December 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v5 0/6] ASoC: Intel: Skylake: Add a clk driver to enable ssp clks early
by Sriram Periyasamy 26 Jan '18

26 Jan '18

For certain platforms, clocks (mclk/sclk/fs) are required to be up before the stream start. Example: some codecs needs the mclk/sclk/fs to be enabled early for a successful clock synchronization. Some platforms require clock to be enabled at boot and be always ON. By sending set_dma_control IPC (with the i2s blobs queried from NHLT), these clocks can be enabled early after the firmware is downloaded. With this series, a virtual clock driver is created which provides interface to send the required IPCs from machine driver to enable the clocks. NHLT is parsed during probe and the clock information is populated. The pointer to blob is cached and sent along with the set_dma_control IPC structure during the clk prepare/unprepare callback. Clocks are created for a ssp if the nhlt table has endpoint configuration for that particular ssp. Kabylake machine driver uses the clock interface to enable the clocks early as it is required by the rt5663 driver for clock synchronization. v4 -> v5 - Remove checks for clock enable status from machine driver since taken care in the framework already - Add check in the skl_clk_set_rate to avoid different rates when clock is enabled already v3 -> v4 - Add missing signed-offs v2 -> v3 - Moved the clk ops and IPCs from Skylake driver to clk driver and reordered commits accordingly - Add the support for extended I2S blob config which supports multiple mclk dividers in NHLT - Enable the clocks as well in DAPM PMU event instead of hw_params in machine drivers as confirmed by codec vendor - Do not register the clk if there is no valid clock source is avail in the I2S blob - Take care of error return in the clk driver - Address rest of the review comments and more optimization added - Fix the warning sound/soc/intel/skylake/skl.c:724:1-3: WARNING: PTR_ERR_OR_ZERO can be used reported by scripts/coccinelle/api/ptr_ret.cocci - Modified DSP replies as human readable to ease the debugging - Add firmware replies for MCLK/SCLK clocks if they are running already v1 -> v2 - Register parent clocks with skylake device. With the patch "clk: Add support for runtime PM" soon to be merged will help DSP to stay active on call to clock enable. Reference: (https://patchwork.kernel.org/patch/9911741/) - Fix the machine driver to enable clocks early for headphone playback path as well to fix a pop noise issue Harsha Priya (1): ASoC: Intel: kbl: Enable mclk and ssp sclk early Naveen M (1): ASoC: Intel: eve: Enable mclk and ssp sclk early Sriram Periyasamy (2): ASoC: Intel: Skylake: Add ssp clock driver ASoC: Intel: Skylake: Add extended I2S config blob support in Clock driver Subhransu S. Prusty (2): ASoC: Intel: Skylake: Make DSP replies more human readable ASoC: Intel: Skylake: Add FW reply for MCLK/SCLK IPC sound/soc/intel/Kconfig | 3 + sound/soc/intel/boards/Kconfig | 2 + sound/soc/intel/boards/kbl_rt5663_max98927.c | 95 ++++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 94 ++++ sound/soc/intel/skylake/Makefile | 5 + sound/soc/intel/skylake/skl-i2s.h | 31 ++ sound/soc/intel/skylake/skl-messages.c | 1 + sound/soc/intel/skylake/skl-nhlt.c | 41 +- sound/soc/intel/skylake/skl-ssp-clk.c | 473 +++++++++++++++++++++ sound/soc/intel/skylake/skl-ssp-clk.h | 38 ++ sound/soc/intel/skylake/skl-sst-ipc.c | 50 ++- sound/soc/intel/skylake/skl.h | 6 + 12 files changed, 815 insertions(+), 24 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rockchip: Use dummy_dai for rt5514 dsp dailink
by Jeffy Chen 25 Jan '18

25 Jan '18

The rt5514 dsp captures pcm data through spi directly, so we should not use rockchip-i2s as it's cpu dai like other codecs. Use dummy_dai for rt5514 dsp dailink to make voice wakeup work again. Reported-by: Jimmy Cheng-Yi Chiang <cychiang(a)google.com> Fixes: (72cfb0f20c75 ASoC: rockchip: Use codec of_node and dai_name for rt5514 dsp) Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- sound/soc/rockchip/rk3399_gru_sound.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index d64fbbd50544..aa8ffd035377 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -367,7 +367,8 @@ static const struct snd_soc_dai_link rockchip_dais[] = { [DAILINK_RT5514_DSP] = { .name = "RT5514 DSP", .stream_name = "Wake on Voice", - .codec_dai_name = "rt5514-dsp-cpu-dai", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", }, }; @@ -528,7 +529,18 @@ static int rockchip_sound_of_parse_dais(struct device *dev, if (index < 0) continue; - np_cpu = (index == DAILINK_CDNDP) ? np_cpu1 : np_cpu0; + switch (index) { + case DAILINK_CDNDP: + np_cpu = np_cpu1; + break; + case DAILINK_RT5514_DSP: + np_cpu = np_codec; + break; + default: + np_cpu = np_cpu0; + break; + } + if (!np_cpu) { dev_err(dev, "Missing 'rockchip,cpu' for %s\n", rockchip_dais[index].name); @@ -538,7 +550,8 @@ static int rockchip_sound_of_parse_dais(struct device *dev, dai = &card->dai_link[card->num_links++]; *dai = rockchip_dais[index]; - dai->codec_of_node = np_codec; + if (!dai->codec_name) + dai->codec_of_node = np_codec; dai->platform_of_node = np_cpu; dai->cpu_of_node = np_cpu; -- 2.11.0

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[alsa-devel] [PATCH 0/2] sunxi: Add Capture support for sun8i-a33
by Mylène Josserand 25 Jan '18

25 Jan '18

Hello everyone, Here is a first series to add support for the capture for Allwinner Sun8I-A33 SoC (sun8i-codec). These patches have been tested on Sun8i-r16 parrot board and the two microphones are working well. Noticed that the DAPM route is not correct: "Right/Left Digital ADC Mixer" widgets should be after the "Right ADC" (according to what I understood from the datasheet). This is currently not the case but when I tried to update it, I got an error about "failing to add routes". I will investigate why in next weeks (and send a V2) but as it can take time, I think it is better to send a V1 and got reviews (because it is the first time I am implementing such features). Thank you in advance, Best regards, Mylène Mylène Josserand (2): ASoC: sun8i-codec: Add ADC support for a33 ARM: dts: sun8i: Add ADC routing arch/arm/boot/dts/sun8i-a33.dtsi | 10 ++++- sound/soc/sunxi/sun8i-codec.c | 82 +++++++++++++++++++++++++++++++++++++++- 2 files changed, 89 insertions(+), 3 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH for-4.15] ASoC: rt5514: don't assume rt5514 component was "attached"
by Brian Norris 24 Jan '18

24 Jan '18

I've found that on Google's "Kevin" Chromebook, the rt5514 codec might not be set up completely, yet its device is still present, and therefore its PM suspend/resume is called. This hits a NULL pointer exception, since we never had the chance to set our drvdata pointer. This resolves crashes seen when trying to resume my system. Fixes: e9c50aa6bd39 ("ASoC: rt5514-spi: check irq status to schedule data copy in resume function") Signed-off-by: Brian Norris <briannorris(a)chromium.org> --- This is a v4.15-rc1 regression sound/soc/codecs/rt5514-spi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 2df91db765ac..9255afcf2c3a 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -482,7 +482,7 @@ static int __maybe_unused rt5514_resume(struct device *dev) if (device_may_wakeup(dev)) disable_irq_wake(irq); - if (rt5514_dsp->substream) { + if (rt5514_dsp && rt5514_dsp->substream) { rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, sizeof(buf)); if (buf[0] & RT5514_IRQ_STATUS_BIT) rt5514_schedule_copy(rt5514_dsp); -- 2.15.1.504.g5279b80103-goog

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[alsa-devel] [PATCH] ASoC: soc-pcm: don't call flush_delayed_work() many times in soc_pcm_private_free()
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f523acebbb74 ("ASoC: add Component level pcm_new/pcm_free v2") added component level pcm_new/pcm_free, but flush_delayed_work() on soc_pcm_private_free() is called in for_each_rtdcom() loop. It doesn't need to be called many times. This patch moves it out of loop. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8075856..998800c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2831,10 +2831,9 @@ static void soc_pcm_private_free(struct snd_pcm *pcm) struct snd_soc_rtdcom_list *rtdcom; struct snd_soc_component *component; + /* need to sync the delayed work before releasing resources */ + flush_delayed_work(&rtd->delayed_work); for_each_rtdcom(rtd, rtdcom) { - /* need to sync the delayed work before releasing resources */ - - flush_delayed_work(&rtd->delayed_work); component = rtdcom->component; if (component->pcm_free) -- 1.9.1

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[alsa-devel] [PATCH v5] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 24 Jan '18

24 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v4 -> v5 - added Note on comment - tidyup comment (delayed work -> schedule_work) sound/soc/codecs/ak4613.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..d85c61e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,83 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * + * Note + * + * To avoid extra delay, we want to avoid preemption here, + * but we can't. Because it uses I2C access which is using IRQ + * and sleep. Thus, delay might be more than 5 LR clocks + * see also + * ak4613_dai_trigger() + */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for writing). + * Thus, use schedule_work() to switching to normal context + * immediately. + * + * Note + * + * Calling ak4613_dummy_write() function might be delayed. + * In such case, ak4613 volume might be temporarily 0dB when + * beggining of playback. + * see also + * ak4613_dummy_write() + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +668,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

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[alsa-devel] [PATCH 4/6] alsa-lib:pcm: fix the hw_ptr update until the boundary available.
by sutar.mounesh＠gmail.com 19 Jan '18

19 Jan '18

From: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> For long time test case, the slave_hw_ptr will exceed the boundary and wraparound the slave_hw_ptr. This slave boundary wraparound will cause the rate->hw_ptr to wraparound irrespective of the rate->boundary availability and due to that the available size goes wrong. Hence, To get the correct available size, - Its necessary to increment the rate->hw_ptr upto the rate->boundary and then wraparound the rate->hw_ptr. - While handling fraction part of slave period, rounded value will be introduced by input_frames(). To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Signed-off-by: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 1f830dd..5bee77c 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -50,7 +50,7 @@ typedef struct _snd_pcm_rate snd_pcm_rate_t; struct _snd_pcm_rate { snd_pcm_generic_t gen; - snd_pcm_uframes_t appl_ptr, hw_ptr; + snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; @@ -563,14 +563,28 @@ static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t sl { snd_pcm_rate_t *rate = pcm->private_data; + snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; + snd_pcm_sframes_t last_slave_hw_ptr_frac; + if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; - /* FIXME: boundary overlap of slave hw_ptr isn't evaluated here! - * e.g. if slave rate is small... - */ - rate->hw_ptr = - (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size + - rate->ops.input_frames(rate->obj, slave_hw_ptr % rate->gen.slave->period_size); + + if (slave_hw_ptr_diff < 0) + slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ + else if (slave_hw_ptr_diff == 0) + return; + last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; + /* While handling fraction part fo slave period, rounded value will be introduced by input_frames(). + * To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and + * add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Hence, + * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - + * fractional part of last_slave_hw_ptr rounded value + + * fractional part of updated slave hw ptr's rounded value ] */ + rate->hw_ptr += ( + (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - + rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + + rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + rate->last_slave_hw_ptr = slave_hw_ptr; rate->hw_ptr %= pcm->boundary; } @@ -635,6 +649,7 @@ static int snd_pcm_rate_prepare(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; @@ -650,6 +665,7 @@ static int snd_pcm_rate_reset(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; -- 1.7.9.5

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[alsa-devel] [PATCH V3 0/2] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 17 Jan '18

17 Jan '18

This small series fixes the probing of bcm2835-i2s, which is broken since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework"). Changes in V3: * split up patch into dt-binding and functional part Changes in V2: * add Fixes, Reviewed-by and Tested-by tags Stefan Wahren (2): dt-bindings: bcm283x: Fix register ranges of bcm2835-i2s ARM: dts: bcm283x: Fix probing of bcm2835-i2s Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) -- 2.7.4

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[alsa-devel] Silencing plops / click sounds on start / stop of playing audio when snd_hda_intel.power_save=1
by Hans de Goede 17 Jan '18

17 Jan '18

Hi All, For Fedora 28 we plan to ship the kernel with snd_hda_intel.power_save defaulting to 1: https://fedoraproject.org/wiki/Changes/ImprovedLaptopBatteryLife We've had an update for F27 in updates-testing for a while which made the same change for F27, but we've received several bug reports about plops / click sounds on start / stop of playing audio: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 I know others have probably already tried fixing this, but I'm wondering if there are any ideas how to fix this / things to try? Ideas of my own: 1) Mute the speakers / headphone output in the mixer before activating power-saving. Question how hard would it be to make this happen? 2) Put a delay between various steps involved in turning things off / on. Regards, Hans

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[alsa-devel] [PATCH 1/4] ASoC: codecs: tas5720: add basic support for TAS5722 devices
by Andrew F. Davis 15 Jan '18

15 Jan '18

From: Andreas Dannenberg <dannenberg(a)ti.com> The TI TAS5722 digital amplifier is very similar to the TAS5720 from an overall and register map perspective. Therefore the existing driver can be extended easily to support this additional device. This commit allows TAS5722 devices to be used in a "subset" type of fashion, without exposing any of the additional features they offer. Signed-off-by: Andreas Dannenberg <dannenberg(a)ti.com> Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- .../devicetree/bindings/sound/tas5720.txt | 4 ++- sound/soc/codecs/tas5720.c | 38 ++++++++++++++++++---- sound/soc/codecs/tas5720.h | 1 + 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas5720.txt b/Documentation/devicetree/bindings/sound/tas5720.txt index 40d94f82beb3..7481653fe8e3 100644 --- a/Documentation/devicetree/bindings/sound/tas5720.txt +++ b/Documentation/devicetree/bindings/sound/tas5720.txt @@ -6,10 +6,12 @@ audio playback. For more product information please see the links below: http://www.ti.com/product/TAS5720L http://www.ti.com/product/TAS5720M +http://www.ti.com/product/TAS5722L Required properties: -- compatible : "ti,tas5720" +- compatible : "ti,tas5720", + "ti,tas5722" - reg : I2C slave address - dvdd-supply : phandle to a 3.3-V supply for the digital circuitry - pvdd-supply : phandle to a supply used for the Class-D amp and the analog diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index a736a2a6976c..5def54d1336d 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -36,6 +36,11 @@ /* Define how often to check (and clear) the fault status register (in ms) */ #define TAS5720_FAULT_CHECK_INTERVAL 200 +enum tas572x_type { + TAS5720, + TAS5722, +}; + static const char * const tas5720_supply_names[] = { "dvdd", /* Digital power supply. Connect to 3.3-V supply. */ "pvdd", /* Class-D amp and analog power supply (connected). */ @@ -47,6 +52,7 @@ struct tas5720_data { struct snd_soc_codec *codec; struct regmap *regmap; struct i2c_client *tas5720_client; + enum tas572x_type devtype; struct regulator_bulk_data supplies[TAS5720_NUM_SUPPLIES]; struct delayed_work fault_check_work; unsigned int last_fault; @@ -264,7 +270,7 @@ static void tas5720_fault_check_work(struct work_struct *work) static int tas5720_codec_probe(struct snd_soc_codec *codec) { struct tas5720_data *tas5720 = snd_soc_codec_get_drvdata(codec); - unsigned int device_id; + unsigned int device_id, expected_device_id; int ret; tas5720->codec = codec; @@ -276,6 +282,11 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) return ret; } + /* + * Take a liberal approach to checking the device ID to allow the + * driver to be used even if the device ID does not match, however + * issue a warning if there is a mismatch. + */ ret = regmap_read(tas5720->regmap, TAS5720_DEVICE_ID_REG, &device_id); if (ret < 0) { dev_err(codec->dev, "failed to read device ID register: %d\n", @@ -283,13 +294,22 @@ static int tas5720_codec_probe(struct snd_soc_codec *codec) goto probe_fail; } - if (device_id != TAS5720_DEVICE_ID) { - dev_err(codec->dev, "wrong device ID. expected: %u read: %u\n", - TAS5720_DEVICE_ID, device_id); - ret = -ENODEV; - goto probe_fail; + switch (tas5720->devtype) { + case TAS5720: + expected_device_id = TAS5720_DEVICE_ID; + break; + case TAS5722: + expected_device_id = TAS5722_DEVICE_ID; + break; + default: + dev_err(codec->dev, "unexpected private driver data\n"); + return -EINVAL; } + if (device_id != expected_device_id) + dev_warn(codec->dev, "wrong device ID. expected: %u read: %u\n", + expected_device_id, device_id); + /* Set device to mute */ ret = snd_soc_update_bits(codec, TAS5720_DIGITAL_CTRL2_REG, TAS5720_MUTE, TAS5720_MUTE); @@ -552,6 +572,8 @@ static int tas5720_probe(struct i2c_client *client, return -ENOMEM; data->tas5720_client = client; + data->devtype = id->driver_data; + data->regmap = devm_regmap_init_i2c(client, &tas5720_regmap_config); if (IS_ERR(data->regmap)) { ret = PTR_ERR(data->regmap); @@ -592,7 +614,8 @@ static int tas5720_remove(struct i2c_client *client) } static const struct i2c_device_id tas5720_id[] = { - { "tas5720", 0 }, + { "tas5720", TAS5720 }, + { "tas5722", TAS5722 }, { } }; MODULE_DEVICE_TABLE(i2c, tas5720_id); @@ -600,6 +623,7 @@ MODULE_DEVICE_TABLE(i2c, tas5720_id); #if IS_ENABLED(CONFIG_OF) static const struct of_device_id tas5720_of_match[] = { { .compatible = "ti,tas5720", }, + { .compatible = "ti,tas5722", }, { }, }; MODULE_DEVICE_TABLE(of, tas5720_of_match); diff --git a/sound/soc/codecs/tas5720.h b/sound/soc/codecs/tas5720.h index 3d077c779b12..bef802afcc69 100644 --- a/sound/soc/codecs/tas5720.h +++ b/sound/soc/codecs/tas5720.h @@ -32,6 +32,7 @@ /* TAS5720_DEVICE_ID_REG */ #define TAS5720_DEVICE_ID 0x01 +#define TAS5722_DEVICE_ID 0x12 /* TAS5720_POWER_CTRL_REG */ #define TAS5720_DIG_CLIP_MASK GENMASK(7, 2) -- 2.15.0

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