[alsa-devel] No sound captured with SGTL5000 on i.MX6 in I²S master mode
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };
&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; }; };
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; }; };
&ssi1 { status = "okay"; codec-handle = <&codec>; fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */ };
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The kernel is 3.17-rc5.
Thanks for reading all this, and for any advice :). JM
Hi Jean-Michel,
On Thu, Sep 18, 2014 at 9:56 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
I am also trying to test fsl_ssi in master mode.
I am running linux-next 20140912 with the following changes to get SSI in master mode:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -532,6 +532,7 @@ };
&ssi2 { + fsl,mode = "i2s-master"; status = "okay"; };
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBS_CFS;
data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");
,but when I do aplay I do not get any sound.
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; };};
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; };};
&ssi1 { status = "okay"; codec-handle = <&codec>;
I think this codec-handle is not needed here.
fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */};
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The kernel is 3.17-rc5.
Thanks for reading all this, and for any advice :).
I am adding some more folks in Cc in case they have some suggestions.
Regards,
Fabio Estevam
Hi
On Thu, Sep 18, 2014 at 3:30 PM, Fabio Estevam festevam@gmail.com wrote:
Hi Jean-Michel,
On Thu, Sep 18, 2014 at 9:56 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
Yes, can you check if the direction of the mux-port is correct and you are really get data.
I am also trying to test fsl_ssi in master mode.
I am running linux-next 20140912 with the following changes to get SSI in master mode:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -532,6 +532,7 @@ };
&ssi2 {
fsl,mode = "i2s-master"; status = "okay";};
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
SND_SOC_DAIFMT_CBS_CFS; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");,but when I do aplay I do not get any sound.
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; };};
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; };};
&ssi1 { status = "okay"; codec-handle = <&codec>;
I think this codec-handle is not needed here.
fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */};
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The clock is not get in
Michael
The kernel is 3.17-rc5.
Thanks for reading all this, and for any advice :).
I am adding some more folks in Cc in case they have some suggestions.
Regards,
Fabio Estevam _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
2014-09-18 15:35 GMT+02:00 Michael Trimarchi michael@amarulasolutions.com:
Hi
On Thu, Sep 18, 2014 at 3:30 PM, Fabio Estevam festevam@gmail.com wrote:
Hi Jean-Michel,
On Thu, Sep 18, 2014 at 9:56 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
Yes, can you check if the direction of the mux-port is correct and you are really get data.
Mmh, can you explain in more details, I don't get it...
I am also trying to test fsl_ssi in master mode.
I am running linux-next 20140912 with the following changes to get SSI in master mode:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -532,6 +532,7 @@ };
&ssi2 {
fsl,mode = "i2s-master"; status = "okay";};
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
SND_SOC_DAIFMT_CBS_CFS; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");,but when I do aplay I do not get any sound.
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; };};
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; };};
&ssi1 { status = "okay"; codec-handle = <&codec>;
I think this codec-handle is not needed here.
fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */};
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The clock is not get in
And what can this mean from a driver/mux perspective ? Thanks, JM
Hi
On Thu, Sep 18, 2014 at 3:45 PM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
2014-09-18 15:35 GMT+02:00 Michael Trimarchi michael@amarulasolutions.com:
Hi
On Thu, Sep 18, 2014 at 3:30 PM, Fabio Estevam festevam@gmail.com wrote:
Hi Jean-Michel,
On Thu, Sep 18, 2014 at 9:56 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
Yes, can you check if the direction of the mux-port is correct and you are really get data.
Mmh, can you explain in more details, I don't get it...
ssi is the peripheral that is rooted from the pins outside (audmux) and audmux should configured according to clock direction and data direction etc. I don't remember now is done now in the latest kernel, because I have in the head too much architecture, but it should be possible to dump it from sysfs how is configured. So you need to check if it match
I am also trying to test fsl_ssi in master mode.
I am running linux-next 20140912 with the following changes to get SSI in master mode:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -532,6 +532,7 @@ };
&ssi2 {
fsl,mode = "i2s-master"; status = "okay";};
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
SND_SOC_DAIFMT_CBS_CFS; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");,but when I do aplay I do not get any sound.
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; };};
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; };};
&ssi1 { status = "okay"; codec-handle = <&codec>;
I think this codec-handle is not needed here.
fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */};
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The clock is not get in
And what can this mean from a driver/mux perspective ?
ssi <--- audmxux ---> pins (configured by pinctrl)
I forget that you see the clock out. BTW too late. What I'm trying to said is that even if pins are configured and routed they can be routed in the wrong direction. Right now you try to generate both clock from the cpu and you can see them but receive path can be configured in syncronous mode or it can work take the clock from another pin. What is your configuration?
Michael
Thanks, JM
2014-09-18 15:55 GMT+02:00 Michael Trimarchi michael@amarulasolutions.com:
Hi
On Thu, Sep 18, 2014 at 3:45 PM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
2014-09-18 15:35 GMT+02:00 Michael Trimarchi michael@amarulasolutions.com:
Hi
On Thu, Sep 18, 2014 at 3:30 PM, Fabio Estevam festevam@gmail.com wrote:
Hi Jean-Michel,
On Thu, Sep 18, 2014 at 9:56 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Hi,
Sorry if double posting, I have not received confirmation on subscription yesterday... So I try again :/.
I have a custom i.MX6 board, and have a sgtl5000 connected on it. I would like to have it used in master mode (it is used only for audio capture) but I get a "pcm_read:2031: read error: Input/output error" when using arecord. I have checked my signals and I get something on I²S which seems to be good. I launch : $> arecord -f cd -Dhw:0,0 -v -V stereo Receive Frame Sync is 44.1KHz Receive Clock Signal is ~1MHz (I can be more precise if needed) Data Receive is present, only when arecord is launched. It stops when there is no capture asked. Both clocks are still present even when no arecord is running.
This suggests, for me, that master mode is working. So, I don't know why I can't get any data to my arecord. Is this a left/right justified master mode ? Anything else ?
Yes, can you check if the direction of the mux-port is correct and you are really get data.
Mmh, can you explain in more details, I don't get it...
ssi is the peripheral that is rooted from the pins outside (audmux) and audmux should configured according to clock direction and data direction etc. I don't remember now is done now in the latest kernel, because I have in the head too much architecture, but it should be possible to dump it from sysfs how is configured. So you need to check if it match
OK, I will try to find out everything I can.
I am also trying to test fsl_ssi in master mode.
I am running linux-next 20140912 with the following changes to get SSI in master mode:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -532,6 +532,7 @@ };
&ssi2 {
fsl,mode = "i2s-master"; status = "okay";};
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
SND_SOC_DAIFMT_CBS_CFS; data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");,but when I do aplay I do not get any sound.
I am wondering if I forgot something in my DT or if I need to do something else :
sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "LINE_IN", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };&audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; };};
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0 >; };};
&ssi1 { status = "okay"; codec-handle = <&codec>;
I think this codec-handle is not needed here.
fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */};
Here is the dmesg part : [ 6.026803] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 6.129380] imx-sgtl5000 sound: snd_soc_register_card failed (-517) [ 6.248849] platform sound: Driver imx-sgtl5000 requests probe deferral [ 6.311643] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 6.393196] imx-sgtl5000 sound: sgtl5000 <-> 2028000.ssi mapping ok
And the user space commands : $> arecord -l **** List of CAPTURE Hardware Devices **** card 0: imx6qvbx3sgtl50 [imx6q-vbx3-sgtl5000], device 0: HiFi sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0
$> arecord -v -V stereo -fdat -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 85333 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
The clock is not get in
And what can this mean from a driver/mux perspective ?
ssi <--- audmxux ---> pins (configured by pinctrl)
I forget that you see the clock out. BTW too late. What I'm trying to said is that even if pins are configured and routed they can be routed in the wrong direction. Right now you try to generate both clock from the cpu and you can see them but receive path can be configured in syncronous mode or it can work take the clock from another pin. What is your configuration?
The master clock is provided by the CPU but I just found out something. I have a FPGA between sgtl5000 and i.MX6. It is here in order to route signals. And when imx-audmux is probed, it does not find the clock, because the FPGA is not configured yet. And as this driver is not returning EPROBE_DEFER, it does not try a little bit later (when the clock will really be here). I don't know if this is the explanation of my issue, but at least this is an interesting catch. I will try and get you informed.
JM
<snip>
I forget that you see the clock out. BTW too late. What I'm trying to said is that even if pins are configured and routed they can be routed in the wrong direction. Right now you try to generate both clock from the cpu and you can see them but receive path can be configured in syncronous mode or it can work take the clock from another pin. What is your configuration?
The master clock is provided by the CPU but I just found out something. I have a FPGA between sgtl5000 and i.MX6. It is here in order to route signals. And when imx-audmux is probed, it does not find the clock, because the FPGA is not configured yet. And as this driver is not returning EPROBE_DEFER, it does not try a little bit later (when the clock will really be here). I don't know if this is the explanation of my issue, but at least this is an interesting catch. I will try and get you informed.
Well, audmux is not getting clock, this is normal I think, the clock I was mentionning is the MCLK of SGTL5000 and this one is linked to FPGA, but found after all. I now had a further look into sound/soc/fsl/fsl_ssi.c and I see this :
sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; else if (!strcmp(sprop, "i2s-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; }
I may have missed something, but it seems that i2s-master is not parsed, and does not set dai_fmt ? Or maybe is it done later ? I tried : else if (!strcmp(sprop, "i2s-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; + else if (!strcmp(sprop, "i2s-master")) + ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFS; }
But this is not better :
arecord -v -V stereo -f cd -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Hardware PCM card 0 'imx6q-vbx3-sgtl5000' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16384 period_size : 4096 period_time : 92879 tstamp_mode : NONE period_step : 1 avail_min : 4096 period_event : 0 start_threshold : 1 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 appl_ptr : 0 hw_ptr : 0 arecord: pcm_read:2031: read error: Input/output error
JM
On Thu, Sep 18, 2014 at 06:09:50PM +0200, Jean-Michel Hautbois wrote:
Well, audmux is not getting clock, this is normal I think, the clock I was mentionning is the MCLK of SGTL5000 and this one is linked to FPGA, but found after all. I now had a further look into sound/soc/fsl/fsl_ssi.c and I see this :
sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; else if (!strcmp(sprop, "i2s-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; }I may have missed something, but it seems that i2s-master is not parsed, and does not set dai_fmt ?
Actually this is kind of obsolete property because initially the SSI driver only supported i2s-slave. We've put an i2s-slave check here is to limit the driver by excluding other modes.
But now, the driver has the capability to derive clock from CCM and output it for external CODEC. So this one could be dropped in fact rather than adding a new i2s-master case IMO.
arecord -v -V stereo -f cd -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
arecord: pcm_read:2031: read error: Input/output error
The problem here should be the AUDMUX configuration issue. The imx- sgtl5000.c driver only supports CODEC in master mode. So if you try to switch the CODEC slave mode, you shall also change not only the CBM_CFM to CBS_CFS but also swap the ext_port and int_port of AUDMUX (a little confusing approach here as the configuration of AUDMUX is routing the data and clocks from a source port to a destination port while each of side, external or internal, might be a source port -- When using CBM_CFM, the source port should be external port; while using CBS_CFS, the source port should be the internal port.)
There may be another topic, however, actually the fsl-asoc-card driver does handle the master/slave mode supports. So if you are trying to add the CODEC slave mode support into imx-sgtl5000. I suggest you to try the fsl-asoc-card instead. Its DT binding's totally compatible with imx-sgtl5000's. What you need to do is just enable it (and disable imx-sgtl500) in the menuconfig or add these enable/disable into imx_v6_v7_defconfig.
Nicolin
Hi
On Thu, Sep 18, 2014 at 10:05:24AM -0700, Nicolin Chen wrote:
On Thu, Sep 18, 2014 at 06:09:50PM +0200, Jean-Michel Hautbois wrote:
Well, audmux is not getting clock, this is normal I think, the clock I was mentionning is the MCLK of SGTL5000 and this one is linked to FPGA, but found after all. I now had a further look into sound/soc/fsl/fsl_ssi.c and I see this :
sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { if (!strcmp(sprop, "ac97-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_AC97; else if (!strcmp(sprop, "i2s-slave")) ssi_private->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; }I may have missed something, but it seems that i2s-master is not parsed, and does not set dai_fmt ?
Actually this is kind of obsolete property because initially the SSI driver only supported i2s-slave. We've put an i2s-slave check here is to limit the driver by excluding other modes.
But now, the driver has the capability to derive clock from CCM and output it for external CODEC. So this one could be dropped in fact rather than adding a new i2s-master case IMO.
arecord -v -V stereo -f cd -D hw:0,0 somefile.wav Recording WAVE 'somefile.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
arecord: pcm_read:2031: read error: Input/output error
The problem here should be the AUDMUX configuration issue. The imx- sgtl5000.c driver only supports CODEC in master mode. So if you try to switch the CODEC slave mode, you shall also change not only the CBM_CFM to CBS_CFS but also swap the ext_port and int_port of AUDMUX (a little confusing approach here as the configuration of AUDMUX is routing the data and clocks from a source port to a destination port while each of side, external or internal, might be a source port -- When using CBM_CFM, the source port should be external port; while using CBS_CFS, the source port should be the internal port.)
There may be another topic, however, actually the fsl-asoc-card driver does handle the master/slave mode supports. So if you are trying to add the CODEC slave mode support into imx-sgtl5000. I suggest you to try the fsl-asoc-card instead. Its DT binding's totally compatible with imx-sgtl5000's. What you need to do is just enable it (and disable imx-sgtl500) in the menuconfig or add these enable/disable into imx_v6_v7_defconfig.
bug was reported by Jean-Michel. I don't know if it works because I can not test but seems ok. Do we need this _set_dai_fmt in probe function?
code is a bit broken just because panic on probe path if pdev is null [...] /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ if (!of_get_property(np, "codec-handle", NULL)) goto done; [...] ssi_private->pdev = platform_device_register_data(&pdev->dev, name, 0, NULL, 0); [...] done: if (ssi_private->dai_fmt) _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..f63bc02 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -758,7 +758,8 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + pr_err("baudclk is missing which is necessary" + " for master mode\n"); return -EINVAL; }
Michael
Nicolin
On Thu, Sep 18, 2014 at 07:26:11PM +0200, Michael Trimarchi wrote:
bug was reported by Jean-Michel. I don't know if it works because I can not test but seems ok. Do we need this _set_dai_fmt in probe function?
I think yes. The AC97 part might need this _set_dai_fmt call in the probe().
code is a bit broken just because panic on probe path if pdev is null [...] /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ if (!of_get_property(np, "codec-handle", NULL)) goto done; [...] ssi_private->pdev = platform_device_register_data(&pdev->dev, name, 0, NULL, 0); [...] done: if (ssi_private->dai_fmt) _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..f63bc02 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -758,7 +758,8 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
pr_err("baudclk is missing which is necessary"" for master mode\n");
Ah, this is a problem. We should not use ssi_private->pdev->dev here. I think it'll be better if adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt().
Nicolin
Hi
On Thu, Sep 18, 2014 at 10:45:27AM -0700, Nicolin Chen wrote:
On Thu, Sep 18, 2014 at 07:26:11PM +0200, Michael Trimarchi wrote:
bug was reported by Jean-Michel. I don't know if it works because I can not test but seems ok. Do we need this _set_dai_fmt in probe function?
I think yes. The AC97 part might need this _set_dai_fmt call in the probe().
code is a bit broken just because panic on probe path if pdev is null [...] /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require * SSI driver to trigger machine driver's probe. */ if (!of_get_property(np, "codec-handle", NULL)) goto done; [...] ssi_private->pdev = platform_device_register_data(&pdev->dev, name, 0, NULL, 0); [...] done: if (ssi_private->dai_fmt) _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..f63bc02 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -758,7 +758,8 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
pr_err("baudclk is missing which is necessary"" for master mode\n");Ah, this is a problem. We should not use ssi_private->pdev->dev here. I think it'll be better if adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt().
Somenthing like this?
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..de6ab06 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; }
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; }
@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
- return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); }
/** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt);
return 0;
Nicolin
On Thu, Sep 18, 2014 at 08:03:31PM +0200, Michael Trimarchi wrote:
Ah, this is a problem. We should not use ssi_private->pdev->dev here. I think it'll be better if adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt().
Somenthing like this?
Yes, would you like to send a separate patch? I think other people and maintainers'd like to take a look at it and decide if it's a good shoot. I'll also take a further look at it after I finish my laundry :)
Thank you Nicolin
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..de6ab06 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; }
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
unsigned int fmt)+static int _fsl_ssi_set_dai_fmt(struct device *dev,
struct fsl_ssi_private *ssi_private,unsigned int fmt){ struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
dev_err(dev, "baudclk is missing which is necessary for master mode\n");@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
- return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
- return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);
}
/** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done: if (ssi_private->dai_fmt)
_fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
_fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,ssi_private->dai_fmt);return 0;
Nicolin
-- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL | | [`as] http://www.amarulasolutions.com |
Hi Fabio
On Thu, Sep 18, 2014 at 8:13 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
On Thu, Sep 18, 2014 at 08:03:31PM +0200, Michael Trimarchi wrote:
Ah, this is a problem. We should not use ssi_private->pdev->dev here. I think it'll be better if adding a new parameter of *dev pointer to this _set_dai_fmt() -- passing pdev->dev in probe() and cpu_dai->dev in fsl_ssi_set_dai_fmt().
Somenthing like this?
Yes, would you like to send a separate patch? I think other people and maintainers'd like to take a look at it and decide if it's a good shoot. I'll also take a further look at it after I finish my laundry :)
Yes sure
Michael
Thank you Nicolin
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb577..de6ab06 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; }
-static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private,
unsigned int fmt)+static int _fsl_ssi_set_dai_fmt(struct device *dev,
struct fsl_ssi_private *ssi_private,unsigned int fmt){ struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt;
if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) {
dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n");
dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; }@@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai);
return _fsl_ssi_set_dai_fmt(ssi_private, fmt);
return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt);}
/** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev)
done: if (ssi_private->dai_fmt)
_fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt);
_fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private,ssi_private->dai_fmt); return 0;Nicolin
-- | Michael Nazzareno Trimarchi Amarula Solutions BV | | COO - Founder Cruquiuskade 47 | | +31(0)851119172 Amsterdam 1018 AM NL | | [`as] http://www.amarulasolutions.com |
Hi Nicolin,
On Thu, Sep 18, 2014 at 2:05 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
The problem here should be the AUDMUX configuration issue. The imx- sgtl5000.c driver only supports CODEC in master mode. So if you try to switch the CODEC slave mode, you shall also change not only the CBM_CFM to CBS_CFS but also swap the ext_port and int_port of AUDMUX (a little confusing approach here as the configuration of AUDMUX is routing the data and clocks from a source port to a destination port while each of side, external or internal, might be a source port -- When using CBM_CFM, the source port should be external port; while using CBS_CFS, the source port should be the internal port.)
Thanks for the hint on audmux swap! I can get ssi in master mode to work correctly with the change below:
--- a/arch/arm/boot/dts/imx6qdl-sabresd.dtsi +++ b/arch/arm/boot/dts/imx6qdl-sabresd.dtsi @@ -111,8 +111,8 @@ "IN3R", "MICBIAS", "DMIC", "MICBIAS", "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; + mux-int-port = <3>; + mux-ext-port = <2>; };
backlight { diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3a3d17c..09ea6a5 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -247,7 +247,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->dai.platform_of_node = ssi_np; data->dai.ops = &imx_hifi_ops; data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBS_CFS;
data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model");
Hi Fabio,
On Thu, Sep 18, 2014 at 02:28:29PM -0300, Fabio Estevam wrote:
Thanks for the hint on audmux swap! I can get ssi in master mode to work correctly with the change below:
mux-int-port = <2>;mux-ext-port = <3>;
mux-int-port = <3>;mux-ext-port = <2>;
Actually this looks a bit tricky as the internal and external ports are not physically swapped :)
So that's why I put the swapping code in the fsl-asoc-card based on CBx_CFx configurations.
Thanks Nicolin
Hi Nicolin,
On Thu, Sep 18, 2014 at 2:49 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
Hi Fabio,
On Thu, Sep 18, 2014 at 02:28:29PM -0300, Fabio Estevam wrote:
Thanks for the hint on audmux swap! I can get ssi in master mode to work correctly with the change below:
mux-int-port = <2>;mux-ext-port = <3>;
mux-int-port = <3>;mux-ext-port = <2>;Actually this looks a bit tricky as the internal and external ports are not physically swapped :)
Yes, this change is just for my own testing/debugging.
So that's why I put the swapping code in the fsl-asoc-card based on CBx_CFx configurations.
Correct, fsl-asoc-card is a nice approach. Good job!
On Thu, Sep 18, 2014 at 02:58:06PM -0300, Fabio Estevam wrote:
So that's why I put the swapping code in the fsl-asoc-card based on CBx_CFx configurations.
Correct, fsl-asoc-card is a nice approach. Good job!
I plan to replace imx-sgtl5000 and imx-wm8962 with it as long as people are happy about the new driver. Would you like to help me testing it on sgtl5000 side when you have time? Because I only have a board with WM8962.
Thank you Nicolin
On Thu, Sep 18, 2014 at 3:07 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
On Thu, Sep 18, 2014 at 02:58:06PM -0300, Fabio Estevam wrote:
So that's why I put the swapping code in the fsl-asoc-card based on CBx_CFx configurations.
Correct, fsl-asoc-card is a nice approach. Good job!
I plan to replace imx-sgtl5000 and imx-wm8962 with it as long as people are happy about the new driver. Would you like to help me testing it on sgtl5000 side when you have time? Because I only have a board with WM8962.
Yes, I can test it on a imx53-qsb, which has sgtl5000. Just Cc me when you have the patch available.
Regards,
Fabio Estevam
On Thu, Sep 18, 2014 at 03:14:08PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:07 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
On Thu, Sep 18, 2014 at 02:58:06PM -0300, Fabio Estevam wrote:
So that's why I put the swapping code in the fsl-asoc-card based on CBx_CFx configurations.
Correct, fsl-asoc-card is a nice approach. Good job!
I plan to replace imx-sgtl5000 and imx-wm8962 with it as long as people are happy about the new driver. Would you like to help me testing it on sgtl5000 side when you have time? Because I only have a board with WM8962.
Yes, I can test it on a imx53-qsb, which has sgtl5000. Just Cc me when you have the patch available.
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Thank you Nicolin
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
On Thu, Sep 18, 2014 at 08:35:31PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
It's just a combinational driver and also made to be compatible with old DTs. Unless we refine the DT binding like simple-card, this hard- code is inevitable. As long as this dai_fmt has been defined, at least AUDMUX part will be automatically configured.
I made this driver was initially to support ASRC for imx-cs42888. But merging the code from imx-sgtl5000 and imx-wm8962 allows us to omit a branch of duplicated code. And that's it :)
Nicolin
Hi,
2014-09-19 1:50 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Thu, Sep 18, 2014 at 08:35:31PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;It's just a combinational driver and also made to be compatible with old DTs. Unless we refine the DT binding like simple-card, this hard- code is inevitable. As long as this dai_fmt has been defined, at least AUDMUX part will be automatically configured.
I made this driver was initially to support ASRC for imx-cs42888. But merging the code from imx-sgtl5000 and imx-wm8962 allows us to omit a branch of duplicated code. And that's it :)
Well, I may have missed something, I don't have fsl-asoc-card in my linux tree (vanilla). Where can I find it ? I can test a SGTL5000 master mode on i.MX6Q.
Thanks, JM
On Fri, Sep 19, 2014 at 09:42:22AM +0200, Jean-Michel Hautbois wrote:
Hi,
2014-09-19 1:50 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Thu, Sep 18, 2014 at 08:35:31PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;It's just a combinational driver and also made to be compatible with old DTs. Unless we refine the DT binding like simple-card, this hard- code is inevitable. As long as this dai_fmt has been defined, at least AUDMUX part will be automatically configured.
I made this driver was initially to support ASRC for imx-cs42888. But merging the code from imx-sgtl5000 and imx-wm8962 allows us to omit a branch of duplicated code. And that's it :)
Well, I may have missed something, I don't have fsl-asoc-card in my linux tree (vanilla). Where can I find it ? I can test a SGTL5000 master mode on i.MX6Q.
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
for-next branch.
Nicolin
2014-09-19 9:52 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Fri, Sep 19, 2014 at 09:42:22AM +0200, Jean-Michel Hautbois wrote:
Hi,
2014-09-19 1:50 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Thu, Sep 18, 2014 at 08:35:31PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;It's just a combinational driver and also made to be compatible with old DTs. Unless we refine the DT binding like simple-card, this hard- code is inevitable. As long as this dai_fmt has been defined, at least AUDMUX part will be automatically configured.
I made this driver was initially to support ASRC for imx-cs42888. But merging the code from imx-sgtl5000 and imx-wm8962 allows us to omit a branch of duplicated code. And that's it :)
Well, I may have missed something, I don't have fsl-asoc-card in my linux tree (vanilla). Where can I find it ? I can test a SGTL5000 master mode on i.MX6Q.
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
for-next branch.
Nicolin
Thanks, now I need to modify my DT in order to have fsl-asoc-card loaded :). JM
Hi,
2014-09-19 11:18 GMT+02:00 Jean-Michel Hautbois jean-michel.hautbois@vodalys.com:
2014-09-19 9:52 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Fri, Sep 19, 2014 at 09:42:22AM +0200, Jean-Michel Hautbois wrote:
Hi,
2014-09-19 1:50 GMT+02:00 Nicolin Chen nicoleotsuka@gmail.com:
On Thu, Sep 18, 2014 at 08:35:31PM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:39 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
It actually is merged into broonie/for-next and even linux-next I think. I'll later send a patch to Shawn's tree for imx_v6_v7_defconfig changes after next merge window's done.
Ok, I see it now, but master/slave configuration is still hardcoded there:
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;It's just a combinational driver and also made to be compatible with old DTs. Unless we refine the DT binding like simple-card, this hard- code is inevitable. As long as this dai_fmt has been defined, at least AUDMUX part will be automatically configured.
I made this driver was initially to support ASRC for imx-cs42888. But merging the code from imx-sgtl5000 and imx-wm8962 allows us to omit a branch of duplicated code. And that's it :)
Well, I may have missed something, I don't have fsl-asoc-card in my linux tree (vanilla). Where can I find it ? I can test a SGTL5000 master mode on i.MX6Q.
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
for-next branch.
Nicolin
Thanks, now I need to modify my DT in order to have fsl-asoc-card loaded :). JM
FYI, This is working ! I can capture my sound from line in with this : sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "A1N1L", "Line In Jack", "A1N1R", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; };
Now, I will add mic support, and I still need to modprobe manually, I need to know why...
Thanks, JM
Hi
I use fsl-asoc-card now.
This is working when compiling as a module, and when doing a modprobe manually, but when I compile it as built-in, with these : CONFIG_SND_SOC_IMX_SGTL5000 is not set CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_SGTL5000=y
I get the following error : [ 13.561618] fsl-asoc-card sound: ASoC: CODEC DAI sgtl5000 not registered [ 13.568503] fsl-asoc-card sound: snd_soc_register_card failed (-517) [ 13.574992] platform sound: Driver fsl-asoc-card requests probe deferral
It loops on it during boot, and never registers...
If I have : CONFIG_SND_SOC_IMX_SGTL5000 is not set CONFIG_SND_SOC_FSL_ASOC_CARD=m CONFIG_SND_SOC_SGTL5000=m
And after login : $> lsmod Module Size Used by snd_soc_sgtl5000 14074 0 $> modprobe snd-soc-fsl-asoc-card [ 65.300995] fsl-asoc-card sound: ASoC: CODEC DAI sgtl5000 not registered [ 65.307963] fsl-asoc-card sound: snd_soc_register_card failed (-517) [ 65.314423] platform sound: Driver fsl-asoc-card requests probe deferral $> rmmod snd-soc-sgtl5000 $> modprobe snd-soc-sgtl5000 [ 138.936753] sgtl5000 1-000a: sgtl5000 revision 0x11 [ 138.987974] 1-000a: 1200 mV normal [ 139.000923] sgtl5000 1-000a: Using internal LDO instead of VDDD [ 139.032169] fsl-asoc-card sound: sgtl5000 <-> 2028000.ssi mapping ok [ 139.038624] fsl-asoc-card sound: ASoC: no source widget found for ASRC-Playback [ 139.045984] fsl-asoc-card sound: ASoC: Failed to add route ASRC-Playback -> direct -> CPU-Playback [ 139.055187] fsl-asoc-card sound: ASoC: no sink widget found for ASRC-Capture [ 139.062253] fsl-asoc-card sound: ASoC: Failed to add route CPU-Capture -> direct -> ASRC-Capture
The routes are not ok, but I think I read something about it another thread, but it works. It seems that if snd-soc-sgtl5000 is already registered then it does not work. It is probably due to my DT ?
Here is an extract of the DT : / { sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx6q-vbx3-sgtl5000"; ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "A1N1L", "Line In Jack", "A1N1R", "Line In Jack"; mux-int-port = <1>; mux-ext-port = <5>; }; };
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>; }; };
Thanks, JM
On Thu, Oct 9, 2014 at 11:02 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Here is an extract of the DT : / { sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000";
I am confused here: you say you use the simple-card, but your DTS still shows imx-audio-sgtl5000.
arch/arm/boot/dts/vf610-twr.dts shows an example of the usage of simple card with sgtl5000.
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>;
This 'sysclk' property is unneeded.
2014-10-09 16:18 GMT+02:00 Fabio Estevam festevam@gmail.com:
On Thu, Oct 9, 2014 at 11:02 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Here is an extract of the DT : / { sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000";
I am confused here: you say you use the simple-card, but your DTS still shows imx-audio-sgtl5000.
I mentioned fsl-asoc-card, not simple card ?
arch/arm/boot/dts/vf610-twr.dts shows an example of the usage of simple card with sgtl5000.
&i2c2 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay";
codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&clks 201>; VDDA-supply = <&vgen6_reg>; VDDIO-supply = <&vgen6_reg>; sysclk = <24000000>;This 'sysclk' property is unneeded.
Right, I removed it thanks. JM
On Thu, Oct 9, 2014 at 11:26 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
2014-10-09 16:18 GMT+02:00 Fabio Estevam festevam@gmail.com:
On Thu, Oct 9, 2014 at 11:02 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Here is an extract of the DT : / { sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000";
I am confused here: you say you use the simple-card, but your DTS still shows imx-audio-sgtl5000.
I mentioned fsl-asoc-card, not simple card ?
Yes, right, but for using fsl-asoc-card you should reference 'audio-cpu' instead of 'ssi-controller'.
That's what Documentation/devicetree/bindings/sound/fsl-asoc-card.txt mentions.
2014-10-09 17:06 GMT+02:00 Fabio Estevam festevam@gmail.com:
On Thu, Oct 9, 2014 at 11:26 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
2014-10-09 16:18 GMT+02:00 Fabio Estevam festevam@gmail.com:
On Thu, Oct 9, 2014 at 11:02 AM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Here is an extract of the DT : / { sound { compatible = "fsl,imx6q-vbx3-sgtl5000", "fsl,imx-audio-sgtl5000";
I am confused here: you say you use the simple-card, but your DTS still shows imx-audio-sgtl5000.
I mentioned fsl-asoc-card, not simple card ?
Yes, right, but for using fsl-asoc-card you should reference 'audio-cpu' instead of 'ssi-controller'.
That's what Documentation/devicetree/bindings/sound/fsl-asoc-card.txt mentions.
Ouch, nice catch :).
Here is what I understand, in my cas, sgtl5000 has its clock routed through a FPGA. If the FPGA has not yet be probed, then, the module fails in loading : [ 4.985762] sgtl5000: probe of 1-000a failed with error -5
This is why after removing it and probing it again manually, it works.
This module should be able to defer probing too, is it going to be done by someone or do I try to send some patch for this ?
BTW, I don't have sound on the microphone input, and I think there is no way to select micbias voltage level neither from the source code nor the DT ? It could be interetsing to have something like sgtl5000-micbias-vg which would give control on the voltage bias parameter ? And maybe also something for bias resistor too ?
I can try to patch something. Thanks, JM
On Thu, Oct 9, 2014 at 12:14 PM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Ouch, nice catch :).
Here is what I understand, in my cas, sgtl5000 has its clock routed through a FPGA. If the FPGA has not yet be probed, then, the module fails in loading : [ 4.985762] sgtl5000: probe of 1-000a failed with error -5
This is why after removing it and probing it again manually, it works.
This module should be able to defer probing too, is it going to be done by someone or do I try to send some patch for this ?
We do defer probing already:
sgtl5000->mclk = devm_clk_get(&client->dev, NULL); if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); /* Defer the probe to see if the clk will be provided later */ if (ret == -ENOENT) return -EPROBE_DEFER; return ret; }
BTW, I don't have sound on the microphone input, and I think there is no way to select micbias voltage level neither from the source code nor the DT ? It could be interetsing to have something like sgtl5000-micbias-vg which would give control on the voltage bias parameter ? And maybe also something for bias resistor too ?
I can try to patch something.
That would be great, thanks.
2014-10-09 17:19 GMT+02:00 Fabio Estevam festevam@gmail.com:
On Thu, Oct 9, 2014 at 12:14 PM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Ouch, nice catch :).
Here is what I understand, in my cas, sgtl5000 has its clock routed through a FPGA. If the FPGA has not yet be probed, then, the module fails in loading : [ 4.985762] sgtl5000: probe of 1-000a failed with error -5
This is why after removing it and probing it again manually, it works.
This module should be able to defer probing too, is it going to be done by someone or do I try to send some patch for this ?
We do defer probing already:
sgtl5000->mclk = devm_clk_get(&client->dev, NULL); if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); /* Defer the probe to see if the clk will be provided later */ if (ret == -ENOENT) return -EPROBE_DEFER; return ret; }
Yes, sorry about that, missed it.
BTW, I don't have sound on the microphone input, and I think there is no way to select micbias voltage level neither from the source code nor the DT ? It could be interetsing to have something like sgtl5000-micbias-vg which would give control on the voltage bias parameter ? And maybe also something for bias resistor too ?
I can try to patch something.
That would be great, thanks.
Do I need to keep actual defaults if the resistor or voltage are not specified in DT ? Or should I set it to off (which is probably more logical) ? And one or two patches (one resistor and one voltage patch, or both in the same) ?
JM
On Fri, Oct 10, 2014 at 12:49 PM, Jean-Michel Hautbois jean-michel.hautbois@vodalys.com wrote:
Do I need to keep actual defaults if the resistor or voltage are not specified in DT ?
Better to keep the behaviour unchanged, so that old dtb's can work exactly in the same way.
Or should I set it to off (which is probably more logical) ? And one or two patches (one resistor and one voltage patch, or both in the same) ?
Splitting in 2 patches is easier for reviewing.
On Thu, Sep 18, 2014 at 3:07 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
I plan to replace imx-sgtl5000 and imx-wm8962 with it as long as people are happy about the new driver. Would you like to help me testing it on sgtl5000 side when you have time? Because I only have a board with WM8962.
Just tested fsl-asoc-card with sgtl5000 and it worked fine on my imx53-qsb.
These messages are a bit annoying though:
[ 1.786201] fsl-asoc-card sound: ASoC: no source widget found for ASRC-Playback [ 1.796065] fsl-asoc-card sound: ASoC: Failed to add route ASRC-Playback -> direct -> CPU-Playback [ 1.807571] fsl-asoc-card sound: ASoC: no sink widget found for ASRC-Capture [ 1.817112] fsl-asoc-card sound: ASoC: Failed to add route CPU-Capture -> direct -> ASRC-Capture
On Mon, Sep 22, 2014 at 12:26:21AM -0300, Fabio Estevam wrote:
On Thu, Sep 18, 2014 at 3:07 PM, Nicolin Chen nicoleotsuka@gmail.com wrote:
I plan to replace imx-sgtl5000 and imx-wm8962 with it as long as people are happy about the new driver. Would you like to help me testing it on sgtl5000 side when you have time? Because I only have a board with WM8962.
Just tested fsl-asoc-card with sgtl5000 and it worked fine on my imx53-qsb.
Thank you! Then I think we can enable it from the next release version.
These messages are a bit annoying though:
[ 1.786201] fsl-asoc-card sound: ASoC: no source widget found for ASRC-Playback [ 1.796065] fsl-asoc-card sound: ASoC: Failed to add route ASRC-Playback -> direct -> CPU-Playback [ 1.807571] fsl-asoc-card sound: ASoC: no sink widget found for ASRC-Capture [ 1.817112] fsl-asoc-card sound: ASoC: Failed to add route CPU-Capture -> direct -> ASRC-Capture
I'll try to find a way to circumvent the ASRC route for non-ASRC links.
Thanks, Nicolin
On Thu, Sep 18, 2014 at 10:30:18AM -0300, Fabio Estevam wrote:
&iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_hog>;
imx6qdl-vbx3 { pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDLPAD_DISP0_DAT13__AUD5_RXFS 0x130b0 MX6QDL_PAD_DISP0_DAT14__AUD5_RXC 0x130b0 MX6QDL_PAD_DISP0_DAT19__AUD5_RXD 0x130b0
The other issue over here is you are using asynchronous mode of SSI, while imx-sgtl5000 is using synchronous configurations for AUDMUX.
You can try it with fsl-asoc-card because it uses asynchronous mode of AUDMUX, although I've not tested it with SSI in its asynchronous mode as I only have a synchronous (SSI) environment.
&ssi1 { status = "okay"; codec-handle = <&codec>; fsl,mode = "i2s-master"; fsl,ssi-asynchronous; /* Not sure about it, but does not change anything */
The driver cares about it. If you look at the symmetric part carefully, you would find some register configurations of SSI does depends on the this property via symmetric check. For example:
if (ssi_private->cpu_dai_drv.symmetric_rates) { /* Need to clear RXDIR when using SYNC mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }
Nicolin
participants (4)
-
Fabio Estevam -
Jean-Michel Hautbois -
Michael Trimarchi -
Nicolin Chen