[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0.
Signed-off-by: Guneshwor Singh guneshwor.o.singh@intel.com --- sound/soc/intel/boards/Kconfig | 11 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 232 +++++++++++++++++++++++++++++++++++++ 3 files changed, 245 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c
diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..822b9143dcb1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,17 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N".
+config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on MFD_INTEL_LPSS && ACPI && I2C + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE
endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..2b49cb6aa84a 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,6 +21,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..04986c015f0f --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,232 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2016-18 Intel Corporation. + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack cnl_headset; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_component *component = codec_dai->component; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_component_set_jack(component, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = KBUILD_MODNAME, + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = cnl_rt274_probe, +}; + +module_platform_driver(cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh guneshwor.o.singh@intel.com"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274");
+static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
+{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai =
snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI);
- int ret, ratio = 100;
- if (!codec_dai)
return -EINVAL;
- /* Codec needs clock for Jack detection and button press */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2,
CNL_FREQ_OUT, SND_SOC_CLOCK_IN);
- if (ret < 0) {
dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret);
return ret;
- }
- if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio);
if (ret) {
dev_err(codec_dai->dev,
"set bclk ratio failed: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK,
CNL_BE_FIXUP_RATE * ratio,
CNL_FREQ_OUT);
if (ret) {
dev_err(codec_dai->dev,
"enable PLL2 failed: %d\n", ret);
return ret;
}
- }
- return 0;
+}
it's not clear to me why you need a clock control? You are not changing anything that really depends on DAPM events, to e.g. take the MCLK down and use a local clock, so could this be moved to hw_params?
+static const struct snd_soc_dapm_route cnl_map[] = {
- {"Headphone Jack", NULL, "HPO Pin"},
- {"MIC", NULL, "Mic Jack"},
- {"DMic", NULL, "SoC DMIC"},
- {"DMIC01 Rx", NULL, "Capture"},
- {"dmic01_hifi", NULL, "DMIC01 Rx"},
- {"AIF1 Playback", NULL, "ssp0 Tx"},
- {"ssp0 Tx", NULL, "codec1_out"},
- {"ssp0 Tx", NULL, "codec0_out"},
I get the routes to connect firmware widgets to codec ones, but why do we need SSP0 TX-> codec1_out? shouldn't this be part of the topology?
- {"ssp0 Rx", NULL, "AIF1 Capture"},
- {"codec0_in", NULL, "ssp0 Rx"},
- {"Headphone Jack", NULL, "Platform Clock"},
- {"Mic Jack", NULL, "Platform Clock"},
+};
+static struct snd_soc_jack_pin cnl_headset_pins[] = {
- {
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
- },
- {
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
- },
+};
+static struct snd_soc_jack cnl_headset;
+static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{
- struct snd_soc_card *card = runtime->card;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
- struct snd_soc_component *component = codec_dai->component;
- int ret;
- ret = snd_soc_card_jack_new(runtime->card, "Headset",
SND_JACK_HEADSET, &cnl_headset,
cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins));
- if (ret)
return ret;
- ret = snd_soc_component_set_jack(component, &cnl_headset, NULL);
- if (ret)
return ret;
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24);
what are the 4 slots used for?
Hi Pierre,
On Thu, Mar 15, 2018 at 07:23:05AM -0500, Pierre-Louis Bossart wrote:
+static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
+{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai =
snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI);
- int ret, ratio = 100;
- if (!codec_dai)
return -EINVAL;
- /* Codec needs clock for Jack detection and button press */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2,
CNL_FREQ_OUT, SND_SOC_CLOCK_IN);
- if (ret < 0) {
dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret);
return ret;
- }
- if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio);
if (ret) {
dev_err(codec_dai->dev,
"set bclk ratio failed: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK,
CNL_BE_FIXUP_RATE * ratio,
CNL_FREQ_OUT);
if (ret) {
dev_err(codec_dai->dev,
"enable PLL2 failed: %d\n", ret);
return ret;
}
- }
- return 0;
+}
it's not clear to me why you need a clock control? You are not changing anything that really depends on DAPM events, to e.g. take the MCLK down and use a local clock, so could this be moved to hw_params?
Yes, we can do in hw_params too. When we implemented it we thought during simultaneous playback and capture, hw_params will be called twice but clock clock event will be called just once.
It does not harm to call twice as done in other machine drivers. Do you suggest to move it to hw_params?
+static const struct snd_soc_dapm_route cnl_map[] = {
- {"Headphone Jack", NULL, "HPO Pin"},
- {"MIC", NULL, "Mic Jack"},
- {"DMic", NULL, "SoC DMIC"},
- {"DMIC01 Rx", NULL, "Capture"},
- {"dmic01_hifi", NULL, "DMIC01 Rx"},
- {"AIF1 Playback", NULL, "ssp0 Tx"},
- {"ssp0 Tx", NULL, "codec1_out"},
- {"ssp0 Tx", NULL, "codec0_out"},
I get the routes to connect firmware widgets to codec ones, but why do we need SSP0 TX-> codec1_out? shouldn't this be part of the topology?
We still have routes for BEs defined here. Only FE ones come from topology.
- {"ssp0 Rx", NULL, "AIF1 Capture"},
- {"codec0_in", NULL, "ssp0 Rx"},
- {"Headphone Jack", NULL, "Platform Clock"},
- {"Mic Jack", NULL, "Platform Clock"},
+};
+static struct snd_soc_jack_pin cnl_headset_pins[] = {
- {
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
- },
- {
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
- },
+};
+static struct snd_soc_jack cnl_headset;
+static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{
- struct snd_soc_card *card = runtime->card;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
- struct snd_soc_component *component = codec_dai->component;
- int ret;
- ret = snd_soc_card_jack_new(runtime->card, "Headset",
SND_JACK_HEADSET, &cnl_headset,
cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins));
- if (ret)
return ret;
- ret = snd_soc_component_set_jack(component, &cnl_headset, NULL);
- if (ret)
return ret;
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24);
what are the 4 slots used for?
Ah, this is a mistake. Thanks for pointing out. It should be snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 24) as we do not have speakers on rt274. Will correct in v3.
On 3/15/18 10:40 PM, Guneshwor Singh wrote:
Hi Pierre,
On Thu, Mar 15, 2018 at 07:23:05AM -0500, Pierre-Louis Bossart wrote:
+static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
+{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai =
snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI);
- int ret, ratio = 100;
- if (!codec_dai)
return -EINVAL;
- /* Codec needs clock for Jack detection and button press */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2,
CNL_FREQ_OUT, SND_SOC_CLOCK_IN);
- if (ret < 0) {
dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret);
return ret;
- }
- if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio);
if (ret) {
dev_err(codec_dai->dev,
"set bclk ratio failed: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK,
CNL_BE_FIXUP_RATE * ratio,
CNL_FREQ_OUT);
if (ret) {
dev_err(codec_dai->dev,
"enable PLL2 failed: %d\n", ret);
return ret;
}
- }
- return 0;
+}
it's not clear to me why you need a clock control? You are not changing anything that really depends on DAPM events, to e.g. take the MCLK down and use a local clock, so could this be moved to hw_params?
Yes, we can do in hw_params too. When we implemented it we thought during simultaneous playback and capture, hw_params will be called twice but clock clock event will be called just once.
Yes, I remember this discussion but what's missing here is that PLL is always set to the max value, in most cases we fall back to a local clock to save power a bit. And it's probably wrong to leave a PLL on if the clock reference is turned off on the SOC side.
It does not harm to call twice as done in other machine drivers. Do you suggest to move it to hw_params?
+static const struct snd_soc_dapm_route cnl_map[] = {
- {"Headphone Jack", NULL, "HPO Pin"},
- {"MIC", NULL, "Mic Jack"},
- {"DMic", NULL, "SoC DMIC"},
- {"DMIC01 Rx", NULL, "Capture"},
- {"dmic01_hifi", NULL, "DMIC01 Rx"},
- {"AIF1 Playback", NULL, "ssp0 Tx"},
- {"ssp0 Tx", NULL, "codec1_out"},
- {"ssp0 Tx", NULL, "codec0_out"},
I get the routes to connect firmware widgets to codec ones, but why do we need SSP0 TX-> codec1_out? shouldn't this be part of the topology?
We still have routes for BEs defined here. Only FE ones come from topology.
The comment was about the codec1_out->SSP0 TX. Why is this hard-coded? You would only need SSP0 TX->AIF1 playback
- {"ssp0 Rx", NULL, "AIF1 Capture"},
- {"codec0_in", NULL, "ssp0 Rx"},
- {"Headphone Jack", NULL, "Platform Clock"},
- {"Mic Jack", NULL, "Platform Clock"},
+};
+static struct snd_soc_jack_pin cnl_headset_pins[] = {
- {
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
- },
- {
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
- },
+};
+static struct snd_soc_jack cnl_headset;
+static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{
- struct snd_soc_card *card = runtime->card;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
- struct snd_soc_component *component = codec_dai->component;
- int ret;
- ret = snd_soc_card_jack_new(runtime->card, "Headset",
SND_JACK_HEADSET, &cnl_headset,
cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins));
- if (ret)
return ret;
- ret = snd_soc_component_set_jack(component, &cnl_headset, NULL);
- if (ret)
return ret;
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24);
what are the 4 slots used for?
Ah, this is a mistake. Thanks for pointing out. It should be snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 24) as we do not have speakers on rt274. Will correct in v3.
then it also begs the question why you needed to have both codec0_out and codec1_out mentioned above - same issue really about hard-coding things that should be topology defined.
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On Fri, Mar 16, 2018 at 02:03:33PM -0500, Pierre-Louis Bossart wrote:
On 3/15/18 10:40 PM, Guneshwor Singh wrote:
Hi Pierre,
On Thu, Mar 15, 2018 at 07:23:05AM -0500, Pierre-Louis Bossart wrote:
+static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
+{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai =
snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI);
- int ret, ratio = 100;
- if (!codec_dai)
return -EINVAL;
- /* Codec needs clock for Jack detection and button press */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2,
CNL_FREQ_OUT, SND_SOC_CLOCK_IN);
- if (ret < 0) {
dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret);
return ret;
- }
- if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio);
if (ret) {
dev_err(codec_dai->dev,
"set bclk ratio failed: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK,
CNL_BE_FIXUP_RATE * ratio,
CNL_FREQ_OUT);
if (ret) {
dev_err(codec_dai->dev,
"enable PLL2 failed: %d\n", ret);
return ret;
}
- }
- return 0;
+}
it's not clear to me why you need a clock control? You are not changing anything that really depends on DAPM events, to e.g. take the MCLK down and use a local clock, so could this be moved to hw_params?
Yes, we can do in hw_params too. When we implemented it we thought during simultaneous playback and capture, hw_params will be called twice but clock clock event will be called just once.
Yes, I remember this discussion but what's missing here is that PLL is always set to the max value, in most cases we fall back to a local clock to save power a bit. And it's probably wrong to leave a PLL on if the clock reference is turned off on the SOC side.
Ok, got you. I should do stop in else part of it something like
else snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, 0, 0);
It does not harm to call twice as done in other machine drivers. Do you suggest to move it to hw_params?
+static const struct snd_soc_dapm_route cnl_map[] = {
- {"Headphone Jack", NULL, "HPO Pin"},
- {"MIC", NULL, "Mic Jack"},
- {"DMic", NULL, "SoC DMIC"},
- {"DMIC01 Rx", NULL, "Capture"},
- {"dmic01_hifi", NULL, "DMIC01 Rx"},
- {"AIF1 Playback", NULL, "ssp0 Tx"},
- {"ssp0 Tx", NULL, "codec1_out"},
- {"ssp0 Tx", NULL, "codec0_out"},
I get the routes to connect firmware widgets to codec ones, but why do we need SSP0 TX-> codec1_out? shouldn't this be part of the topology?
We still have routes for BEs defined here. Only FE ones come from topology.
The comment was about the codec1_out->SSP0 TX. Why is this hard-coded? You would only need SSP0 TX->AIF1 playback
Yes, to maintain similarity with other Intel machine drivers this was done. All routes of BE widgets are hard-coded. The same can be done from topology too. Let me remove these including codec0_in from here in v3.
- {"ssp0 Rx", NULL, "AIF1 Capture"},
- {"codec0_in", NULL, "ssp0 Rx"},
- {"Headphone Jack", NULL, "Platform Clock"},
- {"Mic Jack", NULL, "Platform Clock"},
+};
+static struct snd_soc_jack_pin cnl_headset_pins[] = {
- {
.pin = "Mic Jack",
.mask = SND_JACK_MICROPHONE,
- },
- {
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
- },
+};
+static struct snd_soc_jack cnl_headset;
+static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{
- struct snd_soc_card *card = runtime->card;
- struct snd_soc_dai *codec_dai = runtime->codec_dai;
- struct snd_soc_component *component = codec_dai->component;
- int ret;
- ret = snd_soc_card_jack_new(runtime->card, "Headset",
SND_JACK_HEADSET, &cnl_headset,
cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins));
- if (ret)
return ret;
- ret = snd_soc_component_set_jack(component, &cnl_headset, NULL);
- if (ret)
return ret;
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24);
what are the 4 slots used for?
Ah, this is a mistake. Thanks for pointing out. It should be snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 24) as we do not have speakers on rt274. Will correct in v3.
then it also begs the question why you needed to have both codec0_out and codec1_out mentioned above - same issue really about hard-coding things that should be topology defined.
It should by only codec0_out (codec1_out shouldn't be there at all).
Thanks for the review!
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
On 3/17/18 1:28 AM, Guneshwor Singh wrote:
On Fri, Mar 16, 2018 at 02:03:33PM -0500, Pierre-Louis Bossart wrote:
On 3/15/18 10:40 PM, Guneshwor Singh wrote:
Hi Pierre,
On Thu, Mar 15, 2018 at 07:23:05AM -0500, Pierre-Louis Bossart wrote:
+static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
+{
- struct snd_soc_dapm_context *dapm = w->dapm;
- struct snd_soc_card *card = dapm->card;
- struct snd_soc_dai *codec_dai =
snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI);
- int ret, ratio = 100;
- if (!codec_dai)
return -EINVAL;
- /* Codec needs clock for Jack detection and button press */
- ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2,
CNL_FREQ_OUT, SND_SOC_CLOCK_IN);
- if (ret < 0) {
dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret);
return ret;
- }
- if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio);
if (ret) {
dev_err(codec_dai->dev,
"set bclk ratio failed: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK,
CNL_BE_FIXUP_RATE * ratio,
CNL_FREQ_OUT);
if (ret) {
dev_err(codec_dai->dev,
"enable PLL2 failed: %d\n", ret);
return ret;
}
- }
- return 0;
+}
it's not clear to me why you need a clock control? You are not changing anything that really depends on DAPM events, to e.g. take the MCLK down and use a local clock, so could this be moved to hw_params?
Yes, we can do in hw_params too. When we implemented it we thought during simultaneous playback and capture, hw_params will be called twice but clock clock event will be called just once.
Yes, I remember this discussion but what's missing here is that PLL is always set to the max value, in most cases we fall back to a local clock to save power a bit. And it's probably wrong to leave a PLL on if the clock reference is turned off on the SOC side.
Ok, got you. I should do stop in else part of it something like
else snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, 0, 0);
something like that yes, but likely using an internal codec clock source for e.g. jack detection.
- /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
- ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24);
what are the 4 slots used for?
Ah, this is a mistake. Thanks for pointing out. It should be snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 24) as we do not have speakers on rt274. Will correct in v3.
then it also begs the question why you needed to have both codec0_out and codec1_out mentioned above - same issue really about hard-coding things that should be topology defined.
It should by only codec0_out (codec1_out shouldn't be there at all).
SSP0 is the BE so this need to be hard-codec. codec0_out is a topology item to me and should not be in a machine driver.
participants (2)
-
Guneshwor Singh
-
Pierre-Louis Bossart