[alsa-devel] i.MX6Q no audio with wm8962 Question
Adam Ford
aford173 at gmail.com
Tue Jul 17 21:40:04 CEST 2018
I have an iMX6Q with a wm8962 attached using AUDMUX4 and SSI2, but I
am having difficulty getting the sound to generate from the headphone
jack.
I have verified the power is working, and we're getting the 24MHz
clock signal from CK0 to the codec, and we've played with the
alsamixer to un-mute a variety of options, yet we're still unable to
get sound from the headphone port.
sound {
compatible = "fsl,imx-audio-wm8962";
model = "wm8962-audio";
ssi-controller = <&ssi2>;
audio-codec = <&codec>;
audio-routing =
"Headphone Jack", "HPOUTL",
"Headphone Jack", "HPOUTR",
"Ext Spk", "SPKOUTL",
"Ext Spk", "SPKOUTR",
"AMIC", "MICBIAS",
"IN3R", "AMIC";
mux-int-port = <2>;
mux-ext-port = <4>;
};
I beleive the mux-int-port points to ssi2 which should match the
ssi-controller = <&ssi2> option
I also beleive mux-ext-port = <4> should match the audmux 4 port.
&i2c1 {
pinctrl-names = "default";
pinctrl-0 = <&pinctrl_i2c1>;
clock-frequency = <400000>;
status = "okay";
codec: wm8962 at 1a {
compatible = "wlf,wm8962";
reg = <0x1a>;
clocks = <&clks IMX6QDL_CLK_CKO>;
clock-names = "xclk";
DCVDD-supply = <®_audio>;
DBVDD-supply = <®_audio>;
AVDD-supply = <®_audio>;
CPVDD-supply = <®_audio>;
MICVDD-supply = <®_audio>;
PLLVDD-supply = <®_audio>;
SPKVDD1-supply = <®_audio>;
SPKVDD2-supply = <®_audio>;
gpio-cfg = <
0x0000 /* 0:Default */
0x0000 /* 1:Default */
0x0013 /* 2:FN_DMICCLK */
0x0000 /* 3:Default */
0x8014 /* 4:FN_DMICCDAT */
0x0000 /* 5:Default */
>;
};
};
The snippet of my boot log are as follows:
[ 4.359592] snd-soc-dummy snd-soc-dummy: codec register snd-soc-dummy
[ 4.359615] snd-soc-dummy snd-soc-dummy: ASoC: dai register snd-soc-dummy #1
[ 4.359625] snd-soc-dummy snd-soc-dummy: ASoC: dynamically register
DAI snd-soc-dummy
[ 4.359637] snd-soc-dummy snd-soc-dummy: ASoC: Registered DAI
'snd-soc-dummy-dai'
[ 4.359744] snd-soc-dummy snd-soc-dummy: ASoC: Registered codec
'snd-soc-dummy'
[ 4.359753] snd-soc-dummy snd-soc-dummy: ASoC: platform register
snd-soc-dummy
[ 4.359775] snd-soc-dummy snd-soc-dummy: ASoC: Registered platform
'snd-soc-dummy'
[ 4.365610] wm8962 0-001a: customer id 0 revision F
[ 4.382144] wm8962 0-001a: codec register 0-001a
[ 4.382170] wm8962 0-001a: ASoC: dai register 0-001a #1
[ 4.382178] wm8962 0-001a: ASoC: dynamically register DAI 0-001a
[ 4.382190] wm8962 0-001a: ASoC: Registered DAI 'wm8962'
[ 4.382202] wm8962 0-001a: ASoC: Registered codec 'wm8962.0-001a'
[ 4.383977] fsl-asoc-card sound: ASoC: binding HiFi
[ 4.384007] fsl-asoc-card sound: ASoC: CPU DAI (null) not registered
This seems like an issue,but it looks like it potentially gets resolved later.
[ 4.386043] fsl-asrc 2034000.asrc: ASoC: dai register 2034000.asrc #1
[ 4.386092] fsl-asrc 2034000.asrc: ASoC: dynamically register DAI
2034000.asrc
[ 4.386108] fsl-asrc 2034000.asrc: ASoC: Registered DAI '2034000.asrc'
[ 4.386126] fsl-asrc 2034000.asrc: ASoC: platform register 2034000.asrc
[ 4.386144] fsl-asrc 2034000.asrc: ASoC: Registered platform '2034000.asrc'
[ 4.387439] fsl-ssi-dai 202c000.ssi: No cache defaults, reading back from HW
[ 4.388567] fsl-ssi-dai 202c000.ssi: ASoC: Registered platform '202c000.ssi'
[ 4.388589] fsl-ssi-dai 202c000.ssi: ASoC: dai register 202c000.ssi #1
[ 4.388597] fsl-ssi-dai 202c000.ssi: ASoC: dynamically register DAI
202c000.ssi
[ 4.388610] fsl-ssi-dai 202c000.ssi: ASoC: Registered DAI '202c000.ssi'
[ 4.393581] imx-wm8962 sound: ASoC: binding HiFi
[ 4.396233] input: WM8962 Beep Generator as
/devices/soc0/soc/2100000.aips-bus/21a0000.i2c/i2c-0/0-001a/input/input1
[ 4.442347] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late -2
[ 4.442380] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late -1
[ 4.442392] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 0
[ 4.442403] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 1
[ 4.442412] imx-wm8962 sound: ASoC: probe wm8962-audio dai link 0 late 2
[ 4.458756] imx-wm8962 sound: wm8962 <-> 202c000.ssi mapping ok
When trying to play audio through the headphone jack, I unmute the
headphones in alsamixer, but when I query the interface, it appears to
remain muted.
# amixer -c0 cget numid=24
numid=24,iface=MIXER,name='Headphone Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=127,step=0
: values=107,107
| dBscale-min=-121.00dB,step=1.00dB,mute=1
I have tried unmute with a simiar instruction without any luck.
I also noticed that when I run speaker-test -c2 -t wav, I get the
following message:
[ 1467.051679] wm8962 0-001a: Failed to read DCS status: -1
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 3840 to 17280
Period size range from 1920 to 1920
Using max buffer size 17280
Periods = 4
was set period_size = 1920
was set buffer_size = 17280
0 - Front Left
Time per period = 1.138101
0 - Front Left
^CTransfer failed: Bad address
# [ 1474.553646] wm8962 0-001a: Failed to read DSPCLK: -1
I am assuming that being unable to read the DCS -1 is a bad thing.
Unfortunatly, after this instruction. we can no longer use alsamixer
or the any alsa tool except for the speaker-test app.
When running the speaker test a second time I get even more errors:
Stream parameters are 48000Hz, S16_LE, 2 channels
WAV file(s)
[ 1700.659314] wm8962 0-001a: Unable to sync registers 0x0-0x3. -6
[ 1700.833594] wm8962 0-001a: Failed to read DSPCLK: -1
[ 1700.862662] wm8962 0-001a: ASoC: POST_PMU: HPOUTL PGA event failed: -6
[ 1700.869709] wm8962 0-001a: ASoC: POST_PMU: SPKOUTL PGA event failed: -6
[ 1700.879766] wm8962 0-001a: ASoC: POST_PMU: HPOUTR PGA event failed: -6
[ 1700.888565] wm8962 0-001a: ASoC: POST_PMU: SPKOUTR PGA event failed: -6
[ 1700.922471] wm8962 0-001a: Failed to read DCS status: -1
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 3840 to 17280
Period size range from 1920 to 1920
Using max buffer size 17280
Periods = 4
was set period_size = 1920
was set buffer_size = 17280
0 - Front Left
1 - Front Right
This worked one day, and for some reason, it seems to have stopped,
and its repeatable on multiple boards, so I think it's a software
configuration somewhere.
Might someone have any suggestions?
thank you,
adam
More information about the Alsa-devel
mailing list