How to select between mixer route elements?

How to select between mixer route elements?

[lamikr]

Hi

I am working with the tsc2101 alsa sound driver for the iPAQ h6300 and
userspace phone application called gomunicator for that.
For the tsc2101 mixer I have creted an element "Playback Playback Route"
which is currently initalized to be a enumerated element
in a following way.

    static char *texts[PLAYBACK_TARGET_COUNT] = {
            "Loudspeaker", "Headphone", "Cellphone"
    };
    uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;

The idea is following. If I/user want to play music or record by using
normal speaker and mic I need to select "Loudspeaker"
as a playback target. If I instead want to use headphone for that
purpose I need to select Headphone. And when I want to have a phone call
(this iPAQ module has gsm phone) I route tsc2101 to handle gsm module by
selecting "Cellphone".

The problem is that currently I need to use alsa-mixer before I make the
phone calls which is kind of not very handy.
So to get Tux finally a call home properly I should do something like
this from the userspace:

    old_selection    = get_current_playback_route_selection(),
    select_playback_route("Cellphone");
    make_call()
    end_call()
    select_playback_route(old_selection)

To implement the currently selected route item, I have tried following

    static int *old_playback_item;
    ...
    snd_mixer_selem_id_set_name (sid, "Playback");
    if (elem_playback) {
        printf("Playback found\n");
        if (snd_mixer_selem_is_enumerated(elem_playback)) {
            printf("Playback is enumerated\n");
            //snd_mixer_selem_get_enum_item(handle, 0, old_playback_item);
            printf("index fo the selected playback item = %d\n",
*old_playback_item);
        }
    }

but this fails for me giving a following error to output:

    Playback found
    Playback is enumerated
    gomunicator: simple.c:1969: snd_mixer_selem_get_enum_item: Assertion
`elem->type == SND_MIXER_ELEM_SIMPLE' failed.

Could somebody show how to do that in the alsa-lib? (I have not been
able to find any documentation/examples how to handle alsa route
elements when I tried to search from the net)

Mika



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Re: How to select between mixer route elements?

[Takashi Iwai]

At Tue, 02 May 2006 01:06:00 +0300,
lamikr wrote:
> 
> Hi
> 
> I am working with the tsc2101 alsa sound driver for the iPAQ h6300 and
> userspace phone application called gomunicator for that.
> For the tsc2101 mixer I have creted an element "Playback Playback Route"

The name sounds strange.  Isn't it "Playback Route"?

> which is currently initalized to be a enumerated element
> in a following way.
> 
>     static char *texts[PLAYBACK_TARGET_COUNT] = {
>             "Loudspeaker", "Headphone", "Cellphone"
>     };
>     uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
> 
> The idea is following. If I/user want to play music or record by using
> normal speaker and mic I need to select "Loudspeaker"
> as a playback target. If I instead want to use headphone for that
> purpose I need to select Headphone. And when I want to have a phone call
> (this iPAQ module has gsm phone) I route tsc2101 to handle gsm module by
> selecting "Cellphone".
> 
> The problem is that currently I need to use alsa-mixer before I make the
> phone calls which is kind of not very handy.
> So to get Tux finally a call home properly I should do something like
> this from the userspace:
> 
>     old_selection    = get_current_playback_route_selection(),
>     select_playback_route("Cellphone");
>     make_call()
>     end_call()
>     select_playback_route(old_selection)

It can be easily done via PCM hooks plugin.  The hooks plugin can set 
up the mixer controls after opening a slave PCM device.
For example, "cellphone" PCM below will automatically change "Playback
Route" control to "Cellphone" and keep while it's opened, then revert
to the old value after closed.

	pcm.cellphone {
		type hooks
		slave.pcm "hw:0"
		hooks.0 {
			type ctl_elems
			hook_args [
				{
					name "Playback Route"
					lock true
					preverve true
					value "Cellphone"
				}
			]
		}
	}


> To implement the currently selected route item, I have tried following
> 
>     static int *old_playback_item;
>     ...
>     snd_mixer_selem_id_set_name (sid, "Playback");
>     if (elem_playback) {
>         printf("Playback found\n");
>         if (snd_mixer_selem_is_enumerated(elem_playback)) {
>             printf("Playback is enumerated\n");
>             //snd_mixer_selem_get_enum_item(handle, 0, old_playback_item);
>             printf("index fo the selected playback item = %d\n",
> *old_playback_item);
>         }
>     }
> 
> but this fails for me giving a following error to output:
> 
>     Playback found
>     Playback is enumerated
>     gomunicator: simple.c:1969: snd_mixer_selem_get_enum_item: Assertion
> `elem->type == SND_MIXER_ELEM_SIMPLE' failed.

This is likely a conflict of name space of "Playback" controls.
Do you have other controls with "Playback XXX"?


Takashi


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Re: How to select between mixer route elements?

[lamikr]

Takashi Iwai wrote:

>At Tue, 02 May 2006 01:06:00 +0300,
>lamikr wrote:
>  
>
>>Hi
>>
>>I am working with the tsc2101 alsa sound driver for the iPAQ h6300 and
>>userspace phone application called gomunicator for that.
>>For the tsc2101 mixer I have creted an element "Playback Playback Route"
>>    
>>
>
>The name sounds strange.  Isn't it "Playback Route"?
>  
>
Hmm, did I understood the "Control Names" chapter from the "Writing an
alsa driver" pdf page 46
incorrectly? I understood that each name is defined from "SOURCE
DIRECTION FUNCTION".
So can the "Source" part just be dropped from the route function names?

Is the route with custom element names the recommend ways to make the
alsa mixer interface, if the user can select between multiple playback
targets
but all of them can not be switched on simultaneously? Or should I just
make sure in the "Switch" methods that when one control is unmuted, others
are muted?

>>which is currently initalized to be a enumerated element
>>in a following way.
>>
>>    static char *texts[PLAYBACK_TARGET_COUNT] = {
>>            "Loudspeaker", "Headphone", "Cellphone"
>>    };
>>    uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
>>
>>The idea is following. If I/user want to play music or record by using
>>normal speaker and mic I need to select "Loudspeaker"
>>as a playback target. If I instead want to use headphone for that
>>purpose I need to select Headphone. And when I want to have a phone call
>>(this iPAQ module has gsm phone) I route tsc2101 to handle gsm module by
>>selecting "Cellphone".
>>
>>The problem is that currently I need to use alsa-mixer before I make the
>>phone calls which is kind of not very handy.
>>So to get Tux finally a call home properly I should do something like
>>this from the userspace:
>>
>>    old_selection    = get_current_playback_route_selection(),
>>    select_playback_route("Cellphone");
>>    make_call()
>>    end_call()
>>    select_playback_route(old_selection)
>>    
>>
>
>It can be easily done via PCM hooks plugin.  The hooks plugin can set 
>up the mixer controls after opening a slave PCM device.
>For example, "cellphone" PCM below will automatically change "Playback
>Route" control to "Cellphone" and keep while it's opened, then revert
>to the old value after closed.
>
>	pcm.cellphone {
>		type hooks
>		slave.pcm "hw:0"
>		hooks.0 {
>			type ctl_elems
>			hook_args [
>				{
>					name "Playback Route"
>					lock true
>					preverve true
>					value "Cellphone"
>				}
>			]
>		}
>	}
>  
>
Hmm, this looks as a smart option to do it. Any pointers for the example
code/documentation how to open the slave PCM devices as I am now working
for a first time with the alsa user space code.

>>To implement the currently selected route item, I have tried following
>>
>>    static int *old_playback_item;
>>    ...
>>    snd_mixer_selem_id_set_name (sid, "Playback");
>>    if (elem_playback) {
>>        printf("Playback found\n");
>>        if (snd_mixer_selem_is_enumerated(elem_playback)) {
>>            printf("Playback is enumerated\n");
>>            snd_mixer_selem_get_enum_item(handle, 0, old_playback_item);
>>            printf("index fo the selected playback item = %d\n",
>>*old_playback_item);
>>        }
>>    }
>>
>>but this fails for me giving a following error to output:
>>
>>    Playback found
>>    Playback is enumerated
>>    gomunicator: simple.c:1969: snd_mixer_selem_get_enum_item: Assertion
>>`elem->type == SND_MIXER_ELEM_SIMPLE' failed.
>>    
>>
>
>This is likely a conflict of name space of "Playback" controls.
>Do you have other controls with "Playback XXX"?
>  
>
Yes, I have actually many of them as the tsc2101 codec has multiple adc
and dgc registers to control the volume. I tried to change the "playback
playback route" to simple be a "Playback Route" but I still got the same
error from the method snd_mixer_selem_get_enum_item(handle, 0,
old_playback_item); method.

Below is my whole mixer code I currently have.

/*
* sound/arm/omap/omap-alsa-tsc2101-mixer.c
*
* Alsa Driver for TSC2101 codec for OMAP platform boards.
*
* Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
* Everett Coleman II <gcc80x86@fuzzyneural.net>
*
* Board initialization code is based on the code in TSC2101 OSS driver.
* Copyright (C) 2004 Texas Instruments, Inc.
* Written by Nishanth Menon and Sriram Kannan
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
* NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
* NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
* USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
* ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 675 Mass Ave, Cambridge, MA 02139, USA.
*
* History:
*
* 2006-03-01 Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
* Can switch between headset and loudspeaker playback,
* mute and unmute dgc, set dgc volume. Record source switch,
* keyclick, buzzer and headset volume and handset volume control
* are still missing.
*
*/

#include "omap-alsa-tsc2101.h"
#include "omap-alsa-tsc2101-mixer.h"

#include <linux/types.h>
#include <sound/initval.h>
#include <sound/control.h>

//#define M_DPRINTK(ARGS...) printk(KERN_INFO "<%s>:
",__FUNCTION__);printk(ARGS)
#define M_DPRINTK(ARGS...) /* nop */

#define CHECK_BIT(INDX, ARG) (((ARG) & TSC2101_BIT(INDX)) >> INDX)
#define IS_UNMUTED(INDX, ARG) (((CHECK_BIT(INDX, ARG)) == 0))

#define DGC_DALVL_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
#define DGC_DARVL_EXTRACT(ARG) ((ARG & 0x007f))

#define HGC_ADPGA_HED_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
#define HNGC_ADPGA_HND_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
#define BGC_ADPGA_BGC_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)

static int current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
static int current_rec_src = REC_SRC_SINGLE_ENDED_MICIN_HED;

/*
* Simplified write for the tsc2101 audio registers.
*/
inline void omap_tsc2101_audio_write(u8 address, u16 data)
{
omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS, address, data);
}

/*
* Simplified read for the tsc2101 audio registers.
*/
inline u16 omap_tsc2101_audio_read(u8 address)
{
return (omap_tsc2101_read(PAGE2_AUDIO_CODEC_REGISTERS, address));
}

/*
* For selecting tsc2101 recourd source.
*/
static void set_record_source(int val)
{
u16 data;

/* 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);
data &= ~MPC_MICSEL(7); /* clear all MICSEL bits */
data |= MPC_MICSEL(val);
data |= MPC_MICADC;
omap_tsc2101_audio_write(TSC2101_MIXER_PGA_CTRL, data);

current_rec_src = val;
}

/*
* Converts the Alsa mixer volume (0 - 100) to real
* Digital Gain Control (DGC) value that can be written
* or read from the TSC2101 registry.
*
* Note that the number "OUTPUT_VOLUME_MAX" is smaller than OUTPUT_VOLUME_MIN
* because DGC works as a volume decreaser. (The more bigger value is put
* to DGC, the more the volume of controlled channel is decreased)
*
* In addition the TCS2101 chip would allow the maximum volume reduction
be 63.5 DB
* but according to some tests user can not hear anything with this chip
* when the volume is set to be less than 25 db.
* Therefore this function will return a value that means 38.5 db (63.5
db - 25 db)
* reduction in the channel volume, when mixer is set to 0.
* For mixer value 100, this will return a value that means 0 db volume
reduction.
* ([mute_left_bit]0000000[mute_right_bit]0000000)
*/
int get_mixer_volume_as_dac_gain_control_volume(int vol)
{
u16 retVal;

/* Convert 0 -> 100 volume to 0x7F(min) -> y(max) volume range */
retVal = ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
/* invert the value for getting the proper range 0 min and 100 max */
retVal = OUTPUT_VOLUME_MIN - retVal;

return retVal;
}

/*
* Converts the Alsa mixer volume (0 - 100) to TSC2101
* Digital Gain Control (DGC) volume. Alsa mixer volume 0
* is converted to value meaning the volume reduction of -38.5 db
* and Alsa mixer volume 100 is converted to value meaning the
* reduction of 0 db.
*/
int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int
mixerVolR)
{
u16 val;
int retVal;
int volL;
int volR;

if ((mixerVolL < 0) ||
(mixerVolL > 100) ||
(mixerVolR < 0) ||
(mixerVolR > 100)) {
printk(KERN_ERR "Trying a bad mixer volume as dac gain control volume
value, left (%d), right (%d)!\n", mixerVolL, mixerVolR);
return -EPERM;
}
M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);
volL = get_mixer_volume_as_dac_gain_control_volume(mixerVolL);
volR = get_mixer_volume_as_dac_gain_control_volume(mixerVolR);

val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
/* keep the old mute bit settings */
val &= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) | DGC_DARVL(OUTPUT_VOLUME_MIN));
val |= DGC_DALVL(volL) | DGC_DARVL(volR);
retVal = 2;
if (retVal) {
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
}
M_DPRINTK("to registry: left = %d, right = %d, total = %d\n",
DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
return retVal;
}

/**
* If unmuteLeft/unmuteRight == 0 --> mute
* If unmuteLeft/unmuteRight == 1 --> unmute
*/
int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
{
u16 val;
int count;

count = 0;
val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
* so if values are same, it's time to change the registry value.
*/
if (unmuteLeft != IS_UNMUTED(15, val)) {
if (unmuteLeft == 0) {
/* mute --> turn bit on */
val = val | DGC_DALMU;
}
else {
/* unmute --> turn bit off */
val = val & ~DGC_DALMU;
}
count++;
} /* L */
if (unmuteRight != IS_UNMUTED(7, val)) {
if (unmuteRight == 0) {
/* mute --> turn bit on */
val = val | DGC_DARMU;
}
else {
/* unmute --> turn bit off */
val = val & ~DGC_DARMU;
}
count++;
} /* R */
if (count) {
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
IS_UNMUTED(15, val),
IS_UNMUTED(7, val));
}
return count;
}

/**
* unmute: 0 --> mute, 1 --> unmute
* page2RegIndx: Registry index in tsc2101 page2.
* muteBitIndx: Index number for the bit in registry that indicates
whether muted or unmuted.
*/
int adc_pga_unmute_control(int unmute, int page2regIndx, int muteBitIndx)
{
int count;
u16 val;

count = 0;
val = omap_tsc2101_audio_read(page2regIndx);
/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
* so if the values are same, it's time to change the registry value...
*/
if (unmute != IS_UNMUTED(muteBitIndx, val)) {
if (unmute == 0) {
/* mute --> turn bit on */
val = val | TSC2101_BIT(muteBitIndx);
}
else {
/* unmute --> turn bit off */
val = val & ~TSC2101_BIT(muteBitIndx);
}
M_DPRINTK("changed value, is_unmuted = %d\n", IS_UNMUTED(muteBitIndx, val));
count++;
}
if (count) {
omap_tsc2101_audio_write(page2regIndx, val);
}
return count;
}

/*
* Converts the DGC registry value read from the TSC2101 registry to
* Alsa mixer volume format (0 - 100).
*/
int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
{
u16 retVal;

retVal = OUTPUT_VOLUME_MIN - vol;
retVal = ((retVal - OUTPUT_VOLUME_MAX) * 100) / OUTPUT_VOLUME_RANGE;
/* fix scaling error */
if ((retVal > 0) && (retVal < 100)) {
retVal++;
}
return retVal;
}

/*
* Converts the headset gain control volume (0 - 63.5 db)
* to Alsa mixer volume (0 - 100)
*/
int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
{
u16 retVal;

retVal = ((registerVal * 100) / INPUT_VOLUME_RANGE);
return retVal;
}

/*
* Converts the handset gain control volume (0 - 63.5 db)
* to Alsa mixer volume (0 - 100)
*/
int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal)
{
return get_headset_gain_control_volume_as_mixer_volume(registerVal);
}

/*
* Converts the Alsa mixer volume (0 - 100) to
* headset gain control volume (0 - 63.5 db)
*/
int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal)
{
u16 retVal;

retVal = ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;
return retVal;
}

/*
* Writes Alsa mixer volume (0 - 100) to TSC2101 headset volume registry in
* a TSC2101 format. (0 - 63.5 db)
* In TSC2101 OSS driver this functionality was controlled with
"SET_LINE" parameter.
*/
int set_mixer_volume_as_headset_gain_control_volume(int mixerVol)
{
int volume;
int retVal;
u16 val;

if (mixerVol < 0 || mixerVol > 100) {
M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n", mixerVol);
return -EPERM;
}
M_DPRINTK("mixer volume = %d\n", mixerVol);
/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range */
/* NOTE: 0 is minimum volume and not mute */
volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
/* preserve the old mute settings */
val &= ~(HGC_ADPGA_HED(INPUT_VOLUME_MAX));
val |= HGC_ADPGA_HED(volume);
omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
retVal = 1;

M_DPRINTK("to registry = %d\n", val);
return retVal;
}

/*
* Writes Alsa mixer volume (0 - 100) to TSC2101 handset volume registry in
* a TSC2101 format. (0 - 63.5 db)
* In TSC2101 OSS driver this functionality was controlled with "SET_MIC"
parameter.
*/
int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
{
int volume;
int retVal;
u16 val;

if (mixerVol < 0 || mixerVol > 100) {
M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n", mixerVol);
return -EPERM;
}
M_DPRINTK("mixer volume = %d\n", mixerVol);
/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
* NOTE: 0 is minimum volume and not mute
*/
volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
/* preserve the old mute settigns */
val &= ~(HNGC_ADPGA_HND(INPUT_VOLUME_MAX));
val |= HNGC_ADPGA_HND(volume);
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
retVal = 1;

M_DPRINTK("to registry = %d\n", val);
return retVal;
}

void set_loudspeaker_to_playback_target(void)
{
/* power down SPK1, SPK2 and loudspeaker */
omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
* 1dB AGC hysteresis
* MICes bias 2V
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));

/* 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_AST2SPK1 | AC5_KCL2SPK1 |
AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2);

/* routing selected to SPK1 goes also to OUT8P/OUT8N. (loudspeaker)
* analog sidetone routed to loudspeaker
* buzzer pga routed to loudspeaker
* keyclick routing to loudspeaker
* cellphone input routed to loudspeaker
* mic selection (control register 04h/page2) routed to cell phone output
(CP_OUT)
* routing selected for SPK1 goes also to cellphone output (CP_OUT)
* OUT8P/OUT8N (loudspeakers) unmuted (0 = unmuted)
* Cellphone output is not muted (0 = unmuted)
* Enable loudspeaker short protection control (0 = enable protection)
* VGND short protection control (0 = enable protection)
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
AC6_SPL2LSK | AC6_AST2LSK | AC6_BUZ2LSK | AC6_KCL2LSK |
AC6_CPI2LSK | AC6_MIC2CPO | AC6_SPL2CPO);
current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
}

void set_headphone_to_playback_target(void)
{
/* power down SPK1, SPK2 and loudspeaker */
omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled */
/* 1dB AGC hysteresis */
/* MICes bias 2V */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));

/* 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_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
AC5_HDSCPTC);

/* OUT8P/OUT8N muted, CPOUT muted */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
AC6_VGNDSCPTC);
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);

/*
* 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;

/*disable handset input (d0)*/
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
val = val & 0xfffe;
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
// set volume (d15-d8 to 0)
val = val & 0x80ff;
// increase it
val = val | 0x3b00;
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);

val = 0x8000;
omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);

set_record_source(REC_SRC_MICIN_HND_AND_AUX1);

// hacks start
val = 0x4411;
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_2, val);

val = 0x2a40;
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, val);

// cellphone input pga settings volume (TSC2101_BUZZER_GAIN_CTRL, 1fh)
val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
val = val & 0x807f;
val = val | 0x5d00;
val = val & 0x7fff;
omap_tsc2101_audio_write(TSC2101_BUZZER_GAIN_CTRL, val);

val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
val = val | 0x8189;
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);

val = 0x40;
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);
// hacks end
}

/*
* Checks whether the headset is detected.
* If headset is detected, the type is returned. Type can be
* 0x01 = stereo headset detected
* 0x02 = cellurar headset detected
* 0x03 = stereo + cellurar headset detected
* If headset is not detected 0 is returned.
*/
u16 get_headset_detected(void)
{
u16 curDetected;
u16 curType;
u16 curVal;

curType = 0; /* not detected */
curVal = omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_7);
curDetected = curVal & AC7_HDDETFL;
if (curDetected) {
printk("headset detected, checking type from %d \n", curVal);
curType = ((curVal & 0x6000) >> 13);
printk("headset type detected = %d \n", curType);
}
else {
printk("headset not detected\n");
}
return curType;
}

void init_playback_targets(void)
{
u16 val;

set_loudspeaker_to_playback_target();
/* Left line input volume control
* = SET_LINE in the OSS driver
*/
set_mixer_volume_as_headset_gain_control_volume(DEFAULT_INPUT_VOLUME);

/* Set headset to be controllable by handset mixer
* AGC enable for handset input
* Handset input not muted
*/
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
val = val | HNGC_AGCEN_HND;
val = val & ~HNGC_ADMUT_HND;
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);

/* mic input volume control
* SET_MIC in the OSS driver
*/
set_mixer_volume_as_handset_gain_control_volume(DEFAULT_INPUT_VOLUME);

/* Left/Right headphone channel volume control
* Zero-cross detect on
*/
set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME,
DEFAULT_OUTPUT_VOLUME);
/* unmute */
dac_gain_control_unmute(1, 1);
}

/*
* Initializes tsc2101 recourd source (to line) and playback target (to
loudspeaker)
*/
void snd_omap_init_mixer(void)
{
FN_IN;

/* Headset/Hook switch detect enabled */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, AC7_DETECT);

/* Select headset to record source (MIC_INHED)*/
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
/* Init loudspeaker as a default playback target*/
init_playback_targets();

FN_OUT(0);
}

static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol,
snd_ctl_elem_info_t *uinfo)
{
static char *texts[PLAYBACK_TARGET_COUNT] = {
"Loudspeaker", "Headphone", "Cellphone"
};

uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
}
strcpy(uinfo->value.enumerated.name,
texts[uinfo->value.enumerated.item]);
return 0;
}

static int __pcm_playback_target_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
ucontrol->value.integer.value[0] = current_playback_target;
return 0;
}

static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
int retVal;
int curVal;

retVal = 0;
curVal = ucontrol->value.integer.value[0];
if ((curVal >= 0) &&
(curVal < PLAYBACK_TARGET_COUNT) &&
(curVal != current_playback_target)) {
if (curVal == PLAYBACK_TARGET_LOUDSPEAKER) {
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
set_loudspeaker_to_playback_target();
}
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;
}

static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol,
snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 100;
return 0;
}

/*
* Alsa mixer interface function for getting the volume read from the DGC
in a
* 0 -100 alsa mixer format.
*/
static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 volL;
u16 volR;
u16 val;

val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
M_DPRINTK("registry value = %d!\n", val);
volL = DGC_DALVL_EXTRACT(val);
volR = DGC_DARVL_EXTRACT(val);
/* make sure that other bits are not on */
volL = volL & ~DGC_DALMU;
volR = volR & ~DGC_DARMU;

volL = get_dac_gain_control_volume_as_mixer_volume(volL);
volR = get_dac_gain_control_volume_as_mixer_volume(volR);

ucontrol->value.integer.value[0] = volL; /* L */
ucontrol->value.integer.value[1] = volR; /* R */

M_DPRINTK("mixer volume left = %ld, right = %ld\n",
ucontrol->value.integer.value[0], ucontrol->value.integer.value[1]);
return 0;
}

static int __pcm_playback_volume_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
return
set_mixer_volume_as_dac_gain_control_volume(ucontrol->value.integer.value[0],

ucontrol->value.integer.value[1]);
}

static int __pcm_playback_switch_info(snd_kcontrol_t *kcontrol,
snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}

/*
* When DGC_DALMU (bit 15) is 1, the left channel is muted.
* When DGC_DALMU is 0, left channel is not muted.
* Same logic apply also for the right channel.
*/
static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);

ucontrol->value.integer.value[0] = IS_UNMUTED(15, val); // left
ucontrol->value.integer.value[1] = IS_UNMUTED(7, val); // right
return 0;
}

static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
return dac_gain_control_unmute(ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
}

static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol,
snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 100;
return 0;
}

static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 val;
u16 vol;

val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
M_DPRINTK("registry value = %d\n", val);
vol = HGC_ADPGA_HED_EXTRACT(val);
vol = vol & ~HGC_ADMUT_HED;

vol = get_headset_gain_control_volume_as_mixer_volume(vol);
ucontrol->value.integer.value[0] = vol;

M_DPRINTK("mixer volume returned = %ld\n",
ucontrol->value.integer.value[0]);
return 0;
}

static int __headset_playback_volume_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
return
set_mixer_volume_as_headset_gain_control_volume(ucontrol->value.integer.value[0]);

}

static int __headset_playback_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 HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
* When HGC_ADMUT_HED is 0, headset is not muted.
*/
static int __headset_playback_switch_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);
return 0;
}

static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
// mute/unmute headset
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_HEADSET_GAIN_CTRL,
15);
}

static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol,
snd_ctl_elem_info_t *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 100;
return 0;
}

static int __handset_playback_volume_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 val;
u16 vol;

val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
M_DPRINTK("registry value = %d\n", val);
vol = HNGC_ADPGA_HND_EXTRACT(val);
vol = vol & ~HNGC_ADMUT_HND;
vol = get_handset_gain_control_volume_as_mixer_volume(vol);
ucontrol->value.integer.value[0] = vol;

M_DPRINTK("mixer volume returned = %ld\n",
ucontrol->value.integer.value[0]);
return 0;
}

static int __handset_playback_volume_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
return
set_mixer_volume_as_handset_gain_control_volume(ucontrol->value.integer.value[0]);

}

static int __handset_playback_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 HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
* When HNGC_ADMUT_HND is 0, handset is not muted.
*/
static int __handset_playback_switch_get(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val);
return 0;
}

static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol,
snd_ctl_elem_value_t *ucontrol)
{
// handset mute/unmute
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_HANDSET_GAIN_CTRL,
15);
}

#define CELLPHONE_INPUT_VOLUME_RANGE 0x5d

/*
* Converts the headset gain control volume (0 - 63.5 db)
* to Alsa mixer volume (0 - 100)
*/
int get_cellphone_input_pga_volume_as_mixer_volume(u16 registerVal)
{
u16 retVal;

retVal = ((registerVal * 100) / CELLPHONE_INPUT_VOLUME_RANGE);
return retVal;
}

/*
* Writes Alsa mixer volume (0 - 100) to TSC2101 handset volume registry in
* a TSC2101 format. (0 - 63.5 db)
* In TSC2101 OSS driver this functionality was controlled with "SET_MIC"
parameter.
*/
int set_mixer_volume_as_cellphone_input_pga_volume(int mixerVol)
{
return 1;
}

static snd_kcontrol_new_t tsc2101_control[] __devinitdata = {
{
.name = "Playback Playback Route",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_target_info,
.get = __pcm_playback_target_get,
.put = __pcm_playback_target_put,
}, {
.name = "Master Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_volume_info,
.get = __pcm_playback_volume_get,
.put = __pcm_playback_volume_put,
}, {
.name = "Master Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_switch_info,
.get = __pcm_playback_switch_get,
.put = __pcm_playback_switch_put,
}, {
.name = "Headset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_volume_info,
.get = __headset_playback_volume_get,
.put = __headset_playback_volume_put,
}, {
.name = "Headset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_switch_info,
.get = __headset_playback_switch_get,
.put = __headset_playback_switch_put,
}, {
.name = "Handset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_volume_info,
.get = __handset_playback_volume_get,
.put = __handset_playback_volume_put,
}, {
.name = "Handset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_switch_info,
.get = __handset_playback_switch_get,
.put = __handset_playback_switch_put,
}
};

#ifdef CONFIG_PM

void snd_omap_suspend_mixer(void)
{
}

void snd_omap_resume_mixer(void)
{
snd_omap_init_mixer();
}
#endif

int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
{
int i=0;
int err=0;

if (!tsc2101) {
return -EINVAL;
}
for (i=0; i < ARRAY_SIZE(tsc2101_control); i++) {
if ((err = snd_ctl_add(tsc2101->card,
snd_ctl_new1(&tsc2101_control[i],
tsc2101->card))) < 0) {
return err;
}
}
return 0;
}


Mika


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Re: How to select between mixer route elements?

[lamikr]

>>>       if (snd_mixer_selem_is_enumerated(elem_playback)) {
>>>           printf("Playback is enumerated\n");
>>>           snd_mixer_selem_get_enum_item(handle, 0, old_playback_item);
>>>      
>>>
Sorry, I found the error myself, it was my typo... :-(
I put the mixer handle for snd_mixer_selem_get_enum_item as a first
parameter when I was supposed to put the element handle. (elem_playback)

Anyway I would appreciate if somebody could comment whether it is ok to
put the selection between 3 different playback targets for the route method
(loudspeaker, headphone, cellphone) instead of putting own mute/unmute
switch control for each of them separately?

Mika


-------------------------------------------------------
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642

Re: How to select between mixer route elements?

[Takashi Iwai]

At Tue, 09 May 2006 02:55:53 +0300,
lamikr wrote:
> 
> 
> >>>       if (snd_mixer_selem_is_enumerated(elem_playback)) {
> >>>           printf("Playback is enumerated\n");
> >>>           snd_mixer_selem_get_enum_item(handle, 0, old_playback_item);
> >>>      
> >>>
> Sorry, I found the error myself, it was my typo... :-(
> I put the mixer handle for snd_mixer_selem_get_enum_item as a first
> parameter when I was supposed to put the element handle. (elem_playback)
> 
> Anyway I would appreciate if somebody could comment whether it is ok to
> put the selection between 3 different playback targets for the route method
> (loudspeaker, headphone, cellphone) instead of putting own mute/unmute
> switch control for each of them separately?

Yes, enum vs switches are just like radio-button and check-box in
GUI.  If the items are mutual, you should use an enum control type.


Takashi


-------------------------------------------------------
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