* [PATCHv3 0/6] Motorola Droid 4 Audio Support
@ 2017-07-25 15:10 ` Sebastian Reichel
0 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Hi,
This adds audio support to Motorola Droid 4.
Tested:
- Playing via both DACs using Speaker, Earpiece, Headphone
- Recording using internal Mic
- Volume Controls
Known not to work:
- 3.5mm detection support (requires closed source firmware,
needs further investigation)
- Modem / Bluetooth Audio (actually untested, but probably
needs some quirks)
Changes since PATCHv2:
* Fix a whitespace issue
* Fix const notes Takashi provided
* Fix a DAPM route issue I accidently introduced in v2
Changes since PATCHv1:
* Add patch from Tony exporting soc_dpcm_runtime_update
* Integrate DT patch for vaudio initial mode
* Split dt-binding from codec patch and add Ack from Rob
* Fix CPCAP position in Kconfig/Makefile
* Avoid "err +=" constructs
* Simplify reset function
* Drop cpcap_audio_read/write helpers
* Do not use tertiary operator for mute register value
* Update Input Mux logic
* Switch from simple-audio-card to audio-graph-card
-- Sebastian
Sebastian Reichel (5):
dt-bindings: sound: add motorola,cpcap-audio-codec
ASoC: codec: cpcap: new codec
ARM: dts: motorola-cpcap-mapphone: add audio-codec
ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio
ARM: dts: omap4-droid4: add soundcard
Tony Lindgren (1):
ALSA: pcm: Export soc_dpcm_runtime_update
.../bindings/sound/motorola,cpcap-audio-codec.txt | 19 +
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 +
arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++
sound/soc/codecs/Kconfig | 5 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cpcap.c | 1594 ++++++++++++++++++++
sound/soc/soc-pcm.c | 2 +
7 files changed, 1713 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
create mode 100644 sound/soc/codecs/cpcap.c
--
2.13.2
^ permalink raw reply [flat|nested] 22+ messages in thread
* [PATCHv3 0/6] Motorola Droid 4 Audio Support
@ 2017-07-25 15:10 ` Sebastian Reichel
0 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel-K7yf7f+aM1XWsZ/bQMPhNw,
linux-omap-u79uwXL29TY76Z2rM5mHXA,
devicetree-u79uwXL29TY76Z2rM5mHXA,
linux-kernel-u79uwXL29TY76Z2rM5mHXA, Sebastian Reichel
Hi,
This adds audio support to Motorola Droid 4.
Tested:
- Playing via both DACs using Speaker, Earpiece, Headphone
- Recording using internal Mic
- Volume Controls
Known not to work:
- 3.5mm detection support (requires closed source firmware,
needs further investigation)
- Modem / Bluetooth Audio (actually untested, but probably
needs some quirks)
Changes since PATCHv2:
* Fix a whitespace issue
* Fix const notes Takashi provided
* Fix a DAPM route issue I accidently introduced in v2
Changes since PATCHv1:
* Add patch from Tony exporting soc_dpcm_runtime_update
* Integrate DT patch for vaudio initial mode
* Split dt-binding from codec patch and add Ack from Rob
* Fix CPCAP position in Kconfig/Makefile
* Avoid "err +=" constructs
* Simplify reset function
* Drop cpcap_audio_read/write helpers
* Do not use tertiary operator for mute register value
* Update Input Mux logic
* Switch from simple-audio-card to audio-graph-card
-- Sebastian
Sebastian Reichel (5):
dt-bindings: sound: add motorola,cpcap-audio-codec
ASoC: codec: cpcap: new codec
ARM: dts: motorola-cpcap-mapphone: add audio-codec
ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio
ARM: dts: omap4-droid4: add soundcard
Tony Lindgren (1):
ALSA: pcm: Export soc_dpcm_runtime_update
.../bindings/sound/motorola,cpcap-audio-codec.txt | 19 +
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 7 +
arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++
sound/soc/codecs/Kconfig | 5 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cpcap.c | 1594 ++++++++++++++++++++
sound/soc/soc-pcm.c | 2 +
7 files changed, 1713 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
create mode 100644 sound/soc/codecs/cpcap.c
--
2.13.2
--
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^ permalink raw reply [flat|nested] 22+ messages in thread
* [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
2017-07-25 15:10 ` Sebastian Reichel
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
2017-07-25 15:42 ` Takashi Iwai
-1 siblings, 1 reply; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
From: Tony Lindgren <tony@atomide.com>
Some codecs may need to use this from loadable modules. Without
this patch compilation for this kind of codec will fail with the
following error:
ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
undefined!
Signed-off-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
sound/soc/soc-pcm.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index dcc5ece08668..4de7d86ea6fb 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2539,6 +2539,8 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
mutex_unlock(&card->mutex);
return 0;
}
+EXPORT_SYMBOL_GPL(soc_dpcm_runtime_update);
+
int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
{
struct snd_soc_dpcm *dpcm;
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
2017-07-25 15:10 ` Sebastian Reichel
(?)
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
2017-07-26 11:48 ` Mark Brown
-1 siblings, 1 reply; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.
This adds the DT binding for the codec sub-module found inside
the PMIC.
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
.../bindings/sound/motorola,cpcap-audio-codec.txt | 19 +++++++++++++++++++
1 file changed, 19 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
diff --git a/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt b/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
new file mode 100644
index 000000000000..6b8cd616bd46
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/motorola,cpcap-audio-codec.txt
@@ -0,0 +1,19 @@
+Motorola CPCAP audio CODEC
+--------------------------
+
+This module is part of the CPCAP. For more details about the whole
+chip see Documentation/devicetree/bindings/mfd/motorola-cpcap.txt.
+
+Required properties:
+
+ - compatible : "motorola,cpcap-audio-codec"
+ - vdd-supply : Phandle to audio regulator
+
+Example:
+
+&cpcap {
+ audio-codec {
+ compatible = "motorola,cpcap-audio-codec";
+ vdd-supply = <&vaudio>;
+ };
+};
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* [PATCHv3 3/6] ASoC: codec: cpcap: new codec
2017-07-25 15:10 ` Sebastian Reichel
` (2 preceding siblings ...)
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
-1 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Motorola CPCAP is a PMIC with audio functionality, that can be
found on Motorola Droid 4 and probably a few other phones from
Motorola's Droid series.
The driver has been written from scratch using Motorola's Android
driver, register dumps from running Android and datasheet for NXP
MC13783UG (which is similar to Motorola CPCAP, but not the same).
The chip provides two audio interfaces, that can be muxed to two
different audio codecs. One provides support for stereo output
(named StDAC or HiFi), while the other only provides mono output
(named Voice). Only the Voice codec provides a Capture interface.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
sound/soc/codecs/Kconfig | 5 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/cpcap.c | 1594 +++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 1601 insertions(+)
create mode 100644 sound/soc/codecs/cpcap.c
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 6c78b0b49b81..1615e5acc8fd 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -401,6 +401,11 @@ config SND_SOC_ALC5632
config SND_SOC_BT_SCO
tristate "Dummy BT SCO codec driver"
+config SND_SOC_CPCAP
+ tristate "Motorola CPCAP codec"
+ depends on MFD_CPCAP
+ default MFD_CPCAP
+
config SND_SOC_CQ0093VC
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 1755a54e3dc9..607f9ce84bc4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -35,6 +35,7 @@ snd-soc-ak4671-objs := ak4671.o
snd-soc-ak5386-objs := ak5386.o
snd-soc-arizona-objs := arizona.o
snd-soc-bt-sco-objs := bt-sco.o
+snd-soc-cpcap-objs := cpcap.o
snd-soc-cq93vc-objs := cq93vc.o
snd-soc-cs35l32-objs := cs35l32.o
snd-soc-cs35l33-objs := cs35l33.o
@@ -273,6 +274,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o
obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
+obj-$(CONFIG_SND_SOC_CPCAP) += snd-soc-cpcap.o
obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o
obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o
obj-$(CONFIG_SND_SOC_CS35L34) += snd-soc-cs35l34.o
diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c
new file mode 100644
index 000000000000..f52a6b7b1737
--- /dev/null
+++ b/sound/soc/codecs/cpcap.c
@@ -0,0 +1,1594 @@
+/*
+ * ALSA SoC CPCAP codec driver
+ *
+ * Copyright (C) 2017 Sebastian Reichel <sre@kernel.org>
+ *
+ * Very loosely based on original driver from Motorola:
+ * Copyright (C) 2007 - 2009 Motorola, 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/motorola-cpcap.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+/* Register 513 CPCAP_REG_CC --- CODEC */
+#define CPCAP_BIT_CDC_CLK2 15
+#define CPCAP_BIT_CDC_CLK1 14
+#define CPCAP_BIT_CDC_CLK0 13
+#define CPCAP_BIT_CDC_SR3 12
+#define CPCAP_BIT_CDC_SR2 11
+#define CPCAP_BIT_CDC_SR1 10
+#define CPCAP_BIT_CDC_SR0 9
+#define CPCAP_BIT_CDC_CLOCK_TREE_RESET 8
+#define CPCAP_BIT_MIC2_CDC_EN 7
+#define CPCAP_BIT_CDC_EN_RX 6
+#define CPCAP_BIT_DF_RESET 5
+#define CPCAP_BIT_MIC1_CDC_EN 4
+#define CPCAP_BIT_AUDOHPF_1 3
+#define CPCAP_BIT_AUDOHPF_0 2
+#define CPCAP_BIT_AUDIHPF_1 1
+#define CPCAP_BIT_AUDIHPF_0 0
+
+/* Register 514 CPCAP_REG_CDI --- CODEC Digital Audio Interface */
+#define CPCAP_BIT_CDC_PLL_SEL 15
+#define CPCAP_BIT_CLK_IN_SEL 13
+#define CPCAP_BIT_DIG_AUD_IN 12
+#define CPCAP_BIT_CDC_CLK_EN 11
+#define CPCAP_BIT_CDC_DIG_AUD_FS1 10
+#define CPCAP_BIT_CDC_DIG_AUD_FS0 9
+#define CPCAP_BIT_MIC2_TIMESLOT2 8
+#define CPCAP_BIT_MIC2_TIMESLOT1 7
+#define CPCAP_BIT_MIC2_TIMESLOT0 6
+#define CPCAP_BIT_MIC1_RX_TIMESLOT2 5
+#define CPCAP_BIT_MIC1_RX_TIMESLOT1 4
+#define CPCAP_BIT_MIC1_RX_TIMESLOT0 3
+#define CPCAP_BIT_FS_INV 2
+#define CPCAP_BIT_CLK_INV 1
+#define CPCAP_BIT_SMB_CDC 0
+
+/* Register 515 CPCAP_REG_SDAC --- Stereo DAC */
+#define CPCAP_BIT_FSYNC_CLK_IN_COMMON 11
+#define CPCAP_BIT_SLAVE_PLL_CLK_INPUT 10
+#define CPCAP_BIT_ST_CLOCK_TREE_RESET 9
+#define CPCAP_BIT_DF_RESET_ST_DAC 8
+#define CPCAP_BIT_ST_SR3 7
+#define CPCAP_BIT_ST_SR2 6
+#define CPCAP_BIT_ST_SR1 5
+#define CPCAP_BIT_ST_SR0 4
+#define CPCAP_BIT_ST_DAC_CLK2 3
+#define CPCAP_BIT_ST_DAC_CLK1 2
+#define CPCAP_BIT_ST_DAC_CLK0 1
+#define CPCAP_BIT_ST_DAC_EN 0
+
+/* Register 516 CPCAP_REG_SDACDI --- Stereo DAC Digital Audio Interface */
+#define CPCAP_BIT_ST_L_TIMESLOT2 13
+#define CPCAP_BIT_ST_L_TIMESLOT1 12
+#define CPCAP_BIT_ST_L_TIMESLOT0 11
+#define CPCAP_BIT_ST_R_TIMESLOT2 10
+#define CPCAP_BIT_ST_R_TIMESLOT1 9
+#define CPCAP_BIT_ST_R_TIMESLOT0 8
+#define CPCAP_BIT_ST_DAC_CLK_IN_SEL 7
+#define CPCAP_BIT_ST_FS_INV 6
+#define CPCAP_BIT_ST_CLK_INV 5
+#define CPCAP_BIT_ST_DIG_AUD_FS1 4
+#define CPCAP_BIT_ST_DIG_AUD_FS0 3
+#define CPCAP_BIT_DIG_AUD_IN_ST_DAC 2
+#define CPCAP_BIT_ST_CLK_EN 1
+#define CPCAP_BIT_SMB_ST_DAC 0
+
+/* Register 517 CPCAP_REG_TXI --- TX Interface */
+#define CPCAP_BIT_PTT_TH 15
+#define CPCAP_BIT_PTT_CMP_EN 14
+#define CPCAP_BIT_HS_ID_TX 13
+#define CPCAP_BIT_MB_ON2 12
+#define CPCAP_BIT_MB_ON1L 11
+#define CPCAP_BIT_MB_ON1R 10
+#define CPCAP_BIT_RX_L_ENCODE 9
+#define CPCAP_BIT_RX_R_ENCODE 8
+#define CPCAP_BIT_MIC2_MUX 7
+#define CPCAP_BIT_MIC2_PGA_EN 6
+#define CPCAP_BIT_CDET_DIS 5
+#define CPCAP_BIT_EMU_MIC_MUX 4
+#define CPCAP_BIT_HS_MIC_MUX 3
+#define CPCAP_BIT_MIC1_MUX 2
+#define CPCAP_BIT_MIC1_PGA_EN 1
+#define CPCAP_BIT_DLM 0
+
+/* Register 518 CPCAP_REG_TXMP --- Mic Gain */
+#define CPCAP_BIT_MB_BIAS_R1 11
+#define CPCAP_BIT_MB_BIAS_R0 10
+#define CPCAP_BIT_MIC2_GAIN_4 9
+#define CPCAP_BIT_MIC2_GAIN_3 8
+#define CPCAP_BIT_MIC2_GAIN_2 7
+#define CPCAP_BIT_MIC2_GAIN_1 6
+#define CPCAP_BIT_MIC2_GAIN_0 5
+#define CPCAP_BIT_MIC1_GAIN_4 4
+#define CPCAP_BIT_MIC1_GAIN_3 3
+#define CPCAP_BIT_MIC1_GAIN_2 2
+#define CPCAP_BIT_MIC1_GAIN_1 1
+#define CPCAP_BIT_MIC1_GAIN_0 0
+
+/* Register 519 CPCAP_REG_RXOA --- RX Output Amplifier */
+#define CPCAP_BIT_UNUSED_519_15 15
+#define CPCAP_BIT_UNUSED_519_14 14
+#define CPCAP_BIT_UNUSED_519_13 13
+#define CPCAP_BIT_STDAC_LOW_PWR_DISABLE 12
+#define CPCAP_BIT_HS_LOW_PWR 11
+#define CPCAP_BIT_HS_ID_RX 10
+#define CPCAP_BIT_ST_HS_CP_EN 9
+#define CPCAP_BIT_EMU_SPKR_R_EN 8
+#define CPCAP_BIT_EMU_SPKR_L_EN 7
+#define CPCAP_BIT_HS_L_EN 6
+#define CPCAP_BIT_HS_R_EN 5
+#define CPCAP_BIT_A4_LINEOUT_L_EN 4
+#define CPCAP_BIT_A4_LINEOUT_R_EN 3
+#define CPCAP_BIT_A2_LDSP_L_EN 2
+#define CPCAP_BIT_A2_LDSP_R_EN 1
+#define CPCAP_BIT_A1_EAR_EN 0
+
+/* Register 520 CPCAP_REG_RXVC --- RX Volume Control */
+#define CPCAP_BIT_VOL_EXT3 15
+#define CPCAP_BIT_VOL_EXT2 14
+#define CPCAP_BIT_VOL_EXT1 13
+#define CPCAP_BIT_VOL_EXT0 12
+#define CPCAP_BIT_VOL_DAC3 11
+#define CPCAP_BIT_VOL_DAC2 10
+#define CPCAP_BIT_VOL_DAC1 9
+#define CPCAP_BIT_VOL_DAC0 8
+#define CPCAP_BIT_VOL_DAC_LSB_1dB1 7
+#define CPCAP_BIT_VOL_DAC_LSB_1dB0 6
+#define CPCAP_BIT_VOL_CDC3 5
+#define CPCAP_BIT_VOL_CDC2 4
+#define CPCAP_BIT_VOL_CDC1 3
+#define CPCAP_BIT_VOL_CDC0 2
+#define CPCAP_BIT_VOL_CDC_LSB_1dB1 1
+#define CPCAP_BIT_VOL_CDC_LSB_1dB0 0
+
+/* Register 521 CPCAP_REG_RXCOA --- Codec to Output Amp Switches */
+#define CPCAP_BIT_PGA_CDC_EN 10
+#define CPCAP_BIT_CDC_SW 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_CDC_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_CDC_SW 7
+#define CPCAP_BIT_ALEFT_HS_CDC_SW 6
+#define CPCAP_BIT_ARIGHT_HS_CDC_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_CDC_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_CDC_SW 3
+#define CPCAP_BIT_A2_LDSP_L_CDC_SW 2
+#define CPCAP_BIT_A2_LDSP_R_CDC_SW 1
+#define CPCAP_BIT_A1_EAR_CDC_SW 0
+
+/* Register 522 CPCAP_REG_RXSDOA --- RX Stereo DAC to Output Amp Switches */
+#define CPCAP_BIT_PGA_DAC_EN 12
+#define CPCAP_BIT_ST_DAC_SW 11
+#define CPCAP_BIT_MONO_DAC1 10
+#define CPCAP_BIT_MONO_DAC0 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_DAC_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_DAC_SW 7
+#define CPCAP_BIT_ALEFT_HS_DAC_SW 6
+#define CPCAP_BIT_ARIGHT_HS_DAC_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_DAC_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_DAC_SW 3
+#define CPCAP_BIT_A2_LDSP_L_DAC_SW 2
+#define CPCAP_BIT_A2_LDSP_R_DAC_SW 1
+#define CPCAP_BIT_A1_EAR_DAC_SW 0
+
+/* Register 523 CPCAP_REG_RXEPOA --- RX External PGA to Output Amp Switches */
+#define CPCAP_BIT_PGA_EXT_L_EN 14
+#define CPCAP_BIT_PGA_EXT_R_EN 13
+#define CPCAP_BIT_PGA_IN_L_SW 12
+#define CPCAP_BIT_PGA_IN_R_SW 11
+#define CPCAP_BIT_MONO_EXT1 10
+#define CPCAP_BIT_MONO_EXT0 9
+#define CPCAP_BIT_PGA_OUTR_USBDP_EXT_SW 8
+#define CPCAP_BIT_PGA_OUTL_USBDN_EXT_SW 7
+#define CPCAP_BIT_ALEFT_HS_EXT_SW 6
+#define CPCAP_BIT_ARIGHT_HS_EXT_SW 5
+#define CPCAP_BIT_A4_LINEOUT_L_EXT_SW 4
+#define CPCAP_BIT_A4_LINEOUT_R_EXT_SW 3
+#define CPCAP_BIT_A2_LDSP_L_EXT_SW 2
+#define CPCAP_BIT_A2_LDSP_R_EXT_SW 1
+#define CPCAP_BIT_A1_EAR_EXT_SW 0
+
+/* Register 525 CPCAP_REG_A2LA --- SPK Amplifier and Clock Config for Headset */
+#define CPCAP_BIT_NCP_CLK_SYNC 7
+#define CPCAP_BIT_A2_CLK_SYNC 6
+#define CPCAP_BIT_A2_FREE_RUN 5
+#define CPCAP_BIT_A2_CLK2 4
+#define CPCAP_BIT_A2_CLK1 3
+#define CPCAP_BIT_A2_CLK0 2
+#define CPCAP_BIT_A2_CLK_IN 1
+#define CPCAP_BIT_A2_CONFIG 0
+
+#define SLEEP_ACTIVATE_POWER 2
+#define CLOCK_TREE_RESET_TIME 1
+
+/* constants for ST delay workaround */
+#define STM_STDAC_ACTIVATE_RAMP_TIME 1
+#define STM_STDAC_EN_TEST_PRE 0x090C
+#define STM_STDAC_EN_TEST_POST 0x0000
+#define STM_STDAC_EN_ST_TEST1_PRE 0x2400
+#define STM_STDAC_EN_ST_TEST1_POST 0x0400
+
+struct cpcap_reg_info {
+ u16 reg;
+ u16 mask;
+ u16 val;
+};
+
+static const struct cpcap_reg_info cpcap_default_regs[] = {
+ { CPCAP_REG_CC, 0xFFFF, 0x0000 },
+ { CPCAP_REG_CC, 0xFFFF, 0x0000 },
+ { CPCAP_REG_CDI, 0xBFFF, 0x0000 },
+ { CPCAP_REG_SDAC, 0x0FFF, 0x0000 },
+ { CPCAP_REG_SDACDI, 0x3FFF, 0x0000 },
+ { CPCAP_REG_TXI, 0x0FDF, 0x0000 },
+ { CPCAP_REG_TXMP, 0x0FFF, 0x0400 },
+ { CPCAP_REG_RXOA, 0x01FF, 0x0000 },
+ { CPCAP_REG_RXVC, 0xFF3C, 0x0000 },
+ { CPCAP_REG_RXCOA, 0x07FF, 0x0000 },
+ { CPCAP_REG_RXSDOA, 0x1FFF, 0x0000 },
+ { CPCAP_REG_RXEPOA, 0x7FFF, 0x0000 },
+ { CPCAP_REG_A2LA, BIT(CPCAP_BIT_A2_FREE_RUN),
+ BIT(CPCAP_BIT_A2_FREE_RUN) },
+};
+
+enum cpcap_dai {
+ CPCAP_DAI_HIFI,
+ CPCAP_DAI_VOICE,
+};
+
+struct cpcap_audio {
+ struct snd_soc_codec *codec;
+ struct regmap *regmap;
+
+ u16 vendor;
+
+ int codec_clk_id;
+ int codec_freq;
+ int codec_format;
+};
+
+static int cpcap_st_workaround(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int err = 0;
+
+ /* Only CPCAP from ST requires workaround */
+ if (cpcap->vendor != CPCAP_VENDOR_ST)
+ return 0;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ err = regmap_write(cpcap->regmap, CPCAP_REG_TEST,
+ STM_STDAC_EN_TEST_PRE);
+ if (err)
+ return err;
+ err = regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
+ STM_STDAC_EN_ST_TEST1_PRE);
+ break;
+ case SND_SOC_DAPM_POST_PMU:
+ msleep(STM_STDAC_ACTIVATE_RAMP_TIME);
+
+ err = regmap_write(cpcap->regmap, CPCAP_REG_ST_TEST1,
+ STM_STDAC_EN_ST_TEST1_POST);
+ if (err)
+ return err;
+ err = regmap_write(cpcap->regmap, CPCAP_REG_TEST,
+ STM_STDAC_EN_TEST_POST);
+ break;
+ default:
+ break;
+ }
+
+ return err;
+}
+
+/* Capture Gain Control: 0dB to 31dB in 1dB steps */
+static const DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+/* Playback Gain Control: -33dB to 12dB in 3dB steps */
+static const DECLARE_TLV_DB_SCALE(vol_tlv, -3300, 300, 0);
+
+static const char * const cpcap_phase_invert_texts[] = {
+ "Non Inverted", "Inverted"
+};
+static SOC_ENUM_SINGLE_DECL(cpcap_hifi_l_phase_inv_enum,
+ CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC0, cpcap_phase_invert_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_ext_l_phase_inv_enum,
+ CPCAP_REG_RXEPOA, CPCAP_BIT_MONO_EXT0, cpcap_phase_invert_texts);
+
+static const struct snd_kcontrol_new cpcap_snd_controls[] = {
+ /* Playback Gain */
+ SOC_SINGLE_TLV("HiFi Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_DAC0, 0xF, 0, vol_tlv),
+ SOC_SINGLE_TLV("Voice Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_CDC0, 0xF, 0, vol_tlv),
+ SOC_SINGLE_TLV("Ext Playback Volume",
+ CPCAP_REG_RXVC, CPCAP_BIT_VOL_EXT0, 0xF, 0, vol_tlv),
+
+ /* Capture Gain */
+ SOC_SINGLE_TLV("Mic1 Capture Volume",
+ CPCAP_REG_TXMP, CPCAP_BIT_MIC1_GAIN_0, 0x1F, 0, mic_gain_tlv),
+ SOC_SINGLE_TLV("Mic2 Capture Volume",
+ CPCAP_REG_TXMP, CPCAP_BIT_MIC2_GAIN_0, 0x1F, 0, mic_gain_tlv),
+
+ SOC_ENUM("Hifi Left Phase Playback Switch",
+ cpcap_hifi_l_phase_inv_enum),
+ SOC_ENUM("Ext Left Phase Playback Switch",
+ cpcap_ext_l_phase_inv_enum),
+};
+
+static const char * const cpcap_out_mux_texts[] = {
+ "Off", "Voice", "HiFi", "Ext"
+};
+
+static const char * const cpcap_in_right_mux_texts[] = {
+ "Off", "Mic 1", "Headset Mic", "EMU Mic", "Ext Right"
+};
+
+static const char * const cpcap_in_left_mux_texts[] = {
+ "Off", "Mic 2", "Ext Left"
+};
+
+/*
+ * input muxes use unusual register layout, so that we need to use custom
+ * getter/setter methods
+ */
+static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_left_mux_enum,
+ cpcap_in_left_mux_texts);
+static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_right_mux_enum,
+ cpcap_in_right_mux_texts);
+
+/*
+ * mux uses same bit in CPCAP_REG_RXCOA, CPCAP_REG_RXSDOA & CPCAP_REG_RXEPOA;
+ * even though the register layout makes it look like a mixer, this is a mux.
+ * Enabling multiple inputs will result in no audio being forwarded.
+ */
+static SOC_ENUM_SINGLE_DECL(cpcap_earpiece_mux_enum, 0, 0, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_spkr_r_mux_enum, 0, 1, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_spkr_l_mux_enum, 0, 2, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_line_r_mux_enum, 0, 3, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_line_l_mux_enum, 0, 4, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_hs_r_mux_enum, 0, 5, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_hs_l_mux_enum, 0, 6, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_emu_l_mux_enum, 0, 7, cpcap_out_mux_texts);
+static SOC_ENUM_SINGLE_DECL(cpcap_emu_r_mux_enum, 0, 8, cpcap_out_mux_texts);
+
+static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int shift = e->shift_l;
+ int reg_voice, reg_hifi, reg_ext, status;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXCOA, ®_voice);
+ if (err)
+ return err;
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXSDOA, ®_hifi);
+ if (err)
+ return err;
+ err = regmap_read(cpcap->regmap, CPCAP_REG_RXEPOA, ®_ext);
+ if (err)
+ return err;
+
+ reg_voice = (reg_voice >> shift) & 1;
+ reg_hifi = (reg_hifi >> shift) & 1;
+ reg_ext = (reg_ext >> shift) & 1;
+ status = reg_ext << 2 | reg_hifi << 1 | reg_voice;
+
+ switch (status) {
+ case 0x04:
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case 0x02:
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case 0x01:
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ unsigned int mask = BIT(e->shift_l);
+ u16 reg_voice = 0x00, reg_hifi = 0x00, reg_ext = 0x00;
+ int err;
+
+ switch (muxval) {
+ case 1:
+ reg_voice = mask;
+ break;
+ case 2:
+ reg_hifi = mask;
+ break;
+ case 3:
+ reg_ext = mask;
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA,
+ mask, reg_voice);
+ if (err)
+ return err;
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXSDOA,
+ mask, reg_hifi);
+ if (err)
+ return err;
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXEPOA,
+ mask, reg_ext);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+ soc_dpcm_runtime_update(card);
+
+ return 0;
+}
+
+static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int regval, mask;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_TXI, ®val);
+ if (err)
+ return err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC1_MUX);
+ mask |= BIT(CPCAP_BIT_HS_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_EMU_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_RX_R_ENCODE);
+
+ switch (regval & mask) {
+ case BIT(CPCAP_BIT_RX_R_ENCODE):
+ ucontrol->value.enumerated.item[0] = 4;
+ break;
+ case BIT(CPCAP_BIT_EMU_MIC_MUX):
+ ucontrol->value.enumerated.item[0] = 3;
+ break;
+ case BIT(CPCAP_BIT_HS_MIC_MUX):
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case BIT(CPCAP_BIT_MIC1_MUX):
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ int regval = 0, mask;
+ int err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC1_MUX);
+ mask |= BIT(CPCAP_BIT_HS_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_EMU_MIC_MUX);
+ mask |= BIT(CPCAP_BIT_RX_R_ENCODE);
+
+ switch (muxval) {
+ case 1:
+ regval = BIT(CPCAP_BIT_MIC1_MUX);
+ break;
+ case 2:
+ regval = BIT(CPCAP_BIT_HS_MIC_MUX);
+ break;
+ case 3:
+ regval = BIT(CPCAP_BIT_EMU_MIC_MUX);
+ break;
+ case 4:
+ regval = BIT(CPCAP_BIT_RX_R_ENCODE);
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+ mask, regval);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+ soc_dpcm_runtime_update(card);
+
+ return 0;
+}
+
+static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int regval, mask;
+ int err;
+
+ err = regmap_read(cpcap->regmap, CPCAP_REG_TXI, ®val);
+ if (err)
+ return err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC2_MUX);
+ mask |= BIT(CPCAP_BIT_RX_L_ENCODE);
+
+ switch (regval & mask) {
+ case BIT(CPCAP_BIT_RX_L_ENCODE):
+ ucontrol->value.enumerated.item[0] = 2;
+ break;
+ case BIT(CPCAP_BIT_MIC2_MUX):
+ ucontrol->value.enumerated.item[0] = 1;
+ break;
+ default:
+ ucontrol->value.enumerated.item[0] = 0;
+ break;
+ }
+
+ return 0;
+}
+
+static int cpcap_input_left_mux_put_enum(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_card *card = dapm->card;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned int muxval = ucontrol->value.enumerated.item[0];
+ int regval = 0, mask;
+ int err;
+
+ mask = 0;
+ mask |= BIT(CPCAP_BIT_MIC2_MUX);
+ mask |= BIT(CPCAP_BIT_RX_L_ENCODE);
+
+ switch (muxval) {
+ case 1:
+ regval = BIT(CPCAP_BIT_MIC2_MUX);
+ break;
+ case 2:
+ regval = BIT(CPCAP_BIT_RX_L_ENCODE);
+ break;
+ default:
+ break;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI,
+ mask, regval);
+ if (err)
+ return err;
+
+ snd_soc_dapm_mux_update_power(dapm, kcontrol, muxval, e, NULL);
+ soc_dpcm_runtime_update(card);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new cpcap_input_left_mux =
+ SOC_DAPM_ENUM_EXT("Input Left", cpcap_input_left_mux_enum,
+ cpcap_input_left_mux_get_enum,
+ cpcap_input_left_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_input_right_mux =
+ SOC_DAPM_ENUM_EXT("Input Right", cpcap_input_right_mux_enum,
+ cpcap_input_right_mux_get_enum,
+ cpcap_input_right_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_emu_left_mux =
+ SOC_DAPM_ENUM_EXT("EMU Left", cpcap_emu_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_emu_right_mux =
+ SOC_DAPM_ENUM_EXT("EMU Right", cpcap_emu_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_hs_left_mux =
+ SOC_DAPM_ENUM_EXT("Headset Left", cpcap_hs_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_hs_right_mux =
+ SOC_DAPM_ENUM_EXT("Headset Right", cpcap_hs_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_line_left_mux =
+ SOC_DAPM_ENUM_EXT("Line Left", cpcap_line_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_line_right_mux =
+ SOC_DAPM_ENUM_EXT("Line Right", cpcap_line_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_speaker_left_mux =
+ SOC_DAPM_ENUM_EXT("Speaker Left", cpcap_spkr_l_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_speaker_right_mux =
+ SOC_DAPM_ENUM_EXT("Speaker Right", cpcap_spkr_r_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+static const struct snd_kcontrol_new cpcap_earpiece_mux =
+ SOC_DAPM_ENUM_EXT("Earpiece", cpcap_earpiece_mux_enum,
+ cpcap_output_mux_get_enum, cpcap_output_mux_put_enum);
+
+static const struct snd_kcontrol_new cpcap_hifi_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("HiFi Mono Playback Switch",
+ CPCAP_REG_RXSDOA, CPCAP_BIT_MONO_DAC1, 1, 0),
+};
+static const struct snd_kcontrol_new cpcap_ext_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Ext Mono Playback Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_MONO_EXT0, 1, 0),
+};
+
+static const struct snd_kcontrol_new cpcap_extr_mute_control =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_IN_R_SW, 1, 0);
+static const struct snd_kcontrol_new cpcap_extl_mute_control =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_IN_L_SW, 1, 0);
+
+static const struct snd_kcontrol_new cpcap_voice_loopback =
+ SOC_DAPM_SINGLE("Switch",
+ CPCAP_REG_TXI, CPCAP_BIT_DLM, 1, 0);
+
+static const struct snd_soc_dapm_widget cpcap_dapm_widgets[] = {
+ /* DAIs */
+ SND_SOC_DAPM_AIF_IN("HiFi RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("Voice RX", NULL, 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("Voice TX", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+ /* Power Supply */
+ SND_SOC_DAPM_REGULATOR_SUPPLY("vdd", SLEEP_ACTIVATE_POWER, 0),
+
+ /* Highpass Filters */
+ SND_SOC_DAPM_REG(snd_soc_dapm_pga, "Highpass Filter RX",
+ CPCAP_REG_CC, CPCAP_BIT_AUDIHPF_0, 0x3, 0x3, 0x0),
+ SND_SOC_DAPM_REG(snd_soc_dapm_pga, "Highpass Filter TX",
+ CPCAP_REG_CC, CPCAP_BIT_AUDOHPF_0, 0x3, 0x3, 0x0),
+
+ /* Clocks */
+ SND_SOC_DAPM_SUPPLY("HiFi DAI Clock",
+ CPCAP_REG_SDACDI, CPCAP_BIT_ST_CLK_EN, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Voice DAI Clock",
+ CPCAP_REG_CDI, CPCAP_BIT_CDC_CLK_EN, 0, NULL, 0),
+
+ /* Microphone Bias */
+ SND_SOC_DAPM_SUPPLY("MIC1R Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON1R, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC1L Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON1L, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("MIC2 Bias",
+ CPCAP_REG_TXI, CPCAP_BIT_MB_ON2, 0, NULL, 0),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MICR"),
+ SND_SOC_DAPM_INPUT("HSMIC"),
+ SND_SOC_DAPM_INPUT("EMUMIC"),
+ SND_SOC_DAPM_INPUT("MICL"),
+ SND_SOC_DAPM_INPUT("EXTR"),
+ SND_SOC_DAPM_INPUT("EXTL"),
+
+ /* Capture Route */
+ SND_SOC_DAPM_MUX("Right Capture Route",
+ SND_SOC_NOPM, 0, 0, &cpcap_input_right_mux),
+ SND_SOC_DAPM_MUX("Left Capture Route",
+ SND_SOC_NOPM, 0, 0, &cpcap_input_left_mux),
+
+ /* Capture PGAs */
+ SND_SOC_DAPM_PGA("Microphone 1 PGA",
+ CPCAP_REG_TXI, CPCAP_BIT_MIC1_PGA_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Microphone 2 PGA",
+ CPCAP_REG_TXI, CPCAP_BIT_MIC2_PGA_EN, 0, NULL, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC Right", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_MIC1_CDC_EN, 0),
+ SND_SOC_DAPM_ADC("ADC Left", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_MIC2_CDC_EN, 0),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC_E("DAC HiFi", NULL,
+ CPCAP_REG_SDAC, CPCAP_BIT_ST_DAC_EN, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_DAC_E("DAC Voice", NULL,
+ CPCAP_REG_CC, CPCAP_BIT_CDC_EN_RX, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+ /* Playback PGA */
+ SND_SOC_DAPM_PGA("HiFi PGA",
+ CPCAP_REG_RXSDOA, CPCAP_BIT_PGA_DAC_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Voice PGA",
+ CPCAP_REG_RXCOA, CPCAP_BIT_PGA_CDC_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA_E("Ext Right PGA",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_EXT_R_EN, 0,
+ NULL, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+ SND_SOC_DAPM_PGA_E("Ext Left PGA",
+ CPCAP_REG_RXEPOA, CPCAP_BIT_PGA_EXT_L_EN, 0,
+ NULL, 0,
+ cpcap_st_workaround,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+
+ /* Playback Switch */
+ SND_SOC_DAPM_SWITCH("Ext Right Enable", SND_SOC_NOPM, 0, 0,
+ &cpcap_extr_mute_control),
+ SND_SOC_DAPM_SWITCH("Ext Left Enable", SND_SOC_NOPM, 0, 0,
+ &cpcap_extl_mute_control),
+
+ /* Loopback Switch */
+ SND_SOC_DAPM_SWITCH("Voice Loopback", SND_SOC_NOPM, 0, 0,
+ &cpcap_voice_loopback),
+
+ /* Mono Mixer */
+ SOC_MIXER_ARRAY("HiFi Mono Left Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_hifi_mono_mixer_controls),
+ SOC_MIXER_ARRAY("HiFi Mono Right Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_hifi_mono_mixer_controls),
+ SOC_MIXER_ARRAY("Ext Mono Left Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_ext_mono_mixer_controls),
+ SOC_MIXER_ARRAY("Ext Mono Right Mixer", SND_SOC_NOPM, 0, 0,
+ cpcap_ext_mono_mixer_controls),
+
+ /* Output Routes */
+ SND_SOC_DAPM_MUX("Earpiece Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_earpiece_mux),
+ SND_SOC_DAPM_MUX("Speaker Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_speaker_right_mux),
+ SND_SOC_DAPM_MUX("Speaker Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_speaker_left_mux),
+ SND_SOC_DAPM_MUX("Lineout Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_line_right_mux),
+ SND_SOC_DAPM_MUX("Lineout Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_line_left_mux),
+ SND_SOC_DAPM_MUX("Headset Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_hs_right_mux),
+ SND_SOC_DAPM_MUX("Headset Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_hs_left_mux),
+ SND_SOC_DAPM_MUX("EMU Right Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_emu_right_mux),
+ SND_SOC_DAPM_MUX("EMU Left Playback Route", SND_SOC_NOPM, 0, 0,
+ &cpcap_emu_left_mux),
+
+ /* Output Amplifier */
+ SND_SOC_DAPM_PGA("Earpiece PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A1_EAR_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Speaker Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A2_LDSP_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Speaker Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A2_LDSP_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lineout Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A4_LINEOUT_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Lineout Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_A4_LINEOUT_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Headset Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_HS_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Headset Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_HS_L_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("EMU Right PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_R_EN, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("EMU Left PGA",
+ CPCAP_REG_RXOA, CPCAP_BIT_EMU_SPKR_L_EN, 0, NULL, 0),
+
+ /* Headet Charge Pump */
+ SND_SOC_DAPM_SUPPLY("Headset Charge Pump",
+ CPCAP_REG_RXOA, CPCAP_BIT_ST_HS_CP_EN, 0, NULL, 0),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("EP"),
+ SND_SOC_DAPM_OUTPUT("SPKR"),
+ SND_SOC_DAPM_OUTPUT("SPKL"),
+ SND_SOC_DAPM_OUTPUT("LINER"),
+ SND_SOC_DAPM_OUTPUT("LINEL"),
+ SND_SOC_DAPM_OUTPUT("HSR"),
+ SND_SOC_DAPM_OUTPUT("HSL"),
+ SND_SOC_DAPM_OUTPUT("EMUR"),
+ SND_SOC_DAPM_OUTPUT("EMUL"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* Power Supply */
+ {"HiFi PGA", NULL, "vdd"},
+ {"Voice PGA", NULL, "vdd"},
+ {"Ext Right PGA", NULL, "vdd"},
+ {"Ext Left PGA", NULL, "vdd"},
+ {"Microphone 1 PGA", NULL, "vdd"},
+ {"Microphone 2 PGA", NULL, "vdd"},
+
+ /* Stream -> AIF */
+ {"HiFi RX", NULL, "HiFi Playback"},
+ {"Voice RX", NULL, "Voice Playback"},
+ {"Voice Capture", NULL, "Voice TX"},
+
+ /* AIF clocks */
+ {"HiFi RX", NULL, "HiFi DAI Clock"},
+ {"Voice RX", NULL, "Voice DAI Clock"},
+ {"Voice TX", NULL, "Voice DAI Clock"},
+
+ /* Digital Loopback */
+ {"Voice Loopback", "Switch", "Voice TX"},
+ {"Voice RX", NULL, "Voice Loopback"},
+
+ /* Highpass Filters */
+ {"Highpass Filter RX", NULL, "Voice RX"},
+ {"Voice TX", NULL, "Highpass Filter TX"},
+
+ /* AIF -> DAC mapping */
+ {"DAC HiFi", NULL, "HiFi RX"},
+ {"DAC Voice", NULL, "Highpass Filter RX"},
+
+ /* DAC -> PGA */
+ {"HiFi PGA", NULL, "DAC HiFi"},
+ {"Voice PGA", NULL, "DAC Voice"},
+
+ /* Ext Input -> PGA */
+ {"Ext Right PGA", NULL, "EXTR"},
+ {"Ext Left PGA", NULL, "EXTL"},
+
+ /* Ext PGA -> Ext Playback Switch */
+ {"Ext Right Enable", "Switch", "Ext Right PGA"},
+ {"Ext Left Enable", "Switch", "Ext Left PGA"},
+
+ /* HiFi PGA -> Mono Mixer */
+ {"HiFi Mono Left Mixer", NULL, "HiFi PGA"},
+ {"HiFi Mono Left Mixer", "HiFi Mono Playback Switch", "HiFi PGA"},
+ {"HiFi Mono Right Mixer", NULL, "HiFi PGA"},
+ {"HiFi Mono Right Mixer", "HiFi Mono Playback Switch", "HiFi PGA"},
+
+ /* Ext Playback Switch -> Ext Mono Mixer */
+ {"Ext Mono Right Mixer", NULL, "Ext Right Enable"},
+ {"Ext Mono Right Mixer", "Ext Mono Playback Switch", "Ext Left Enable"},
+ {"Ext Mono Left Mixer", NULL, "Ext Left Enable"},
+ {"Ext Mono Left Mixer", "Ext Mono Playback Switch", "Ext Right Enable"},
+
+ /* HiFi Mono Mixer -> Output Route */
+ {"Earpiece Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Speaker Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Speaker Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"Lineout Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Lineout Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"Headset Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"Headset Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+ {"EMU Right Playback Route", "HiFi", "HiFi Mono Right Mixer"},
+ {"EMU Left Playback Route", "HiFi", "HiFi Mono Left Mixer"},
+
+ /* Voice PGA -> Output Route */
+ {"Earpiece Playback Route", "Voice", "Voice PGA"},
+ {"Speaker Right Playback Route", "Voice", "Voice PGA"},
+ {"Speaker Left Playback Route", "Voice", "Voice PGA"},
+ {"Lineout Right Playback Route", "Voice", "Voice PGA"},
+ {"Lineout Left Playback Route", "Voice", "Voice PGA"},
+ {"Headset Right Playback Route", "Voice", "Voice PGA"},
+ {"Headset Left Playback Route", "Voice", "Voice PGA"},
+ {"EMU Right Playback Route", "Voice", "Voice PGA"},
+ {"EMU Left Playback Route", "Voice", "Voice PGA"},
+
+ /* Ext Mono Mixer -> Output Route */
+ {"Earpiece Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Speaker Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Speaker Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"Lineout Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Lineout Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"Headset Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"Headset Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+ {"EMU Right Playback Route", "Ext", "Ext Mono Right Mixer"},
+ {"EMU Left Playback Route", "Ext", "Ext Mono Left Mixer"},
+
+ /* Output Route -> Output Amplifier */
+ {"Earpiece PGA", NULL, "Earpiece Playback Route"},
+ {"Speaker Right PGA", NULL, "Speaker Right Playback Route"},
+ {"Speaker Left PGA", NULL, "Speaker Left Playback Route"},
+ {"Lineout Right PGA", NULL, "Lineout Right Playback Route"},
+ {"Lineout Left PGA", NULL, "Lineout Left Playback Route"},
+ {"Headset Right PGA", NULL, "Headset Right Playback Route"},
+ {"Headset Left PGA", NULL, "Headset Left Playback Route"},
+ {"EMU Right PGA", NULL, "EMU Right Playback Route"},
+ {"EMU Left PGA", NULL, "EMU Left Playback Route"},
+
+ /* Output Amplifier -> Output */
+ {"EP", NULL, "Earpiece PGA"},
+ {"SPKR", NULL, "Speaker Right PGA"},
+ {"SPKL", NULL, "Speaker Left PGA"},
+ {"LINER", NULL, "Lineout Right PGA"},
+ {"LINEL", NULL, "Lineout Left PGA"},
+ {"HSR", NULL, "Headset Right PGA"},
+ {"HSL", NULL, "Headset Left PGA"},
+ {"EMUR", NULL, "EMU Right PGA"},
+ {"EMUL", NULL, "EMU Left PGA"},
+
+ /* Headset Charge Pump -> Headset */
+ {"HSR", NULL, "Headset Charge Pump"},
+ {"HSL", NULL, "Headset Charge Pump"},
+
+ /* Mic -> Mic Route */
+ {"Right Capture Route", "Mic 1", "MICR"},
+ {"Right Capture Route", "Headset Mic", "HSMIC"},
+ {"Right Capture Route", "EMU Mic", "EMUMIC"},
+ {"Right Capture Route", "Ext Right", "EXTR"},
+ {"Left Capture Route", "Mic 2", "MICL"},
+ {"Left Capture Route", "Ext Left", "EXTL"},
+
+ /* Input Route -> Microphone PGA */
+ {"Microphone 1 PGA", NULL, "Right Capture Route"},
+ {"Microphone 2 PGA", NULL, "Left Capture Route"},
+
+ /* Microphone PGA -> ADC */
+ {"ADC Right", NULL, "Microphone 1 PGA"},
+ {"ADC Left", NULL, "Microphone 2 PGA"},
+
+ /* ADC -> Stream */
+ {"Highpass Filter TX", NULL, "ADC Right"},
+ {"Highpass Filter TX", NULL, "ADC Left"},
+
+ /* Mic Bias */
+ {"MICL", NULL, "MIC1L Bias"},
+ {"MICR", NULL, "MIC1R Bias"},
+};
+
+static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai,
+ int clk_id, int freq)
+{
+ u16 clkfreqreg, clkfreqshift;
+ u16 clkfreqmask, clkfreqval;
+ u16 clkidreg, clkidshift;
+ u16 mask, val;
+ int err;
+
+ switch (dai) {
+ case CPCAP_DAI_HIFI:
+ clkfreqreg = CPCAP_REG_SDAC;
+ clkfreqshift = CPCAP_BIT_ST_DAC_CLK0;
+ clkidreg = CPCAP_REG_SDACDI;
+ clkidshift = CPCAP_BIT_ST_DAC_CLK_IN_SEL;
+ break;
+ case CPCAP_DAI_VOICE:
+ clkfreqreg = CPCAP_REG_CC;
+ clkfreqshift = CPCAP_BIT_CDC_CLK0;
+ clkidreg = CPCAP_REG_CDI;
+ clkidshift = CPCAP_BIT_CLK_IN_SEL;
+ break;
+ default:
+ dev_err(cpcap->codec->dev, "invalid DAI: %d", dai);
+ return -EINVAL;
+ }
+
+ /* setup clk id */
+ if (clk_id < 0 || clk_id > 1) {
+ dev_err(cpcap->codec->dev, "invalid clk id %d", clk_id);
+ return -EINVAL;
+ }
+ err = regmap_update_bits(cpcap->regmap, clkidreg, BIT(clkidshift),
+ clk_id ? BIT(clkidshift) : 0);
+ if (err)
+ return err;
+
+ /* enable PLL for Voice DAI */
+ if (dai == CPCAP_DAI_VOICE) {
+ mask = BIT(CPCAP_BIT_CDC_PLL_SEL);
+ val = BIT(CPCAP_BIT_CDC_PLL_SEL);
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+ mask, val);
+ if (err)
+ return err;
+ }
+
+ /* setup frequency */
+ clkfreqmask = 0x7 << clkfreqshift;
+ switch (freq) {
+ case 15360000:
+ clkfreqval = 0x01 << clkfreqshift;
+ break;
+ case 16800000:
+ clkfreqval = 0x02 << clkfreqshift;
+ break;
+ case 19200000:
+ clkfreqval = 0x03 << clkfreqshift;
+ break;
+ case 26000000:
+ clkfreqval = 0x04 << clkfreqshift;
+ break;
+ case 33600000:
+ clkfreqval = 0x05 << clkfreqshift;
+ break;
+ case 38400000:
+ clkfreqval = 0x06 << clkfreqshift;
+ break;
+ default:
+ dev_err(cpcap->codec->dev, "unsupported freq %u", freq);
+ return -EINVAL;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, clkfreqreg,
+ clkfreqmask, clkfreqval);
+ if (err)
+ return err;
+
+ if (dai == CPCAP_DAI_VOICE) {
+ cpcap->codec_clk_id = clk_id;
+ cpcap->codec_freq = freq;
+ }
+
+ return 0;
+}
+
+static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai,
+ int samplerate)
+{
+ struct snd_soc_codec *codec = cpcap->codec;
+ u16 sampreg, sampmask, sampshift, sampval, sampreset;
+ int err, sampreadval;
+
+ switch (dai) {
+ case CPCAP_DAI_HIFI:
+ sampreg = CPCAP_REG_SDAC;
+ sampshift = CPCAP_BIT_ST_SR0;
+ sampreset = BIT(CPCAP_BIT_DF_RESET_ST_DAC) |
+ BIT(CPCAP_BIT_ST_CLOCK_TREE_RESET);
+ break;
+ case CPCAP_DAI_VOICE:
+ sampreg = CPCAP_REG_CC;
+ sampshift = CPCAP_BIT_CDC_SR0;
+ sampreset = BIT(CPCAP_BIT_DF_RESET) |
+ BIT(CPCAP_BIT_CDC_CLOCK_TREE_RESET);
+ break;
+ default:
+ dev_err(codec->dev, "invalid DAI: %d", dai);
+ return -EINVAL;
+ }
+
+ sampmask = 0xF << sampshift | sampreset;
+ switch (samplerate) {
+ case 48000:
+ sampval = 0x8 << sampshift;
+ break;
+ case 44100:
+ sampval = 0x7 << sampshift;
+ break;
+ case 32000:
+ sampval = 0x6 << sampshift;
+ break;
+ case 24000:
+ sampval = 0x5 << sampshift;
+ break;
+ case 22050:
+ sampval = 0x4 << sampshift;
+ break;
+ case 16000:
+ sampval = 0x3 << sampshift;
+ break;
+ case 12000:
+ sampval = 0x2 << sampshift;
+ break;
+ case 11025:
+ sampval = 0x1 << sampshift;
+ break;
+ case 8000:
+ sampval = 0x0 << sampshift;
+ break;
+ default:
+ dev_err(codec->dev, "unsupported samplerate %d", samplerate);
+ return -EINVAL;
+ }
+ err = regmap_update_bits(cpcap->regmap, sampreg,
+ sampmask, sampval | sampreset);
+ if (err)
+ return err;
+
+ /* Wait for clock tree reset to complete */
+ mdelay(CLOCK_TREE_RESET_TIME);
+
+ err = regmap_read(cpcap->regmap, sampreg, &sampreadval);
+ if (err)
+ return err;
+
+ if (sampreadval & sampreset) {
+ dev_err(codec->dev, "reset self-clear failed: %04x",
+ sampreadval);
+ return -EIO;
+ }
+
+ return 0;
+}
+
+static int cpcap_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int rate = params_rate(params);
+
+ dev_dbg(codec->dev, "HiFi setup HW params: rate=%d", rate);
+ return cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, rate);
+}
+
+static int cpcap_hifi_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ dev_dbg(dev, "HiFi setup sysclk: clk_id=%u, freq=%u", clk_id, freq);
+ return cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, clk_id, freq);
+}
+
+static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+ static const u16 reg = CPCAP_REG_SDACDI;
+ static const u16 mask =
+ BIT(CPCAP_BIT_SMB_ST_DAC) |
+ BIT(CPCAP_BIT_ST_CLK_INV) |
+ BIT(CPCAP_BIT_ST_FS_INV) |
+ BIT(CPCAP_BIT_ST_DIG_AUD_FS0) |
+ BIT(CPCAP_BIT_ST_DIG_AUD_FS1) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT0) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT1) |
+ BIT(CPCAP_BIT_ST_L_TIMESLOT2) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT0) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT1) |
+ BIT(CPCAP_BIT_ST_R_TIMESLOT2);
+ u16 val = 0x0000;
+
+ dev_dbg(dev, "HiFi setup dai format (%08x)", fmt);
+
+ /*
+ * "HiFi Playback" should always be configured as
+ * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master
+ * SND_SOC_DAIFMT_I2S - I2S mode
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val &= ~BIT(CPCAP_BIT_SMB_ST_DAC);
+ break;
+ default:
+ dev_err(dev, "HiFi dai fmt failed: CPCAP should be master");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= BIT(CPCAP_BIT_ST_FS_INV);
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ val &= ~BIT(CPCAP_BIT_ST_FS_INV);
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val |= BIT(CPCAP_BIT_ST_FS_INV);
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ val &= ~BIT(CPCAP_BIT_ST_FS_INV);
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ break;
+ default:
+ dev_err(dev, "HiFi dai fmt failed: unsupported clock invert mode");
+ return -EINVAL;
+ }
+
+ if (val & BIT(CPCAP_BIT_ST_CLK_INV))
+ val &= ~BIT(CPCAP_BIT_ST_CLK_INV);
+ else
+ val |= BIT(CPCAP_BIT_ST_CLK_INV);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS0);
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS1);
+ break;
+ default:
+ /* 01 - 4 slots network mode */
+ val |= BIT(CPCAP_BIT_ST_DIG_AUD_FS0);
+ val &= ~BIT(CPCAP_BIT_ST_DIG_AUD_FS1);
+ /* L on slot 1 */
+ val |= BIT(CPCAP_BIT_ST_L_TIMESLOT0);
+ break;
+ }
+
+ dev_dbg(dev, "HiFi dai format: val=%04x", val);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+}
+
+static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg = CPCAP_REG_RXSDOA;
+ static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW);
+ u16 val;
+
+ if (mute)
+ val = 0;
+ else
+ val = BIT(CPCAP_BIT_ST_DAC_SW);
+
+ dev_dbg(codec->dev, "HiFi mute: %d", mute);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+}
+
+static const struct snd_soc_dai_ops cpcap_dai_hifi_ops = {
+ .hw_params = cpcap_hifi_hw_params,
+ .set_sysclk = cpcap_hifi_set_dai_sysclk,
+ .set_fmt = cpcap_hifi_set_dai_fmt,
+ .digital_mute = cpcap_hifi_set_mute,
+};
+
+static int cpcap_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct device *dev = codec->dev;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg_cdi = CPCAP_REG_CDI;
+ int rate = params_rate(params);
+ int channels = params_channels(params);
+ int direction = substream->stream;
+ u16 val, mask;
+ int err;
+
+ dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d",
+ rate, direction, channels);
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate);
+ if (err)
+ return err;
+
+ if (direction == SNDRV_PCM_STREAM_CAPTURE) {
+ mask = 0x0000;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1;
+ mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT0;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT1;
+ mask |= CPCAP_BIT_MIC2_TIMESLOT2;
+ val = 0x0000;
+ if (channels >= 2)
+ val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0);
+ err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val);
+ if (err)
+ return err;
+ }
+
+ return 0;
+}
+
+static int cpcap_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
+ unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "Voice setup sysclk: clk_id=%u, freq=%u",
+ clk_id, freq);
+ return cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, clk_id, freq);
+}
+
+static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 mask = BIT(CPCAP_BIT_SMB_CDC) |
+ BIT(CPCAP_BIT_CLK_INV) |
+ BIT(CPCAP_BIT_FS_INV) |
+ BIT(CPCAP_BIT_CDC_DIG_AUD_FS0) |
+ BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ u16 val = 0x0000;
+ int err;
+
+ dev_dbg(codec->dev, "Voice setup dai format (%08x)", fmt);
+
+ /*
+ * "Voice Playback" and "Voice Capture" should always be
+ * configured as SND_SOC_DAIFMT_CBM_CFM - codec clk & frm
+ * master
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ val &= ~BIT(CPCAP_BIT_SMB_CDC);
+ break;
+ default:
+ dev_err(codec->dev, "Voice dai fmt failed: CPCAP should be the master");
+ val &= ~BIT(CPCAP_BIT_SMB_CDC);
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_IF:
+ val |= BIT(CPCAP_BIT_CLK_INV);
+ val |= BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ val |= BIT(CPCAP_BIT_CLK_INV);
+ val &= ~BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ val |= BIT(CPCAP_BIT_FS_INV);
+ break;
+ case SND_SOC_DAIFMT_NB_NF:
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ val &= ~BIT(CPCAP_BIT_FS_INV);
+ break;
+ default:
+ dev_err(codec->dev, "Voice dai fmt failed: unsupported clock invert mode");
+ break;
+ }
+
+ if (val & BIT(CPCAP_BIT_CLK_INV))
+ val &= ~BIT(CPCAP_BIT_CLK_INV);
+ else
+ val |= BIT(CPCAP_BIT_CLK_INV);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ /* 11 - true I2S mode */
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS0);
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ break;
+ default:
+ /* 4 timeslots network mode */
+ val |= BIT(CPCAP_BIT_CDC_DIG_AUD_FS0);
+ val &= ~BIT(CPCAP_BIT_CDC_DIG_AUD_FS1);
+ break;
+ }
+
+ dev_dbg(codec->dev, "Voice dai format: val=%04x", val);
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, mask, val);
+ if (err)
+ return err;
+
+ cpcap->codec_format = val;
+ return 0;
+}
+
+static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ static const u16 reg = CPCAP_REG_RXCOA;
+ static const u16 mask = BIT(CPCAP_BIT_CDC_SW);
+ u16 val;
+
+ if (mute)
+ val = 0;
+ else
+ val = BIT(CPCAP_BIT_CDC_SW);
+
+ dev_dbg(codec->dev, "Voice mute: %d", mute);
+ return regmap_update_bits(cpcap->regmap, reg, mask, val);
+};
+
+static const struct snd_soc_dai_ops cpcap_dai_voice_ops = {
+ .hw_params = cpcap_voice_hw_params,
+ .set_sysclk = cpcap_voice_set_dai_sysclk,
+ .set_fmt = cpcap_voice_set_dai_fmt,
+ .digital_mute = cpcap_voice_set_mute,
+};
+
+static struct snd_soc_dai_driver cpcap_dai[] = {
+{
+ .id = 0,
+ .name = "cpcap-hifi",
+ .playback = {
+ .stream_name = "HiFi Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE,
+ },
+ .ops = &cpcap_dai_hifi_ops,
+},
+{
+ .id = 1,
+ .name = "cpcap-voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .stream_name = "Voice Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &cpcap_dai_voice_ops,
+},
+};
+
+static int cpcap_dai_mux(struct cpcap_audio *cpcap, bool swap_dai_configuration)
+{
+ u16 hifi_val, voice_val;
+ u16 hifi_mask = BIT(CPCAP_BIT_DIG_AUD_IN_ST_DAC);
+ u16 voice_mask = BIT(CPCAP_BIT_DIG_AUD_IN);
+ int err;
+
+
+
+ if (!swap_dai_configuration) {
+ /* Codec on DAI0, HiFi on DAI1 */
+ voice_val = 0;
+ hifi_val = hifi_mask;
+ } else {
+ /* Codec on DAI1, HiFi on DAI0 */
+ voice_val = voice_mask;
+ hifi_val = 0;
+ }
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI,
+ voice_mask, voice_val);
+ if (err)
+ return err;
+
+ err = regmap_update_bits(cpcap->regmap, CPCAP_REG_SDACDI,
+ hifi_mask, hifi_val);
+ if (err)
+ return err;
+
+ return 0;
+}
+
+static int cpcap_audio_reset(struct snd_soc_codec *codec,
+ bool swap_dai_configuration)
+{
+ struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec);
+ int i, err = 0;
+
+ dev_dbg(codec->dev, "init audio codec");
+
+ for (i = 0; i < ARRAY_SIZE(cpcap_default_regs); i++) {
+ err = regmap_update_bits(cpcap->regmap,
+ cpcap_default_regs[i].reg,
+ cpcap_default_regs[i].mask,
+ cpcap_default_regs[i].val);
+ if (err)
+ return err;
+ }
+
+ /* setup default settings */
+ err = cpcap_dai_mux(cpcap, swap_dai_configuration);
+ if (err)
+ return err;
+
+ err = cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, 0, 26000000);
+ if (err)
+ return err;
+ err = cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, 0, 26000000);
+ if (err)
+ return err;
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, 48000);
+ if (err)
+ return err;
+
+ err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, 48000);
+ if (err)
+ return err;
+
+ return 0;
+}
+
+static int cpcap_soc_probe(struct snd_soc_codec *codec)
+{
+ struct cpcap_audio *cpcap;
+ int err;
+
+ cpcap = devm_kzalloc(codec->dev, sizeof(*cpcap), GFP_KERNEL);
+ if (!cpcap)
+ return -ENOMEM;
+ snd_soc_codec_set_drvdata(codec, cpcap);
+ cpcap->codec = codec;
+
+ cpcap->regmap = dev_get_regmap(codec->dev->parent, NULL);
+ if (!cpcap->regmap)
+ return -ENODEV;
+ snd_soc_codec_init_regmap(codec, cpcap->regmap);
+
+ err = cpcap_get_vendor(codec->dev, cpcap->regmap, &cpcap->vendor);
+ if (err)
+ return err;
+
+ return cpcap_audio_reset(codec, false);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_cpcap = {
+ .probe = cpcap_soc_probe,
+
+ .component_driver = {
+ .controls = cpcap_snd_controls,
+ .num_controls = ARRAY_SIZE(cpcap_snd_controls),
+ .dapm_widgets = cpcap_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+ },
+};
+
+static int cpcap_codec_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cpcap,
+ cpcap_dai, ARRAY_SIZE(cpcap_dai));
+}
+
+static int cpcap_codec_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+#ifdef CONFIG_OF
+static const struct of_device_id cpcap_audio_of_match[] = {
+ { .compatible = "motorola,cpcap-audio-codec", },
+ {},
+};
+MODULE_DEVICE_TABLE(of, cpcap_audio_of_match);
+#endif
+
+static struct platform_driver cpcap_codec_driver = {
+ .probe = cpcap_codec_probe,
+ .remove = cpcap_codec_remove,
+ .driver = {
+ .name = "cpcap-codec",
+ .of_match_table = of_match_ptr(cpcap_audio_of_match),
+ },
+};
+module_platform_driver(cpcap_codec_driver);
+
+MODULE_ALIAS("platform:cpcap-codec");
+MODULE_DESCRIPTION("ASoC CPCAP codec driver");
+MODULE_AUTHOR("Sebastian Reichel");
+MODULE_LICENSE("GPL");
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* [PATCHv3 4/6] ARM: dts: motorola-cpcap-mapphone: add audio-codec
2017-07-25 15:10 ` Sebastian Reichel
` (3 preceding siblings ...)
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
-1 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Add node for audio-codec to its DT file.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 6 ++++++
1 file changed, 6 insertions(+)
diff --git a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
index 1eb5da1dc8f0..86033aee8a29 100644
--- a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
+++ b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
@@ -68,6 +68,12 @@
};
};
+ cpcap_audio: audio-codec {
+ #sound-dai-cells = <1>;
+ compatible = "motorola,cpcap-audio-codec";
+ vdd-supply = <&vaudio>;
+ };
+
cpcap_rtc: rtc {
compatible = "motorola,cpcap-rtc";
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* [PATCHv3 5/6] ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio
2017-07-25 15:10 ` Sebastian Reichel
` (4 preceding siblings ...)
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
-1 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Set default mode for vaudio, which may be left in standby mode
if the system is booted via kexec from Android.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi | 1 +
1 file changed, 1 insertion(+)
diff --git a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
index 86033aee8a29..7c6d61fb5cf2 100644
--- a/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
+++ b/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
@@ -261,5 +261,6 @@
regulator-min-microvolt = <2775000>;
regulator-max-microvolt = <2775000>;
regulator-enable-ramp-delay = <1000>;
+ regulator-initial-mode = <0x00>; /* NORMAL */
};
};
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* [PATCHv3 6/6] ARM: dts: omap4-droid4: add soundcard
2017-07-25 15:10 ` Sebastian Reichel
` (5 preceding siblings ...)
(?)
@ 2017-07-25 15:10 ` Sebastian Reichel
-1 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-25 15:10 UTC (permalink / raw)
To: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring, Tony Lindgren
Cc: Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel, Sebastian Reichel
Add sound support to Motorola Droid 4 using simple-soundcard
and CPCAP's audio codec. This does not yet correctly represent
the whole audio routing, since McBSP3 is also connected to
Bluetooth and MDM6600 modem (and probably also 4G modem).
These extra DAI links are not yet supported and have not been
tested.
Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
---
arch/arm/boot/dts/omap4-droid4-xt894.dts | 84 ++++++++++++++++++++++++++++++++
1 file changed, 84 insertions(+)
diff --git a/arch/arm/boot/dts/omap4-droid4-xt894.dts b/arch/arm/boot/dts/omap4-droid4-xt894.dts
index 10ca1c174995..a182f9fb1c7a 100644
--- a/arch/arm/boot/dts/omap4-droid4-xt894.dts
+++ b/arch/arm/boot/dts/omap4-droid4-xt894.dts
@@ -116,6 +116,26 @@
};
};
+
+ soundcard {
+ compatible = "audio-graph-card";
+ label = "Droid 4 Audio";
+
+ simple-graph-card,widgets =
+ "Speaker", "Earpiece",
+ "Speaker", "Loudspeaker",
+ "Headphone", "Headphone Jack",
+ "Microphone", "Internal Mic";
+
+ simple-graph-card,routing =
+ "Earpiece", "EP",
+ "Loudspeaker", "SPKR",
+ "Headphone Jack", "HSL",
+ "Headphone Jack", "HSR",
+ "MICR", "Internal Mic";
+
+ dais = <&mcbsp2_port>, <&mcbsp3_port>;
+ };
};
&dss {
@@ -479,6 +499,24 @@
OMAP4_IOPAD(0x112, PIN_OUTPUT_PULLUP | MUX_MODE5) /* uart4_rts */
>;
};
+
+ mcbsp2_pins: pinmux_mcbsp2_pins {
+ pinctrl-single,pins = <
+ OMAP4_IOPAD(0x0f6, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_clkx */
+ OMAP4_IOPAD(0x0f8, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_dr */
+ OMAP4_IOPAD(0x0fa, PIN_OUTPUT | MUX_MODE0) /* abe_mcbsp2_dx */
+ OMAP4_IOPAD(0x0fc, PIN_INPUT | MUX_MODE0) /* abe_mcbsp2_fsx */
+ >;
+ };
+
+ mcbsp3_pins: pinmux_mcbsp3_pins {
+ pinctrl-single,pins = <
+ OMAP4_IOPAD(0x106, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_dr */
+ OMAP4_IOPAD(0x108, PIN_OUTPUT | MUX_MODE1) /* abe_mcbsp3_dx */
+ OMAP4_IOPAD(0x10a, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_clkx */
+ OMAP4_IOPAD(0x10c, PIN_INPUT | MUX_MODE1) /* abe_mcbsp3_fsx */
+ >;
+ };
};
&omap4_pmx_wkup {
@@ -552,3 +590,49 @@
"0", "0", "1";
};
};
+
+&mcbsp2 {
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&mcbsp2_pins>;
+ status = "okay";
+
+ mcbsp2_port: port {
+ cpu_dai2: endpoint {
+ dai-format = "i2s";
+ remote-endpoint = <&cpcap_audio_codec0>;
+ frame-master = <&cpcap_audio_codec0>;
+ bitclock-master = <&cpcap_audio_codec0>;
+ };
+ };
+};
+
+&mcbsp3 {
+ #sound-dai-cells = <0>;
+ pinctrl-names = "default";
+ pinctrl-0 = <&mcbsp3_pins>;
+ status = "okay";
+
+ mcbsp3_port: port {
+ cpu_dai3: endpoint {
+ dai-format = "dsp_a";
+ frame-master = <&cpcap_audio_codec1>;
+ bitclock-master = <&cpcap_audio_codec1>;
+ remote-endpoint = <&cpcap_audio_codec1>;
+ };
+ };
+};
+
+&cpcap_audio {
+ port@0 {
+ cpcap_audio_codec0: endpoint {
+ remote-endpoint = <&cpu_dai2>;
+ };
+ };
+
+ port@1 {
+ cpcap_audio_codec1: endpoint {
+ remote-endpoint = <&cpu_dai3>;
+ };
+ };
+};
--
2.13.2
^ permalink raw reply related [flat|nested] 22+ messages in thread
* Re: [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
2017-07-25 15:10 ` [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update Sebastian Reichel
@ 2017-07-25 15:42 ` Takashi Iwai
0 siblings, 0 replies; 22+ messages in thread
From: Takashi Iwai @ 2017-07-25 15:42 UTC (permalink / raw)
To: Sebastian Reichel
Cc: Tony Lindgren, Liam Girdwood, Mark Brown, Rob Herring,
Sebastian Reichel, alsa-devel, Jaroslav Kysela, devicetree,
linux-kernel, linux-omap
On Tue, 25 Jul 2017 17:10:25 +0200,
Sebastian Reichel wrote:
>
> From: Tony Lindgren <tony@atomide.com>
>
> Some codecs may need to use this from loadable modules. Without
> this patch compilation for this kind of codec will fail with the
> following error:
>
> ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
> undefined!
>
> Signed-off-by: Tony Lindgren <tony@atomide.com>
> Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
If you do export the function, please make its comments prettier, so
that it appears in kernel documentation properly as an API function.
Also, in general, we use snd_ prefix for the exported stuff.
You'd need to rename it accordingly, too.
Last but not least, this is specific to ASoC, so use "ASoC" prefix to
the subject line.
thanks,
Takashi
> ---
> sound/soc/soc-pcm.c | 2 ++
> 1 file changed, 2 insertions(+)
>
> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
> index dcc5ece08668..4de7d86ea6fb 100644
> --- a/sound/soc/soc-pcm.c
> +++ b/sound/soc/soc-pcm.c
> @@ -2539,6 +2539,8 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
> mutex_unlock(&card->mutex);
> return 0;
> }
> +EXPORT_SYMBOL_GPL(soc_dpcm_runtime_update);
> +
> int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
> {
> struct snd_soc_dpcm *dpcm;
> --
> 2.13.2
>
>
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
@ 2017-07-25 15:42 ` Takashi Iwai
0 siblings, 0 replies; 22+ messages in thread
From: Takashi Iwai @ 2017-07-25 15:42 UTC (permalink / raw)
To: Sebastian Reichel
Cc: devicetree, alsa-devel, linux-omap, Tony Lindgren, linux-kernel,
Liam Girdwood, Rob Herring, Sebastian Reichel, Mark Brown
On Tue, 25 Jul 2017 17:10:25 +0200,
Sebastian Reichel wrote:
>
> From: Tony Lindgren <tony@atomide.com>
>
> Some codecs may need to use this from loadable modules. Without
> this patch compilation for this kind of codec will fail with the
> following error:
>
> ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
> undefined!
>
> Signed-off-by: Tony Lindgren <tony@atomide.com>
> Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
If you do export the function, please make its comments prettier, so
that it appears in kernel documentation properly as an API function.
Also, in general, we use snd_ prefix for the exported stuff.
You'd need to rename it accordingly, too.
Last but not least, this is specific to ASoC, so use "ASoC" prefix to
the subject line.
thanks,
Takashi
> ---
> sound/soc/soc-pcm.c | 2 ++
> 1 file changed, 2 insertions(+)
>
> diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
> index dcc5ece08668..4de7d86ea6fb 100644
> --- a/sound/soc/soc-pcm.c
> +++ b/sound/soc/soc-pcm.c
> @@ -2539,6 +2539,8 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
> mutex_unlock(&card->mutex);
> return 0;
> }
> +EXPORT_SYMBOL_GPL(soc_dpcm_runtime_update);
> +
> int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute)
> {
> struct snd_soc_dpcm *dpcm;
> --
> 2.13.2
>
>
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
2017-07-25 15:42 ` Takashi Iwai
(?)
@ 2017-07-26 11:38 ` Mark Brown
-1 siblings, 0 replies; 22+ messages in thread
From: Mark Brown @ 2017-07-26 11:38 UTC (permalink / raw)
To: Takashi Iwai
Cc: Sebastian Reichel, Tony Lindgren, Liam Girdwood, Rob Herring,
Sebastian Reichel, alsa-devel, Jaroslav Kysela, devicetree,
linux-kernel, linux-omap
[-- Attachment #1: Type: text/plain, Size: 219 bytes --]
On Tue, Jul 25, 2017 at 05:42:59PM +0200, Takashi Iwai wrote:
> Last but not least, this is specific to ASoC, so use "ASoC" prefix to
> the subject line.
Right, that's a good way to get the patch skipped in review :(
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
@ 2017-07-26 11:48 ` Mark Brown
0 siblings, 0 replies; 22+ messages in thread
From: Mark Brown @ 2017-07-26 11:48 UTC (permalink / raw)
To: Sebastian Reichel
Cc: Sebastian Reichel, Liam Girdwood, Rob Herring, Tony Lindgren,
Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel
[-- Attachment #1: Type: text/plain, Size: 1055 bytes --]
On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> Motorola CPCAP is a PMIC with audio functionality, that can be
> found on Motorola Droid 4 and probably a few other phones from
> Motorola's Droid series.
Please submit patches using subject lines reflecting the style for the
subsystem. This makes it easier for people to identify relevant
patches. Look at what existing commits in the area you're changing are
doing and make sure your subject lines visually resemble what they're
doing.
> +&cpcap {
> + audio-codec {
> + compatible = "motorola,cpcap-audio-codec";
> + vdd-supply = <&vaudio>;
> + };
> +};
I'd expect supplies (especially generically named supplies like this) to
be looked up at the chip level - aside from my general concerns with MFD
subnodes like this in the case of supplies it's especially problematic
as it makes it harder to do the generic chip level hookup in the DT and
it precludes other parts of the chip using the same supply (which seems
especially likely with a generically named supply like this).
[-- Attachment #2: signature.asc --]
[-- Type: application/pgp-signature, Size: 488 bytes --]
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
@ 2017-07-26 11:48 ` Mark Brown
0 siblings, 0 replies; 22+ messages in thread
From: Mark Brown @ 2017-07-26 11:48 UTC (permalink / raw)
To: Sebastian Reichel
Cc: Sebastian Reichel, Liam Girdwood, Rob Herring, Tony Lindgren,
Jaroslav Kysela, Takashi Iwai, alsa-devel-K7yf7f+aM1XWsZ/bQMPhNw,
linux-omap-u79uwXL29TY76Z2rM5mHXA,
devicetree-u79uwXL29TY76Z2rM5mHXA,
linux-kernel-u79uwXL29TY76Z2rM5mHXA
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On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> Motorola CPCAP is a PMIC with audio functionality, that can be
> found on Motorola Droid 4 and probably a few other phones from
> Motorola's Droid series.
Please submit patches using subject lines reflecting the style for the
subsystem. This makes it easier for people to identify relevant
patches. Look at what existing commits in the area you're changing are
doing and make sure your subject lines visually resemble what they're
doing.
> +&cpcap {
> + audio-codec {
> + compatible = "motorola,cpcap-audio-codec";
> + vdd-supply = <&vaudio>;
> + };
> +};
I'd expect supplies (especially generically named supplies like this) to
be looked up at the chip level - aside from my general concerns with MFD
subnodes like this in the case of supplies it's especially problematic
as it makes it harder to do the generic chip level hookup in the DT and
it precludes other parts of the chip using the same supply (which seems
especially likely with a generically named supply like this).
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
@ 2017-07-27 9:01 ` Sebastian Reichel
0 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-27 9:01 UTC (permalink / raw)
To: Mark Brown
Cc: Liam Girdwood, Rob Herring, Tony Lindgren, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-omap, devicetree, linux-kernel
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Hi,
On Wed, Jul 26, 2017 at 12:48:28PM +0100, Mark Brown wrote:
> On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> > Motorola CPCAP is a PMIC with audio functionality, that can be
> > found on Motorola Droid 4 and probably a few other phones from
> > Motorola's Droid series.
>
> Please submit patches using subject lines reflecting the style for the
> subsystem. This makes it easier for people to identify relevant
> patches. Look at what existing commits in the area you're changing are
> doing and make sure your subject lines visually resemble what they're
> doing.
Right, I did not notice, that ASoC does not follow general
"dt-bindings: <subsys>:" DT bindings subject style. How
do Rob and Mark find them?
> > +&cpcap {
> > + audio-codec {
> > + compatible = "motorola,cpcap-audio-codec";
> > + vdd-supply = <&vaudio>;
> > + };
> > +};
>
> I'd expect supplies (especially generically named supplies like this) to
> be looked up at the chip level - aside from my general concerns with MFD
> subnodes like this in the case of supplies it's especially problematic
> as it makes it harder to do the generic chip level hookup in the DT and
> it precludes other parts of the chip using the same supply (which seems
> especially likely with a generically named supply like this).
I don't follow you here. Why can't other parts of the chip use the
same supply? Regarding the other point: Handling the audio-codec
differently than all other sub-modules of cpcap seems much more
problematic to me and the codec is basically the last one
missing:
https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
-- Sebastian
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
@ 2017-07-27 9:01 ` Sebastian Reichel
0 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-27 9:01 UTC (permalink / raw)
To: Mark Brown
Cc: Liam Girdwood, Rob Herring, Tony Lindgren, Jaroslav Kysela,
Takashi Iwai, alsa-devel-K7yf7f+aM1XWsZ/bQMPhNw,
linux-omap-u79uwXL29TY76Z2rM5mHXA,
devicetree-u79uwXL29TY76Z2rM5mHXA,
linux-kernel-u79uwXL29TY76Z2rM5mHXA
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Hi,
On Wed, Jul 26, 2017 at 12:48:28PM +0100, Mark Brown wrote:
> On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> > Motorola CPCAP is a PMIC with audio functionality, that can be
> > found on Motorola Droid 4 and probably a few other phones from
> > Motorola's Droid series.
>
> Please submit patches using subject lines reflecting the style for the
> subsystem. This makes it easier for people to identify relevant
> patches. Look at what existing commits in the area you're changing are
> doing and make sure your subject lines visually resemble what they're
> doing.
Right, I did not notice, that ASoC does not follow general
"dt-bindings: <subsys>:" DT bindings subject style. How
do Rob and Mark find them?
> > +&cpcap {
> > + audio-codec {
> > + compatible = "motorola,cpcap-audio-codec";
> > + vdd-supply = <&vaudio>;
> > + };
> > +};
>
> I'd expect supplies (especially generically named supplies like this) to
> be looked up at the chip level - aside from my general concerns with MFD
> subnodes like this in the case of supplies it's especially problematic
> as it makes it harder to do the generic chip level hookup in the DT and
> it precludes other parts of the chip using the same supply (which seems
> especially likely with a generically named supply like this).
I don't follow you here. Why can't other parts of the chip use the
same supply? Regarding the other point: Handling the audio-codec
differently than all other sub-modules of cpcap seems much more
problematic to me and the codec is basically the last one
missing:
https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
-- Sebastian
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
2017-07-25 15:42 ` Takashi Iwai
@ 2017-07-27 10:01 ` Sebastian Reichel
-1 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-27 10:01 UTC (permalink / raw)
To: Takashi Iwai
Cc: Tony Lindgren, Liam Girdwood, Mark Brown, Rob Herring,
alsa-devel, Jaroslav Kysela, devicetree, linux-kernel,
linux-omap
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Hi,
On Tue, Jul 25, 2017 at 05:42:59PM +0200, Takashi Iwai wrote:
> On Tue, 25 Jul 2017 17:10:25 +0200,
> Sebastian Reichel wrote:
> > Some codecs may need to use this from loadable modules. Without
> > this patch compilation for this kind of codec will fail with the
> > following error:
> >
> > ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
> > undefined!
> >
> > Signed-off-by: Tony Lindgren <tony@atomide.com>
> > Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
>
> If you do export the function, please make its comments prettier, so
> that it appears in kernel documentation properly as an API function.
>
> Also, in general, we use snd_ prefix for the exported stuff.
> You'd need to rename it accordingly, too.
>
> Last but not least, this is specific to ASoC, so use "ASoC" prefix to
> the subject line.
After fixing this locally I noticed, that I no longer need to call
soc_dpcm_runtime_update at all and it can remain private. So this
patch will be dropped in v4.
-- Sebastian
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update
@ 2017-07-27 10:01 ` Sebastian Reichel
0 siblings, 0 replies; 22+ messages in thread
From: Sebastian Reichel @ 2017-07-27 10:01 UTC (permalink / raw)
To: Takashi Iwai
Cc: devicetree, alsa-devel, Tony Lindgren, linux-kernel,
Liam Girdwood, Rob Herring, Mark Brown, linux-omap
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Hi,
On Tue, Jul 25, 2017 at 05:42:59PM +0200, Takashi Iwai wrote:
> On Tue, 25 Jul 2017 17:10:25 +0200,
> Sebastian Reichel wrote:
> > Some codecs may need to use this from loadable modules. Without
> > this patch compilation for this kind of codec will fail with the
> > following error:
> >
> > ERROR: "soc_dpcm_runtime_update" [sound/soc/codecs/snd-soc-cpcap.ko]
> > undefined!
> >
> > Signed-off-by: Tony Lindgren <tony@atomide.com>
> > Signed-off-by: Sebastian Reichel <sebastian.reichel@collabora.co.uk>
>
> If you do export the function, please make its comments prettier, so
> that it appears in kernel documentation properly as an API function.
>
> Also, in general, we use snd_ prefix for the exported stuff.
> You'd need to rename it accordingly, too.
>
> Last but not least, this is specific to ASoC, so use "ASoC" prefix to
> the subject line.
After fixing this locally I noticed, that I no longer need to call
soc_dpcm_runtime_update at all and it can remain private. So this
patch will be dropped in v4.
-- Sebastian
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 0/6] Motorola Droid 4 Audio Support
2017-07-25 15:10 ` Sebastian Reichel
` (6 preceding siblings ...)
(?)
@ 2017-07-27 12:01 ` Tony Lindgren
-1 siblings, 0 replies; 22+ messages in thread
From: Tony Lindgren @ 2017-07-27 12:01 UTC (permalink / raw)
To: Sebastian Reichel
Cc: Sebastian Reichel, Liam Girdwood, Mark Brown, Rob Herring,
Jaroslav Kysela, Takashi Iwai, alsa-devel, linux-omap,
devicetree, linux-kernel
* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170725 08:11]:
> Hi,
>
> This adds audio support to Motorola Droid 4.
>
> Tested:
> - Playing via both DACs using Speaker, Earpiece, Headphone
> - Recording using internal Mic
> - Volume Controls
>
> Known not to work:
> - 3.5mm detection support (requires closed source firmware,
> needs further investigation)
> - Modem / Bluetooth Audio (actually untested, but probably
> needs some quirks)
>
> Changes since PATCHv2:
> * Fix a whitespace issue
> * Fix const notes Takashi provided
> * Fix a DAPM route issue I accidently introduced in v2
Audio works for me after applying the cpcap-regulator fixes
and with the dts kobject ref fix I posted earlier today:
Tested-by: Tony Lindgren <tony@atomide.com>
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
2017-07-27 9:01 ` Sebastian Reichel
@ 2017-08-10 16:53 ` Tony Lindgren
-1 siblings, 0 replies; 22+ messages in thread
From: Tony Lindgren @ 2017-08-10 16:53 UTC (permalink / raw)
To: Sebastian Reichel
Cc: Mark Brown, Liam Girdwood, Rob Herring, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-omap, devicetree, linux-kernel
* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170727 02:02]:
> Hi,
>
> On Wed, Jul 26, 2017 at 12:48:28PM +0100, Mark Brown wrote:
> > On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> > > Motorola CPCAP is a PMIC with audio functionality, that can be
> > > found on Motorola Droid 4 and probably a few other phones from
> > > Motorola's Droid series.
> >
> > Please submit patches using subject lines reflecting the style for the
> > subsystem. This makes it easier for people to identify relevant
> > patches. Look at what existing commits in the area you're changing are
> > doing and make sure your subject lines visually resemble what they're
> > doing.
>
> Right, I did not notice, that ASoC does not follow general
> "dt-bindings: <subsys>:" DT bindings subject style. How
> do Rob and Mark find them?
>
> > > +&cpcap {
> > > + audio-codec {
> > > + compatible = "motorola,cpcap-audio-codec";
> > > + vdd-supply = <&vaudio>;
> > > + };
> > > +};
> >
> > I'd expect supplies (especially generically named supplies like this) to
> > be looked up at the chip level - aside from my general concerns with MFD
> > subnodes like this in the case of supplies it's especially problematic
> > as it makes it harder to do the generic chip level hookup in the DT and
> > it precludes other parts of the chip using the same supply (which seems
> > especially likely with a generically named supply like this).
>
> I don't follow you here. Why can't other parts of the chip use the
> same supply? Regarding the other point: Handling the audio-codec
> differently than all other sub-modules of cpcap seems much more
> problematic to me and the codec is basically the last one
> missing:
>
> https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
Mark, any comments on the above? I'm just wondering if the
related dts changes are safe for me to pick.
Regards,
Tony
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola, cpcap-audio-codec
@ 2017-08-10 16:53 ` Tony Lindgren
0 siblings, 0 replies; 22+ messages in thread
From: Tony Lindgren @ 2017-08-10 16:53 UTC (permalink / raw)
To: Sebastian Reichel
Cc: devicetree, alsa-devel, linux-omap, linux-kernel, Takashi Iwai,
Rob Herring, Liam Girdwood, Mark Brown
* Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170727 02:02]:
> Hi,
>
> On Wed, Jul 26, 2017 at 12:48:28PM +0100, Mark Brown wrote:
> > On Tue, Jul 25, 2017 at 05:10:26PM +0200, Sebastian Reichel wrote:
> > > Motorola CPCAP is a PMIC with audio functionality, that can be
> > > found on Motorola Droid 4 and probably a few other phones from
> > > Motorola's Droid series.
> >
> > Please submit patches using subject lines reflecting the style for the
> > subsystem. This makes it easier for people to identify relevant
> > patches. Look at what existing commits in the area you're changing are
> > doing and make sure your subject lines visually resemble what they're
> > doing.
>
> Right, I did not notice, that ASoC does not follow general
> "dt-bindings: <subsys>:" DT bindings subject style. How
> do Rob and Mark find them?
>
> > > +&cpcap {
> > > + audio-codec {
> > > + compatible = "motorola,cpcap-audio-codec";
> > > + vdd-supply = <&vaudio>;
> > > + };
> > > +};
> >
> > I'd expect supplies (especially generically named supplies like this) to
> > be looked up at the chip level - aside from my general concerns with MFD
> > subnodes like this in the case of supplies it's especially problematic
> > as it makes it harder to do the generic chip level hookup in the DT and
> > it precludes other parts of the chip using the same supply (which seems
> > especially likely with a generically named supply like this).
>
> I don't follow you here. Why can't other parts of the chip use the
> same supply? Regarding the other point: Handling the audio-codec
> differently than all other sub-modules of cpcap seems much more
> problematic to me and the codec is basically the last one
> missing:
>
> https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arch/arm/boot/dts/motorola-cpcap-mapphone.dtsi
Mark, any comments on the above? I'm just wondering if the
related dts changes are safe for me to pick.
Regards,
Tony
^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec
2017-08-10 16:53 ` [PATCHv3 2/6] dt-bindings: sound: add motorola, cpcap-audio-codec Tony Lindgren
@ 2017-08-18 11:47 ` Mark Brown
-1 siblings, 0 replies; 22+ messages in thread
From: Mark Brown @ 2017-08-18 11:47 UTC (permalink / raw)
To: Tony Lindgren
Cc: Sebastian Reichel, Liam Girdwood, Rob Herring, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-omap, devicetree, linux-kernel
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On Thu, Aug 10, 2017 at 09:53:59AM -0700, Tony Lindgren wrote:
> * Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170727 02:02]:
> > I don't follow you here. Why can't other parts of the chip use the
> > same supply? Regarding the other point: Handling the audio-codec
They can, they know they're part of the parent chip and should be
looking for the supply on the chip level.
> > differently than all other sub-modules of cpcap seems much more
> > problematic to me and the codec is basically the last one
> > missing:
> Mark, any comments on the above? I'm just wondering if the
> related dts changes are safe for me to pick.
No, they don't make sense.
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^ permalink raw reply [flat|nested] 22+ messages in thread
* Re: [PATCHv3 2/6] dt-bindings: sound: add motorola, cpcap-audio-codec
@ 2017-08-18 11:47 ` Mark Brown
0 siblings, 0 replies; 22+ messages in thread
From: Mark Brown @ 2017-08-18 11:47 UTC (permalink / raw)
To: Tony Lindgren
Cc: devicetree, alsa-devel, linux-omap, linux-kernel, Takashi Iwai,
Liam Girdwood, Rob Herring, Sebastian Reichel
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On Thu, Aug 10, 2017 at 09:53:59AM -0700, Tony Lindgren wrote:
> * Sebastian Reichel <sebastian.reichel@collabora.co.uk> [170727 02:02]:
> > I don't follow you here. Why can't other parts of the chip use the
> > same supply? Regarding the other point: Handling the audio-codec
They can, they know they're part of the parent chip and should be
looking for the supply on the chip level.
> > differently than all other sub-modules of cpcap seems much more
> > problematic to me and the codec is basically the last one
> > missing:
> Mark, any comments on the above? I'm just wondering if the
> related dts changes are safe for me to pick.
No, they don't make sense.
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^ permalink raw reply [flat|nested] 22+ messages in thread
end of thread, other threads:[~2017-08-18 11:47 UTC | newest]
Thread overview: 22+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2017-07-25 15:10 [PATCHv3 0/6] Motorola Droid 4 Audio Support Sebastian Reichel
2017-07-25 15:10 ` Sebastian Reichel
2017-07-25 15:10 ` [PATCHv3 1/6] ALSA: pcm: Export soc_dpcm_runtime_update Sebastian Reichel
2017-07-25 15:42 ` Takashi Iwai
2017-07-25 15:42 ` Takashi Iwai
2017-07-26 11:38 ` Mark Brown
2017-07-27 10:01 ` Sebastian Reichel
2017-07-27 10:01 ` Sebastian Reichel
2017-07-25 15:10 ` [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec Sebastian Reichel
2017-07-26 11:48 ` Mark Brown
2017-07-26 11:48 ` Mark Brown
2017-07-27 9:01 ` Sebastian Reichel
2017-07-27 9:01 ` Sebastian Reichel
2017-08-10 16:53 ` Tony Lindgren
2017-08-10 16:53 ` [PATCHv3 2/6] dt-bindings: sound: add motorola, cpcap-audio-codec Tony Lindgren
2017-08-18 11:47 ` [PATCHv3 2/6] dt-bindings: sound: add motorola,cpcap-audio-codec Mark Brown
2017-08-18 11:47 ` [PATCHv3 2/6] dt-bindings: sound: add motorola, cpcap-audio-codec Mark Brown
2017-07-25 15:10 ` [PATCHv3 3/6] ASoC: codec: cpcap: new codec Sebastian Reichel
2017-07-25 15:10 ` [PATCHv3 4/6] ARM: dts: motorola-cpcap-mapphone: add audio-codec Sebastian Reichel
2017-07-25 15:10 ` [PATCHv3 5/6] ARM: dts: motorola-cpcap-mapphone: set initial mode for vaudio Sebastian Reichel
2017-07-25 15:10 ` [PATCHv3 6/6] ARM: dts: omap4-droid4: add soundcard Sebastian Reichel
2017-07-27 12:01 ` [PATCHv3 0/6] Motorola Droid 4 Audio Support Tony Lindgren
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