct snd_soc_dai_link arndale_rt5631_dai[] = {
+	{
+		.name = "RT5631 HiFi",
+		.stream_name = "Primary",
+		.codec_dai_name = "rt5631-hifi",
+		.dai_fmt = SND_SOC_DAIFMT_I2S
+			| SND_SOC_DAIFMT_NB_NF
+			| SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &arndale_ops,
+	},
+};
+
+static struct snd_soc_card arndale_rt5631 = {
+	.name = "Arndale RT5631",
+	.dai_link = arndale_rt5631_dai,
+	.num_links = ARRAY_SIZE(arndale_rt5631_dai),
+};
+
+static int arndale_audio_probe(struct platform_device *pdev)
+{
+	int n, ret;
+	struct device_node *np = pdev->dev.of_node;
+	struct snd_soc_card *card = &arndale_rt5631;
+
+	card->dev = &pdev->dev;
+
+	for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) {
+		if (!arndale_rt5631_dai[n].cpu_dai_name) {
+			arndale_rt5631_dai[n].cpu_of_node = of_parse_phandle(np,
+					"samsung,audio-cpu", n);
+
+			if (!arndale_rt5631_dai[n].cpu_of_node) {
+				dev_err(&pdev->dev,
+				"Property 'samsung,audio-cpu' missing or invalid\n");
+				return -EINVAL;
+			}
+		}
+		if (!arndale_rt5631_dai[n].platform_name)
+			arndale_rt5631_dai[n].platform_of_node =
+					arndale_rt5631_dai[n].cpu_of_node;
+
+		arndale_rt5631_dai[n].codec_name = NULL;
+		arndale_rt5631_dai[n].codec_of_node = of_parse_phandle(np,
+					"samsung,audio-codec", n);
+		if (!arndale_rt5631_dai[0].codec_of_node) {
+			dev_err(&pdev->dev,
+			"Property 'samsung,audio-codec' missing or invalid\n");
+			return -EINVAL;
+		}
+	}
+
+	ret = devm_snd_soc_register_card(card->dev, card);
+
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
+
+	return ret;
+}
+
+static int arndale_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
+	{ .compatible = "samsung,arndale-rt5631", },
+	{ .compatible = "samsung,arndale-alc5631", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, samsung_arndale_rt5631_of_match);
+
+static struct platform_driver arndale_audio_driver = {
+	.driver = {
+		.name   = "arndale-audio",
+		.owner  = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
+	},
+	.probe = arndale_audio_probe,
+	.remove = arndale_audio_remove,
+};
+
+module_platform_driver(arndale_audio_driver);
+
+MODULE_AUTHOR("Claude <claude@insignal.co.kr>");
+MODULE_DESCRIPTION("ALSA SoC Driver for Arndale Board");
+MODULE_LICENSE("GPL");
-- 
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