Audio Routing in ti sdk 7

Srini

Dear TI E2E Community,

I have stereo codec , where I have two MIC input channels to stereo codec & two output (ie., Left & right channels) of the stereo codec, performing the stereo operations where the same I have verified in ti sdk7 (ie., stereo operations)

Could you please let me know how the stereo channels can be converted to two different & independent Mono channels,

as am new to alsa, hope some configurations needs to be done in alsa or does it requires any changes in the codec driver please let me know

Could you please kindly let me know how this can be converted to two different Mono channels from stereo in ti sdk 7

As am trying this from many days couldn't able to succeed

Kindly do the needful as early as possible

Awaiting for your replies

Many Many Thanks in advance

over 9 years ago

0 Ivan Matrakov over 9 years ago

TI__Expert 4475 points

Hi Srini,

If I understand well you are want to split one stereo signal to two mono. This is made in many of audio/video programs, but I advice you to see convertion of ffmpeg program. The conversion is located in: ffmpeg-2.0.1/libavcodec/resample.c file in static void stereo_to_mono(short *output, short *input, int n1). The good here is that you can test how this is work with ffpmeg program and to compare with your results. Of course you have to modify code for your purpose. Also I post you some usefull site to see how this is made:
www.dsprelated.com/.../1.php
www.codeproject.com/.../How-to-convert-between-most-audio-formats-in-NET

BR
Ivan

0 Srini over 9 years ago in reply to Ivan Matrakov

Genius 5250 points

Dear Ivan,

Thanks a lot for your quick replies,

As am new to ffmepg I feel it will be more compilicated rather than hope I I get in low level alsa driver only, could you please let me know whether this can be done in low level alsa driver only??

As actually my requirement is I have to place a GSM call via stereo codec where FAR END & NEAR END communications (ie., FULL DUPLEX - FAR END MIC ---> NEAR END SPEAKER & NEAR END MIC ---> FAR END SPEAKER) needs to established

1) Actually my query is related to I need to convert two ADC channels (of UDA1345TS , Could you please refer the block diagram of UDA1345TS datashee) ie., VINL & VINR as two different mic input channels & two DAC channels ie., VOUTL & VOUTR as two different mono output channels,

As currently it is left & right channels (stereo) (ie., VOUTL & VOUTR as stereo), in my codec driver only ie., sound/soc/codecs/uda134x.c

/* Common DAPM widgets */
static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = {

SND_SOC_DAPM_INPUT("VINL"),
SND_SOC_DAPM_INPUT("VINR"),
SND_SOC_DAPM_OUTPUT("VOUTL"),
SND_SOC_DAPM_OUTPUT("VOUTR"),
};

static const struct snd_soc_dapm_route uda134x_dapm_routes[] = {

{ "ADC", NULL, "VINL" },
{ "ADC", NULL, "VINR" },
{ "VOUTL", NULL, "DAC" },
{ "VOUTR", NULL, "DAC" },
};

Is there any possibilities that above routing which is configured for stereo , how this can be changed to two different mono MIC input channels & two different mono output channels so that (ie., FULL DUPLEX - FAR END MIC ---> NEAR END SPEAKER & NEAR END MIC ---> FAR END SPEAKER) can be achieved??

Could you please let me know how this can be done in low level driver itself ie., in ALSA in ti sdk 7 ie., in my codec driver only ie., sound/soc/codecs/uda134x.c ie., VINL & VINR as two different mic input channels & two DAC channels ie., VOUTL & VOUTR as two different mono output channels,

Is there any thing other than that I need to take care in audio routing part of ALSA or any configuration needs to be done in etc/asound.conf

2) OR am planning to use alsa loop back feature does this is going to resolve my requirement (ie., FULL DUPLEX), if so could you please suggest me how audio loop back can be tested as am not aware how to test this feature ?? I have configured the kernel as below

│ Symbol: SND_ALOOP [=y] │
│ Type : tristate │
│ Prompt: Generic loopback driver (PCM) │
│ Location: │
│ -> Device Drivers │
│ -> Sound card support (SOUND [=y]) │
│ -> Advanced Linux Sound Architecture (SND [=y]) │
│ -> Generic sound devices (SND_DRIVERS [=y]) │

And in in etc/asound.conf

# default device
pcm.!default {
type plug
slave.pcm "loop"
}

Please correct me if my understanding ins wrong inorder to achieve my requirment (ie., FULL DUPLEX ) gsm call on audio

Once again appreciate a lot for all your quick responses

Kindly do the needful as early as possible

Awaiting for your replies

Many Many Thanks in advance again

0 Margarita Gashova over 9 years ago

TI__Guru* 80190 points

Hello,

You could try to use gstreamer for example:

Here is example pipeline which uses alsa for capture then split the two channels and save them in different files:

gst-launch -v alsasrc device="plughw:X,X" ! audio/x-raw-int, signed=true, width=32, depth=32, format=S32LE, rate=96000, channels=2
! deinterleave name=d d.src0 ! queue ! audioconvert ! audio/x-raw-int,channels=1,width=16,depth=16 ! filesink location=first.pcm d.src1 ! queue ! audioconvert ! audio/x-raw-int,channels=1,width=16,depth=16 ! audioresample ! 'audio/x-raw-int, endianness=(int)1234, signed=(boolean)true,rate=(int)44100' ! filesink location=second.pcm

It also will convert the rate etc, which you could change. You shall change this as well: device="plughw:X,X".

BR
Margarita

0 Srini over 9 years ago in reply to Margarita Gashova

Genius 5250 points

Thanks a lot for your quick repsonses

I am planning to verify this feature in low level alsa driver

Could you please let me know with detailed procedure how "SND_ALOOP" feature can be tested & verified in ti sdk 7 ie., alsa loop back I have configured the driver in linux

I have added in etc/asound.conf

# default device
pcm.!default {
type plug
slave.pcm "loop"
}

But am unable to verify this alsa loop back feature with my stereo codec i.e, am unable to record what ever am playing

As I read am going through ALSA loopback tutorial

Could you please get me the full source code of the below explained link

http://www.saunalahti.fi/~s7l/blog/2005/08/21/Full%20Duplex%20ALSA.html

Could you please let me know how to make this code working ie., does this code requires any etc/asound.conf configurations please let me know

Once again appreciate a lot for all your quick responses

Kindly do the needful as early as possible

Awaiting for your replies

Many Many Thanks in advance again

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

1) Could anybody please let me know how the above feature can be verified in alsa loop back feature in ti sdk 7 with detailed steps as early as possible

2) I have a stereo codec with two mic inputs (MIC1 & MIC2) & two ouputs (left & right channels)
Could you please confirm does this alsa loop back corresponds, MIC1 to left channel & MIC2 to right channel ie., Is it possible to feed the MIC1 input to left channel of stereo codec & MIC2 input to right channel of stereo codec using this alsa loop back feature???

Once again Many Thanks & appreciate a lot for all your quick responses

Kindly do the needful as early as possible

Awaiting for your replies

Many Many Thanks in advance again

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

Dear Technical Team,

Please kindly ignore the answer for the above query w.r.t alsa loop back

1) I prefer to take care in low level alsa driver only as am a platform guy & am not aware of gstreamer or ffmeg, could you please first confirm whether the GSM call via stereo codec ie., can it be handled & taken care in the routing part of alsa??

2) I have modified the routing in sound/soc/codecs/uda134x.c inorder to handle a GSM call via stereo codec, could anybody please confirm whether the following configurations are correct or not?? in ti sdk 7 or if this is not the correct link could you please let me know the respective link to be posted w.r.t the below query

static const struct snd_kcontrol_new uda1345_snd_controls[] = {

SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1),

SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),

SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),

SOC_DAPM_PIN_SWITCH("GSM Line Out"),

SOC_DAPM_PIN_SWITCH("GSM Line In"),

SOC_DAPM_PIN_SWITCH("Headset Mic"),

SOC_DAPM_PIN_SWITCH("Handset Mic"),

};

/* Common DAPM widgets */

static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = {

#if 0

SND_SOC_DAPM_INPUT("VINL"),

SND_SOC_DAPM_INPUT("VINR"),

SND_SOC_DAPM_OUTPUT("VOUTL"),

SND_SOC_DAPM_OUTPUT("VOUTR"),

#endif

SND_SOC_DAPM_LINE("GSM Line Out", NULL),

SND_SOC_DAPM_LINE("GSM Line In", NULL),

SND_SOC_DAPM_MIC("Headset Mic", NULL),

SND_SOC_DAPM_MIC("Handset Mic", NULL),

};

static const struct snd_soc_dapm_route uda134x_dapm_routes[] = {

#if 0

{ "ADC", NULL, "VINL" },

{ "ADC", NULL, "VINR" },

{ "VOUTL", NULL, "DAC" },

{ "VOUTR", NULL, "DAC" },

#endif

{ "ADC", NULL, "VINL" },

{ "ADC", NULL, "VINR" },

{"VINL", NULL, "GSM Line In"},

{"VINR", NULL, "GSM Line In"},

{ "VOUTL", NULL, "DAC" },

{ "VOUTR", NULL, "DAC" },

{"GSM Line Out", NULL, "VOUTL"},

{"GSM Line Out", NULL, "VOUTR"},

/* Connections to Headset */

{"VINL", NULL, "Mic Bias"},

{"Mic Bias", NULL, "Headset Mic"},

/* Call Mic */

{"VINR", NULL, "Mic Bias"},

{"Mic Bias", NULL, "Handset Mic"},

};

Kindly do the needful as early as possible, as am stuck with this from many days

Awaiting for your replies

Many Many Thanks in advance again

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

could anybody please reply to my above queries w.r.t routing or loop back

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

Kinldy requesting any TI engineers, anybody please help me out w.r.t this thread as early as possible from two days there are no responses for this queries, Kindly do the needful as early as possible, because of which my release is getting blocked

SND_ALOOP is not working, could you please let me know what is the problem am unable to playback & am unable to record as well

am planning to use alsa loop back feature (ie., FULL DUPLEX), if so could you please suggest me how audio loop back can be tested as am not aware how to test this feature ?? I have configured the kernel as below in ti sdk 7

│ Symbol: SND_ALOOP [=y] │
│ Type : tristate │
│ Prompt: Generic loopback driver (PCM) │
│ Location: │
│ -> Device Drivers │
│ -> Sound card support (SOUND [=y]) │
│ -> Advanced Linux Sound Architecture (SND [=y]) │
│ -> Generic sound devices (SND_DRIVERS [=y]) │

mcasp0_pins: mcasp0_pins {
pinctrl-single,pins = <
0x1ac (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_ahclkx - MCLK */
0x190 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_aclkx - BCK */
0x194 (PIN_INPUT_PULLDOWN| MUX_MODE0) /* mcasp0_fsx - WS */
//0x198 (PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr0 - DATAO */
//0x1a8 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr1 - DATAI */

0x198 (PIN_INPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr0 - DATAO */
0x1a8 (PIN_OUTPUT_PULLDOWN | MUX_MODE0) /* mcasp0_axr1 - DATAI */

>;
};

&mcasp0 {
pinctrl-names = "default", "sleep";
pinctrl-0 = <&mcasp0_pins>;
pinctrl-1 = <&mcasp0_sleep_pins>;

status = "okay";

op-mode = <0>; /* MCASP_IIS_MODE */
tdm-slots = <2>;
/* 16 serializer */
serial-dir = < /* 0: INACTIVE, 1: TX, 2: RX */
2 1 0 0 /*AXR1 --> DATA 0/P FROM PROCESSOR, AXR0 --> DATA I/P PROCESSOR */
>;
tx-num-evt = <1>;
rx-num-evt = <1>;
};

# default Arago configuration
# default device
pcm.!default {
type plug
slave.pcm "loop"
}

# output device
pcm.loopout {
type dmix
ipc_key 328211
slave.pcm "hw:Loopback,0,0"
}

# input device
pcm.loopin {
type dsnoop
ipc_key 686592
slave.pcm "hw:Loopback,1,0"
}

# duplex plug device
pcm.loop {
type plug
slave {
pcm {
type asym
playback.pcm "loopout"
capture.pcm "loopin"
}
}
}

root@am335x-evm:/# aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 1: UDA1345TS [TI UDA1345TS], device 0: UDA134x uda134x-hifi-0 []
Subdevices: 1/1
Subdevice #0: subdevice #0
root@am335x-evm:/# arecord -f dat -D loop | aplay -f dat -D hw:0,0
Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

0 Ivan Matrakov over 9 years ago in reply to Srini

TI__Expert 4475 points

Srini,

This is the code to convert stereo to mono. Please see code and adapt to your case.

/* n1: number of samples */
static void stereo_to_mono(short *output, short *input, int n1)
{
short *p, *q;
int n = n1;

p = input;
q = output;
while (n >= 4) {
q[0] = (p[0] + p[1]) >> 1;
q[1] = (p[2] + p[3]) >> 1;
q[2] = (p[4] + p[5]) >> 1;
q[3] = (p[6] + p[7]) >> 1;
q += 4;
p += 8;
n -= 4;
}
while (n > 0) {
q[0] = (p[0] + p[1]) >> 1;
q++;
p += 2;
n--;
}
}

BR
Ivan

0 Srini over 9 years ago in reply to Ivan Matrakov

Genius 5250 points

Dear Ivan,

Thanks a lot for your quick responses & for your prompt replies,

As am new to ALSA, Could you please help me out how this can be achieved in ALSA, it is getting very difficult for me implement the above logic in ALSA,

Could you please let me know is there any configuration that can be done in asound.conf file?? inorder to convert stereo to mono

Basically I need to map VINL to VOUTL & VINR to VOUTR (please refer the block diagram of UDA1345TS datasheet page no .6, www.nxp.com/.../UDA1345TS.pdf) please let me know is this mappings can be done in asound.conf file

Am extremely sorry to push you hard once again

Kindly do the needful as early as possible, as am stuck with this from many days

Awaiting for your replies

Many Many Thanks in advance again

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

Dear Ivan,

I did the following changes to separate out the stereo output (ie., downmixing stereo to mono) ie., left & right channels to two different mono's in the etc/asound.conf

# default Arago configuration
pcm.card0 {
type hw
card 0
}

ctl.card0 {
type hw
card 0
}

pcm.monocardright {
slave.pcm card0
slave.channels 2
# type plug
type route
ttable {
# Copy both input channels to output channel 0 (Left).
0.0 1
1.1 1
# Send nothing to output channel 1 (Right).
0.1 0
1.1 0
}
}

ctl.monocardright {
type hw
card 0
}

pcm.monocardleft {
slave.pcm card0
slave.channels 2
# type plug
type route
ttable {
# Copy both input channels to output channel 0 (Left).
0.1 1
0.1 1
# Send nothing to output channel 1 (Right).
0.0 0
1.1 0
}
}

ctl.monocardleft {
type hw
card 0
}

pcm.!default monocardright
pcm.!default monocardleft

Now I have two mic inputs, ie., VINL & VINR , if possible could you please let me know how VINL can be mapped to monocardleft & VINR can be mapped to monocardright , I hope this changes can also be done in asound.conf only, could you please help me out in mapping this

sorry to push you hard once again

Kindly do the needful as early as possible, as am stuck with this from many days

Awaiting for your replies

Many Many Thanks in advance again

0 Srini over 9 years ago in reply to Srini

Genius 5250 points

Dear TI E2E Community

Now am able to map mic 1 input i.e, VINL to VOUTL & mic 2 input to VINR to VOUTR with the attached asound.conf & after this mapping & I tested one by one VINL to VOUTL path & then I tested VINR to VOUTR path I was able to get working independently

For my GSM two way call to work I used an application as shown in the attachment which uses both VINL to VOUTL & VINR to VOUTR (for two way call) it opens successfully for playback_handle_right & capture_handle_right but it throws device resource busy when it attempts to open

////LEFT CHANNEL
   if ((err = snd_pcm_open (&playback_handle_left, pldevice, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
   fprintf (stderr, "cannot open playback left audio device %s (%s)\n", pldevice, snd_strerror (err)); exit (1);
   }

Could you please let me know how this issue can be resolved , I understand since both the mono's is mapped to one sound card when it is trying to reopen again (ie., playback_handle_left is trying to open again since because playback_handle_right is already opened) it might be throwing device resource busy error

Could you please let me know how this issue ie., device resource busy can be resolved ??

4834.asound_conf.txt 6735.alsa_loopback_min.c

Kindly do the needful as early as possible

Awaiting for your replies,

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Audio Routing in ti sdk 7