Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart...

Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.

Bard Liao (1):
  ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name

Kai Vehmanen (2):
  ASoC: Intel: sof_sdw: add support for systems without i915 audio
  ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded

Libin Yang (1):
  ASoC: Intel: common: change match table ehl-rt5660

Pierre-Louis Bossart (1):
  ASoC: Intel: sof_sdw_rt711: remove properties in card remove

Yong Zhi (1):
  ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq

 sound/soc/intel/boards/sof_rt5682.c           |  9 +++++-
 sound/soc/intel/boards/sof_sdw.c              | 31 +++++++++++++------
 sound/soc/intel/boards/sof_sdw_common.h       |  2 ++
 sound/soc/intel/boards/sof_sdw_hdmi.c         |  6 ++++
 sound/soc/intel/boards/sof_sdw_rt711.c        | 17 +++++++++-
 .../intel/common/soc-acpi-intel-ehl-match.c   |  2 +-
 6 files changed, 54 insertions(+), 13 deletions(-)

base-commit: 22e9b54307987787efa0ee534aa9e31982ec1161
--
2.25.1

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 7ff659e285702d1815c5f9b82701117f8f494ab1..776a60529e70a9a638cb26f50755d71b3740c388 100644 (file)
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -161,6 +161,7 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai);
 int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
                             struct snd_soc_pcm_runtime *rtd, int num);
 bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
 void snd_soc_dai_action(struct snd_soc_dai *dai,
                        int stream, int action);
 static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 67b5faa64ec3492cde0f51d305d9d500f145cd7e..929bb1798c43f9a5c03af199052a9682aae8ae34 100644 (file)
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -331,13 +331,6 @@ static int max98373_probe(struct snd_soc_component *component)
        regmap_write(max98373->regmap,
                MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2,
                0x1);
-       /* Set inital volume (0dB) */
-       regmap_write(max98373->regmap,
-               MAX98373_R203D_AMP_DIG_VOL_CTRL,
-               0x00);
-       regmap_write(max98373->regmap,
-               MAX98373_R203E_AMP_PATH_GAIN,
-               0x00);
        /* Enable DC blocker */
        regmap_write(max98373->regmap,
                MAX98373_R203F_AMP_DSP_CFG,
@@ -397,7 +390,6 @@ const struct snd_soc_component_driver soc_codec_dev_max98373 = {
        .num_dapm_widgets       = ARRAY_SIZE(max98373_dapm_widgets),
        .dapm_routes            = max98373_audio_map,
        .num_dapm_routes        = ARRAY_SIZE(max98373_audio_map),
-       .idle_bias_on           = 1,
        .use_pmdown_time        = 1,
        .endianness             = 1,
        .non_legacy_dai_naming  = 1,

diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 89b1c8b680047519c6f44dc5762bcb47920103b4..5fb9653d9131f313d0b4f4c00bf6107be9f9f7a4 100644 (file)
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
                regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
                *mic = buf & 0x80000000;
        }
-       if (!*mic) {
+
+       if (!*hp) {
                snd_soc_dapm_disable_pin(dapm, "HV");
                snd_soc_dapm_disable_pin(dapm, "VREF");
-       }
-       if (!*hp)
                snd_soc_dapm_disable_pin(dapm, "LDO1");
-       snd_soc_dapm_sync(dapm);
+               snd_soc_dapm_sync(dapm);
+       }
 
        return 0;
 }

diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index fab066a75ce0254772709024cc747697a9540509..a4713bd6508d52bfed3087a97d96bd1e425fa202 100644 (file)
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -970,13 +970,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
                rt5682_enable_push_button_irq(component, false);
                snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
                        RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-               if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+               if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
                        snd_soc_component_update_bits(component,
-                               RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
-               else
+                               RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
+               if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
                        snd_soc_component_update_bits(component,
-                               RT5682_PWR_ANLG_1,
-                               RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+                               RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
                snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
                        RT5682_PWR_CBJ, 0);
                snd_soc_component_update_bits(component, RT5682_MICBIAS_2,
@@ -1088,7 +1087,8 @@ void rt5682_jack_detect_handler(struct work_struct *work)
                        /* jack was out, report jack type */
                        rt5682->jack_type =
                                rt5682_headset_detect(rt5682->component, 1);
-               } else {
+               } else if ((rt5682->jack_type & SND_JACK_HEADSET) ==
+                       SND_JACK_HEADSET) {
                        /* jack is already in, report button event */
                        rt5682->jack_type = SND_JACK_HEADSET;
                        btn_type = rt5682_button_detect(rt5682->component);
@@ -1614,8 +1614,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
                0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
        SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
                rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
-       SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
-               NULL, 0),
+       SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0),
        SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
 
        /* ASRC */
@@ -2505,6 +2504,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw)
        snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
        snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
                                RT5682_PWR_MB, RT5682_PWR_MB);
+
+       snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2");
+       snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+                       RT5682_PWR_VREF2 | RT5682_PWR_FV2,
+                       RT5682_PWR_VREF2);
+       usleep_range(55000, 60000);
+       snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+                       RT5682_PWR_FV2, RT5682_PWR_FV2);
+
        snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
        snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
        snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
@@ -2530,9 +2538,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw)
        snd_soc_dapm_mutex_lock(dapm);
 
        snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+       snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2");
        if (!rt5682->jack_type)
                snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+                               RT5682_PWR_VREF2 | RT5682_PWR_FV2 |
                                RT5682_PWR_MB, 0);
+
        snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
        snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
        snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");

diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 89c6507d5566be0f793ca0427824e8fd71b24f85..c86231dfcf4f84d68cf127addfe704c203b4bab8 100644 (file)
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
 
 /* Boost mixer */
 static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
 };
 
 /* Input PGA */
@@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
                iface |= 0x0008;
                break;
        case SND_SOC_DAIFMT_DSP_A:
+               if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF ||
+                   (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) {
+                       return -EINVAL;
+               }
                iface |= 0x00018;
                break;
        default:

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 9ad35d9940fef6436761ed9d851e06610e7e138c..97b4f5480a31cb94c6c39a2a7d6466744e7d017d 100644 (file)
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
        if (ret < 0)
                goto out_put_node;
 
-       dai_link->dpcm_playback         = 1;
-       dai_link->dpcm_capture          = 1;
+       snd_soc_dai_link_set_capabilities(dai_link);
+
        dai_link->ops                   = &graph_ops;
        dai_link->init                  = asoc_simple_dai_init;
 

diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 55e9f8800b3e103638e38b651aa9e37ac37fdcc7..04d4d28ed5112891cfce65920247c81a929e289e 100644 (file)
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
        if (ret < 0)
                goto out_put_node;
 
-       dai_link->dpcm_playback         = 1;
-       dai_link->dpcm_capture          = 1;
+       snd_soc_dai_link_set_capabilities(dai_link);
+
        dai_link->ops                   = &simple_ops;
        dai_link->init                  = asoc_simple_dai_init;
 

diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index c9da91147770112d5d59cdbe93d6b348137e4ad3..725304779426066f4261b670b99045562427ecc7 100644 (file)
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -367,6 +367,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
        {
                .name = "Codec DSP",
                .stream_name = "Wake on Voice",
+               .capture_only = 1,
                .ops = &bdw_rt5677_dsp_ops,
                SND_SOC_DAILINK_REG(dsp),
        },

diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 71b39e579af9c280e5e85481ca765976d87a11ef..414ae4bb5224a96d58d2586c2d2d092395ebe988 100644 (file)
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -552,8 +552,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 
        if (cnt) {
                ret = device_add_properties(codec_dev, props);
-               if (ret)
+               if (ret) {
+                       put_device(codec_dev);
                        return ret;
+               }
        }
 
        devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);

diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
index 45e07d8860137c0f7464da975170bd567fce8be3..badafc1d54d20f49aa3fdc52909d65cc53e20dd0 100644 (file)
--- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
@@ -12,7 +12,7 @@
 
 struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = {
        {
-               .id = "INTC1027",
+               .id = "10EC5660",
                .drv_name = "ehl_rt5660",
                .sof_fw_filename = "sof-ehl.ri",
                .sof_tplg_filename = "sof-ehl-rt5660.tplg",

diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 89f7f64747cd0a562e3915fbf3918947b37d671e..47f2d93224fea963f851acff3d7e7db45985cc16 100644 (file)
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 
        lb = &card->dai_link[*index + 1];
 
-       lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+       lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
        if (!lb->name)
                return -ENOMEM;
 

diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index c89a1929d141aea7ae2ed43081408c879527a36b..98f0c98b06bbd78dbe56e39d80d8ba4be5e6e24b 100644 (file)
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -393,6 +393,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
        return stream->channels_min;
 }
 
+/*
+ * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs
+ */
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
+{
+       struct snd_soc_dai_link_component *cpu;
+       struct snd_soc_dai_link_component *codec;
+       struct snd_soc_dai *dai;
+       bool supported[SNDRV_PCM_STREAM_LAST + 1];
+       int direction;
+       int i;
+
+       for_each_pcm_streams(direction) {
+               supported[direction] = true;
+
+               for_each_link_cpus(dai_link, i, cpu) {
+                       dai = snd_soc_find_dai(cpu);
+                       if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+                               supported[direction] = false;
+                               break;
+                       }
+               }
+               if (!supported[direction])
+                       continue;
+               for_each_link_codecs(dai_link, i, codec) {
+                       dai = snd_soc_find_dai(codec);
+                       if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+                               supported[direction] = false;
+                               break;
+                       }
+               }
+       }
+
+       dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
+       dai_link->dpcm_capture  = supported[SNDRV_PCM_STREAM_CAPTURE];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities);
+
 void snd_soc_dai_action(struct snd_soc_dai *dai,
                        int stream, int action)
 {

diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c4930b0c0a77d0c613a4d3b41ee982bd2a66b..adc7c37145d6407c173c942cd58f61103f0c3bbb 100644 (file)
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev)
        struct snd_sof_pdata *pdata = sdev->pdata;
        int ret;
 
-       ret = snd_sof_dsp_power_down_notify(sdev);
-       if (ret < 0)
-               dev_warn(dev, "error: %d failed to prepare DSP for device removal",
-                        ret);
-
        if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
                cancel_work_sync(&sdev->probe_work);
 
        if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+               ret = snd_sof_dsp_power_down_notify(sdev);
+               if (ret < 0)
+                       dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+                                ret);
+
                snd_sof_fw_unload(sdev);
                snd_sof_ipc_free(sdev);
                snd_sof_free_debug(sdev);

diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20a1bacf57b2e0cf3cadd2e0f636d486215..a4fa8451d8cb39feaedf7e0d26b5a7f1d57e955c 100644 (file)
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8_dai[] = {
 {
        .name = "esai-port",
+       .playback = {
+               .channels_min = 1,
+               .channels_max = 8,
+       },
+       .capture = {
+               .channels_min = 1,
+               .channels_max = 8,
+       },
 },
 };
 

diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e2990f8722aebebf8f591333d948c53a02..287114a37688c676f503495a2c95c8032683ce92 100644 (file)
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8m_dai[] = {
 {
        .name = "sai-port",
+       .playback = {
+               .channels_min = 1,
+               .channels_max = 32,
+       },
+       .capture = {
+               .channels_min = 1,
+               .channels_max = 32,
+       },
 },
 };