Add gsm phone support for the mixer in tsc2101 alsa driver.

PATCH In review lamikr 5 participants 10 messages Aug 18, 2006 – Sep 26, 2006 Sound

Share & tools

lamikr

author

2006-08-18 01:18 UTC #0 lore raw

PATCH 5/5 Add gsm phone support for the mixer in tsc2101 alsa driver.

Add gsm phone support for the mixer in tsc2101 alsa driver. This allows selecting cellphone for the playback target via alsa mixer interface. Selection will connect gsm module to speaker and microphone allowing the user to speak to phone and listen the opponents voice. Tested in ipaq h63xx by using gomunicator as an application. Tux wanna call home, and now Tux can finally call home :-)

Signed-off-by Mika Laitio <[email protected]>

From: Mika Laitio <[email protected]>
Date: Fri, 18 Aug 2006 03:15:58 +0300
Subject: [PATCH 5/5] add gsm phone support for the mixer in tsc2101 alsa driver.

Signed-off-by Mika Laitio <[email protected]>

---
 sound/arm/omap/omap-alsa-tsc2101-mixer.c |  290 +++++++++++++++++++++++++++++-
 sound/arm/omap/omap-alsa-tsc2101-mixer.h |    3 
 2 files changed, 286 insertions(+), 7 deletions(-)

sound/arm/omap/omap-alsa-tsc2101-mixer.c +284 -6

index 1c7627f..0f92266 100644
@@ -88,7 +88,6 @@ static void set_record_source(int val)
 	
 	/* Mute Analog Sidetone
 	 * Analog sidetone gain db?
-	 * Cell Phone In not connected to ADC
 	 * Input selected by MICSEL connected to ADC
 	 */
 	data	= MPC_ASTMU | MPC_ASTG(0x45);

@@ -417,6 +416,211 @@ void set_headphone_to_playback_target(vo
 	current_playback_target	= PLAYBACK_TARGET_HEADPHONE;
 }
 
+void set_telephone_to_playback_target(void)
+{
+	/* 
+	 * 0110 1101 0101 1100
+	 * power down MICBIAS_HED, Analog sidetone, SPK2, DAC, 
+	 * Driver virtual ground, loudspeaker. Values D2-d5 are flags.
+	 */	 
+	omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
+			CPC_MBIAS_HED | CPC_ASTPWD | CPC_SP2PWDN | CPC_DAPWDN |
+			CPC_VGPWDN | CPC_LSPWDN);
+			
+	/* 
+	 * 0010 1010 0100 0000
+	 * ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+	 * 1dB AGC hysteresis
+	 * MICes bias 2V
+	 */
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4,
+			AC4_MB_HND | AC4_MB_HED(0) | AC4_AGCHYS(1) | 
+			AC4_BISTPD | AC4_ASSTPD | AC4_DASTPD);
+	printk("set_telephone_to_playback_target(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+			
+	/* 
+	 * 1110 0010 0000 0010
+	 * DAC left and right routed to SPK1/SPK2
+	 * SPK1/SPK2 unmuted
+	 * keyclicks routed to SPK1/SPK2
+	 */	 
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
+			AC5_DIFFIN | AC5_DAC2SPK1(3) | 
+		  	AC5_CPI2SPK1 | AC5_MUTSPK2);
+	
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
+			AC6_MIC2CPO | AC6_MUTLSPK | 
+			AC6_LDSCPTC | AC6_VGNDSCPTC | AC6_CAPINTF);
+	current_playback_target	= PLAYBACK_TARGET_CELLPHONE;
+}
+
+/*
+ * 1100 0101 1101 0000
+ * 
+ * #define MPC_ASTMU           TSC2101_BIT(15)
+ * #define MPC_ASTG(ARG)       (((ARG) & 0x7F) << 8)
+ * #define MPC_MICSEL(ARG)     (((ARG) & 0x07) << 5)
+ * #define MPC_MICADC          TSC2101_BIT(4)
+ * #define MPC_CPADC           TSC2101_BIT(3)
+ * #define MPC_ASTGF           (0x01)
+ */
+static void set_telephone_to_record_source(void)
+{
+	u16	val;
+	
+	/* 
+	 * D0       = 0: 
+	 * 		--> AGC is off for handset input.
+	 *		--> ADC PGA is controlled by the ADMUT_HDN + ADPGA_HND
+	 *          (D15, D14-D8)
+	 * D4 - D1  = 0000 
+	 * 		--> AGC time constant for handset input, 
+	 * 		attack time = 8 mc, decay time = 100 ms
+	 * D7 - D5  = 000
+	 * 		--> AGC Target gain for handset input = -5.5 db
+	 * D14 - D8 = 011 1100
+	 * 		--> ADC handset PGA settings = 60 = 30 db
+	 * D15 		= 0
+	 * 		--> Handset input ON (unmuted)
+	 */
+	val	= 0x3c00;	// 0011 1100 0000 0000 = 60 = 30
+	omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
+	
+	/*
+	 * D0		= 0
+	 * 		--> AGC is off for headset/Aux input
+	 * 		--> ADC headset/Aux PGA is contoller by ADMUT_HED + ADPGA_HED
+	 *          (D15, D14-D8)
+	 * D4 - D1	= 0000 
+	 * 		--> Agc constant for headset/Aux input,
+	 *      	attack time = 8 mc, decay time = 100 ms      
+	 * D7 - D5	= 000
+	 * 		--> AGC target gain for headset input = -5.5 db
+	 * D14 - D8 = 000 0000
+	 * 		--> Adc headset/AUX pga settings = 0 db
+	 * D15		= 1
+	 * 		--> Headset/AUX input muted
+	 * 
+	 * Mute headset aux input
+	 */
+	val	= 0x8000;	// 1000 0000 0000 0000
+	omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
+	set_record_source(REC_SRC_MICIN_HND_AND_AUX1);
+
+	// hacks start
+	/* D0		= flag, Headset/Aux or handset PGA flag
+	 * 		--> & with 1 (= 1 -->gain applied == pga register settings)
+	 * D1		= 0, DAC channel PGA soft stepping control
+	 * 		--> 0.5 db change every WCLK
+	 * D2		= flag, DAC right channel PGA flag
+	 * 		--> & with 1
+	 * D3		= flag, DAC left channel PGA flag
+	 * 		-- > & with 1
+	 * D7 - D4	= 0001, keyclick length
+	 * 		--> 4 periods key clicks
+	 * D10 - D8 = 100, keyclick frequenzy
+	 * 		--> 1 kHz, 
+	 * D11		= 0, Headset/Aux or handset soft stepping control
+	 * 		--> 0,5 db change every WCLK or ADWS
+	 * D14 -D12 = 100, Keyclick applitude control
+	 * 		--> Medium amplitude
+	 * D15		= 0, keyclick disabled
+	 */
+	val	= omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_2);
+	val	= val & 0x441d;
+	val	= val | 0x4410;	// D14, D10, D4 bits == 1
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_2, val);
+
+	/*
+	 * D0		= 0	(reserved, write always 0)
+	 * D1		= flag,
+	 * 			--> & with 1
+	 * D2 - D5	= 0000 (reserved, write always 0000)
+	 * D6		= 1
+	 * 			--> MICBIAS_HND = 2.0 v
+	 * D8 - D7	= 00
+	 * 			--> MICBIAS_HED = 3.3 v
+	 * D10 - D9	= 01, 
+	 * 			--> Mic AGC hysteric selection = 2 db
+	 * D11		= 1, 
+	 * 			--> Disable buzzer PGA soft stepping
+	 * D12		= 0,
+	 * 			--> Enable CELL phone PGA soft stepping control
+	 * D13		= 1
+	 * 			--> Disable analog sidetone soft stepping control
+	 * D14		= 0
+	 * 			--> Enable DAC PGA soft stepping control
+	 * D15		= 0,
+	 * 			--> Enable headset/Aux or Handset soft stepping control
+	 */
+	val	= omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4);
+	val	= val & 0x2a42;	// 0010 1010 0100 0010
+	val	= val | 0x2a40;	// bits D13, D11, D9, D6 == 1
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, val);
+	printk("set_telephone_to_record_source(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+	/*
+	 * D0		= 0
+	 * 		--> reserved, write always = 0
+	 * D1		= flag, read only
+	 * 		--> & with 1
+	 * D5 - D2	= 1111, Buzzer input PGA settings
+	 * 		--> 0 db
+	 * D6		= 1,
+	 * 		--> power down buzzer input pga
+	 * D7		= flag, read only
+	 * 		--> & with 1
+	 * D14 - D8	= 101 1101
+	 * 		--> 12 DB
+	 * D15		= 0
+	 * 		--> power up cell phone input PGA
+	 */
+	val	= omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+	val	= val & 0x5dfe;
+	val	= val | 0x5dfe;	// bits, D14, D12, D11, D10, D8, D6, D5,D4,D3,D2
+	omap_tsc2101_audio_write(TSC2101_BUZZER_GAIN_CTRL, val);
+	
+	/* D6 - D0	= 000 1001
+	 * 		--> -4.5 db for DAC right channel volume control
+	 * D7		= 1
+	 * 		-->  DAC right channel muted
+	 * D14 - D8 = 000 1001
+	 * 		--> -4.5 db for DAC left channel volume control
+	 * D15 		= 1
+	 * 		--> DAC left channel muted
+	 */
+	//val	= omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
+	val	= 0x8989;
+	omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);	
+	
+	/*  0000 0000 0100 0000
+	 * 
+	 * D1 - D0	= 0
+	 * 		--> GPIO 1 pin output is three stated
+	 * D2		= 0
+	 * 		--> Disaple GPIO2 for CLKOUT mode
+	 * D3		= 0
+	 * 		--> Disable GPUI1 for interrupt detection
+	 * D4		= 0
+	 * 		--> Disable GPIO2 for headset detection interrupt
+	 * D5		= reserved, always 0
+	 * D7 - D6	= 01
+	 * 		--> 8 ms clitch detection
+	 * D8		= reserved, write only 0
+	 * D10 -D9	= 00
+	 * 		--> 16 ms de bouncing programmatitily 
+	 *          for glitch detection during headset detection
+	 * D11		= flag for button press
+	 * D12		= flag for headset detection
+	 * D14-D13	= 00
+	 * 		--> type of headset detected = 00 == no stereo headset deected
+	 * D15		= 0
+	 * 		--> Disable headset detection
+	 * 
+	 * */
+	val	= 0x40;
+	omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);	
+}
+
 /*
  * Checks whether the headset is detected.
  * If headset is detected, the type is returned. Type can be

@@ -498,7 +702,7 @@ void snd_omap_init_mixer(void)
 static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
 {
 	static char *texts[PLAYBACK_TARGET_COUNT] = {
-        	"Loudspeaker", "Headphone"
+        	"Loudspeaker", "Headphone", "Cellphone"
 	};
 
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;

@@ -528,12 +732,18 @@ static int __pcm_playback_target_put(snd
 	if ((curVal >= 0) &&
 	    (curVal < PLAYBACK_TARGET_COUNT) &&
 	    (curVal != current_playback_target)) {		
-		if (curVal == 0) {
-			set_loudspeaker_to_playback_target();		
+		if (curVal == PLAYBACK_TARGET_LOUDSPEAKER) {
+			set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
+			set_loudspeaker_to_playback_target();
 		}
-		else {
+		else if (curVal == PLAYBACK_TARGET_HEADPHONE) {
+			set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
 			set_headphone_to_playback_target();
 		}
+		else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
+			set_telephone_to_record_source();
+			set_telephone_to_playback_target();
+		}
 		retVal	= 1;
 	}
 	return retVal;

@@ -726,6 +936,58 @@ static int __handset_playback_switch_put
 				15);
 }
 
+static int __cellphone_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+{
+	uinfo->type 			= SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count			= 1;
+	uinfo->value.integer.min	= 0;
+	uinfo->value.integer.max	= 1;
+	return 0;
+}
+
+/* When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
+ * When BGC_MUT_CP = 0, power up cellphone input pga.
+ */
+static int __cellphone_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+{
+	u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+	ucontrol->value.integer.value[0]	= IS_UNMUTED(15, val);
+	return 0;
+}
+
+static int __cellphone_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+{
+	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+				TSC2101_BUZZER_GAIN_CTRL,
+				15);	
+}
+
+static int __buzzer_input_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+{
+	uinfo->type 			= SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count			= 1;
+	uinfo->value.integer.min	= 0;
+	uinfo->value.integer.max	= 1;
+	return 0;
+}
+
+/* When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
+ * When BGC_MUT_BU = 0, power up cellphone input pga.
+ */
+static int __buzzer_input_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+{
+	u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+	ucontrol->value.integer.value[0]	= IS_UNMUTED(6, val);
+	return 0;
+}
+
+static int __buzzer_input_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+{
+	return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+				TSC2101_BUZZER_GAIN_CTRL,
+				6);	
+}
+
 static snd_kcontrol_new_t tsc2101_control[] __devinitdata = {
 	{
 		.name  = "Target Playback Route",

@@ -783,7 +1045,23 @@ static snd_kcontrol_new_t tsc2101_contro
 		.info  = __handset_playback_switch_info,
 		.get   = __handset_playback_switch_get,
 		.put   = __handset_playback_switch_put,
-	}	
+	}, {
+		.name  = "Cellphone Input Switch",
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index = 0,
+		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info  = __cellphone_input_switch_info,
+		.get   = __cellphone_input_switch_get,
+		.put   = __cellphone_input_switch_put,
+	}, {
+		.name  = "Buzzer Input Switch",
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.index = 0,
+		.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info  = __buzzer_input_switch_info,
+		.get   = __buzzer_input_switch_get,
+		.put   = __buzzer_input_switch_put,
+	}
 };
 
 #ifdef CONFIG_PM

sound/arm/omap/omap-alsa-tsc2101-mixer.h +2 -1

index ff852b5..3f27e27 100644
@@ -56,9 +56,10 @@ #define INPUT_VOLUME_MIN 		0x0
 #define INPUT_VOLUME_MAX		0x7D
 #define INPUT_VOLUME_RANGE		(INPUT_VOLUME_MAX - INPUT_VOLUME_MIN)
 
-#define PLAYBACK_TARGET_COUNT		0x02
+#define PLAYBACK_TARGET_COUNT		0x03
 #define PLAYBACK_TARGET_LOUDSPEAKER	0x00
 #define PLAYBACK_TARGET_HEADPHONE	0x01
+#define PLAYBACK_TARGET_CELLPHONE	0x02
 
 /* following are used for register 03h Mixer PGA control bits D7-D5 for selecting record source */
 #define REC_SRC_TARGET_COUNT		0x08

Signature · 1 line

1.4.2

[email protected]

2006-09-05 15:18 UTC #1 lore raw

Quoted 6 lines · lamikr

Add gsm phone support for the mixer in tsc2101 alsa driver. This allows selecting cellphone for the playback target via alsa mixer interface. Selection will connect gsm module to speaker and microphone allowing the user to speak to phone and listen the opponents voice. Tested in ipaq h63xx by using gomunicator as an application. Tux wanna call home, and now Tux can finally call home :-)

Cool that you got the phone features working :)

Tony

lamikr

author

2006-09-06 20:07 UTC #2 lore raw

Quoted 6 lines

Quoted 3 lines
now Tux can finally call home :-)
Cool that you got the phone features working :)

Yes, and gprs is also working nicely with the pppd so I can finally start eating dog food :-)
Some problems there still are

1) As we do not yet have any kind of multiplexing support to gsm module (currently directly accesing dev/ttyS1 for at commands)
our phone app is not able to run simultaneously with the ppp. I am not sure should I resolve this in the kernel space or user space.

2) I can not suspend/resume by using power button but we have still problem from waking up the iPAQ
from suspend for the incoming calls.
So far I have tried to enable UART2 wakeup by using

    level2_wake |= OMAP_IRQ_BIT(INT_MPUIO);

in omap_pm_wakeup_setup() method of plat-omap/pm.c but did not have success with that.
Could you have any idea for this one?

Mika

Pavel Machek

2006-09-07 19:31 UTC #3 lore raw

Hi!

Quoted 5 lines

Quoted 5 lines
Quoted 2 lines
now Tux can finally call home :-)
Cool that you got the phone features working :)

Quoted 8 lines

Yes, and gprs is also working nicely with the pppd so I can finally start eating dog food :-)
Some problems there still are
1) As we do not yet have any kind of multiplexing support to gsm module (currently directly accesing dev/ttyS1 for at commands)
our phone app is not able to run simultaneously with the ppp. I am not sure should I resolve this in the kernel space or user space.

Userspace, I'd say. I've seen that package (google for gsm 07.14, IIRC), but could not get it to work.

Pavel

Signature · 1 line

Thanks for all the (sleeping) penguins.

Vladimir

2006-09-25 11:42 UTC #4 lore raw

In reply to lamikr

Quoted 5 lines

1) As we do not yet have any kind of multiplexing support to gsm module (currently directly accesing dev/ttyS1 for at commands)
our phone app is not able to run simultaneously with the ppp. I am not sure should I resolve this in the kernel space or user space.

I work on getting linux running on Siemens SX1 mobile phone. and I use GSM multiplexer daemon from here -
http://developer.berlios.de/projects/gsmmux/
it works fine for me.

lamikr

author

2006-09-25 16:12 UTC #5 lore raw

Quoted 10 lines · Vladimir

Quoted 5 lines
1) As we do not yet have any kind of multiplexing support to gsm module (currently directly accesing dev/ttyS1 for at commands)
our phone app is not able to run simultaneously with the ppp. I am not sure should I resolve this in the kernel space or user space.
I work on getting linux running on Siemens SX1 mobile phone. and I use GSM multiplexer daemon from here -
http://developer.berlios.de/projects/gsmmux/
it works fine for me.

Have you tried to use pppd over that?
Currently the userspace apps are not in handy enought shape and we need for example to close the phone app
(http://xanadux.cvs.sourceforge.net/xanadux/gomunicator/)
before establishing gprs connection with pppd.
(http://www.handhelds.org/moin/moin.cgi/hpipaqh6300gprs)

Mika

Vovan

2006-09-25 17:12 UTC #6 lore raw

Quoted 7 lines · 25.09.06 в 20:12 lamikr wrote:

Quoted 6 lines · Vladimir
Quoted 5 lines
1) As we do not yet have any kind of multiplexing support to gsm module (currently directly accesing dev/ttyS1 for at commands)
our phone app is not able to run simultaneously with the ppp. I am not sure should I resolve this in the kernel space or user space.

Quoted 5 lines

Quoted 4 lines
I work on getting linux running on Siemens SX1 mobile phone. and I use GSM multiplexer daemon from here -
http://developer.berlios.de/projects/gsmmux/
it works fine for me.
Have you tried to use pppd over that?

No, I havn`t tried pppd yet. I am still implementing IPC protocols with modem that control sound,battery etc. But sending plain AT commands works fine.

Quoted 5 lines

Currently the userspace apps are not in handy enought shape and we need for example to close the phone app
(http://xanadux.cvs.sourceforge.net/xanadux/gomunicator/)
before establishing gprs connection with pppd.
(http://www.handhelds.org/moin/moin.cgi/hpipaqh6300gprs)

I am still far from such userspace support :)

Quoted 4 lines

Mika

Pavel Machek

2006-09-26 19:45 UTC #7 lore raw

In reply to Vladimir

Hi!

Quoted 11 lines

Quoted 8 lines
1) As we do not yet have any kind of multiplexing support to gsm module
(currently directly accesing dev/ttyS1 for at commands) our phone app is not able to run simultaneously with the ppp. I am not
sure should I resolve this in the kernel space or user space.
I work on getting linux running on Siemens SX1 mobile phone.

Do you have web pages with current state somewhere? SX1 should be *cheap* toy for experiments...
Pavel

Signature · 1 line

Thanks for all the (sleeping) penguins.

Vovan

2006-09-26 20:25 UTC #8 lore raw

Quoted 8 lines

Quoted 4 lines
Quoted 1 line
I work on getting linux running on Siemens SX1 mobile phone.
Do you have web pages with current state somewhere? SX1 should be *cheap* toy for experiments...
Pavel

Sure:
http://www.handhelds.org/moin/moin.cgi/SiemensSX1

Pavel Machek

2006-09-26 20:48 UTC #9 lore raw

Hi!

Quoted 5 lines

Quoted 5 lines
Quoted 2 lines
I work on getting linux running on Siemens SX1 mobile phone.
Do you have web pages with current state somewhere? SX1 should be *cheap* toy for experiments...

I googled a bit, and its price in czech republic is ~$100..$140.

Quoted 2 lines

Sure:
http://www.handhelds.org/moin/moin.cgi/SiemensSX1

Thanks! ...so you were able to get most of stuff working, but do not have qt/gpe working, and probably do not have suspend working so battery life is really bad, right?

OTOH you can dual boot with symbian... so battery life is not _as_ bad problem. Do you need windows machine to apply patch allowing you to run uboot? Is there easy way to tell difference between 32MB and 24MB sx1's? Getting 24MB part would spoil all the fun, I'm afraid...

Pavel

Signature · 2 lines

(english) http://www.livejournal.com/~pavelmachek
(cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html