From fb629fa2587d0c150792d87e3053664bfc8dc78c Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 20 Sep 2019 15:02:11 +0200 Subject: ASoC: samsung: arndale: Add missing OF node dereferencing Ensure there is no OF node references kept when the driver is removed/unbound. Reviewed-by: Charles Keepax Signed-off-by: Sylwester Nawrocki Acked-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20190920130218.32690-3-s.nawrocki@samsung.com Signed-off-by: Mark Brown --- sound/soc/samsung/arndale_rt5631.c | 34 ++++++++++++++++++++++++++++++---- 1 file changed, 30 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index c213913eb984..fd8c6642fb0d 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -5,6 +5,7 @@ // Author: Claude #include +#include #include #include @@ -74,6 +75,17 @@ static struct snd_soc_card arndale_rt5631 = { .num_links = ARRAY_SIZE(arndale_rt5631_dai), }; +static void arndale_put_of_nodes(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + int i; + + for_each_card_prelinks(card, i, dai_link) { + of_node_put(dai_link->cpus->of_node); + of_node_put(dai_link->codecs->of_node); + } +} + static int arndale_audio_probe(struct platform_device *pdev) { int n, ret; @@ -103,18 +115,31 @@ static int arndale_audio_probe(struct platform_device *pdev) if (!arndale_rt5631_dai[0].codecs->of_node) { dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto err_put_of_nodes; } } ret = devm_snd_soc_register_card(card->dev, card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + goto err_put_of_nodes; + } + return 0; - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); - +err_put_of_nodes: + arndale_put_of_nodes(card); return ret; } +static int arndale_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + arndale_put_of_nodes(card); + return 0; +} + static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = { { .compatible = "samsung,arndale-rt5631", }, { .compatible = "samsung,arndale-alc5631", }, @@ -129,6 +154,7 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match), }, .probe = arndale_audio_probe, + .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); -- cgit v1.2.3 From ca2347190adb5e4eece73a2b16e96e651c46246b Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 20 Sep 2019 15:02:10 +0200 Subject: ASoC: wm8994: Do not register inapplicable controls for WM1811 In case of WM1811 device there are currently being registered controls referring to registers not existing on that device. It has been noticed when getting values of "AIF1ADC2 Volume", "AIF1DAC2 Volume" controls was failing during ALSA state restoring at boot time: "amixer: Mixer hw:0 load error: Device or resource busy" Reading some registers through I2C was failing with EBUSY error and indeed these registers were not available according to the datasheet. To fix this controls not available on WM1811 are moved to a separate array and registered only for WM8994 and WM8958. There are some further differences between WM8994 and WM1811, e.g. registers 603h, 604h, 605h, which are not covered in this patch. Acked-by: Charles Keepax Acked-by: Krzysztof Kozlowski Signed-off-by: Sylwester Nawrocki Link: https://lore.kernel.org/r/20190920130218.32690-2-s.nawrocki@samsung.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 43 ++++++++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c3d06e8bc54f..d5fb7f5dd551 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -533,13 +533,10 @@ static SOC_ENUM_SINGLE_DECL(dac_osr, static SOC_ENUM_SINGLE_DECL(adc_osr, WM8994_OVERSAMPLING, 1, osr_text); -static const struct snd_kcontrol_new wm8994_snd_controls[] = { +static const struct snd_kcontrol_new wm8994_common_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1_ADC1_RIGHT_VOLUME, 1, 119, 0, digital_tlv), -SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, - WM8994_AIF1_ADC2_RIGHT_VOLUME, - 1, 119, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2_ADC_RIGHT_VOLUME, 1, 119, 0, digital_tlv), @@ -556,8 +553,6 @@ SOC_ENUM("AIF2DACR Source", aif2dacr_src), SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv), -SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, - WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv), @@ -565,17 +560,12 @@ SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv), SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv), SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0), -SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0), WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2), WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1), WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0), -WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), -WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), -WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), - WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2), WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1), WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0), @@ -594,9 +584,6 @@ SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0), SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf), SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0), -SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf), -SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), - SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf), SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0), @@ -637,6 +624,24 @@ SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2, 8, 1, 0), }; +/* Controls not available on WM1811 */ +static const struct snd_kcontrol_new wm8994_snd_controls[] = { +SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1_ADC2_RIGHT_VOLUME, + 1, 119, 0, digital_tlv), +SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv), + +SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0), + +WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2), +WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1), +WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0), + +SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf), +SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0), +}; + static const struct snd_kcontrol_new wm8994_eq_controls[] = { SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0, eq_tlv), @@ -4258,13 +4263,15 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_handle_pdata(wm8994); wm_hubs_add_analogue_controls(component); - snd_soc_add_component_controls(component, wm8994_snd_controls, - ARRAY_SIZE(wm8994_snd_controls)); + snd_soc_add_component_controls(component, wm8994_common_snd_controls, + ARRAY_SIZE(wm8994_common_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets, ARRAY_SIZE(wm8994_dapm_widgets)); switch (control->type) { case WM8994: + snd_soc_add_component_controls(component, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); if (control->revision < 4) { @@ -4284,8 +4291,10 @@ static int wm8994_component_probe(struct snd_soc_component *component) } break; case WM8958: + snd_soc_add_component_controls(component, wm8994_snd_controls, + ARRAY_SIZE(wm8994_snd_controls)); snd_soc_add_component_controls(component, wm8958_snd_controls, - ARRAY_SIZE(wm8958_snd_controls)); + ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); if (control->revision < 1) { -- cgit v1.2.3 From 901e822b2e365dac4727e0ddffb444a2554b0a89 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 23 Sep 2019 17:22:57 +0300 Subject: ASoC: soc-component: fix a couple missing error assignments There were a couple places where the return value wasn't assigned so the error handling wouldn't trigger. Fixes: 5c0769af4caf ("ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()") Fixes: 95aef3553384 ("ASoC: soc-dai: add snd_soc_dai_trigger()") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/20190923142257.GB31251@mwanda Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..a1b99ac57d9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; -- cgit v1.2.3 From 752c938a5c14b8cbf0ed3ffbfa637fb166255c3f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Sep 2019 14:06:24 +0300 Subject: ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create() The "template.id" variable is an enum and in this context GCC will treat it as an unsigned int so it can never be less than zero. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/20190925110624.GR3264@mwanda Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index b8690715abb5..c25939c5611b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1588,7 +1588,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, /* map user to kernel widget ID */ template.id = get_widget_id(le32_to_cpu(w->id)); - if (template.id < 0) + if ((int)template.id < 0) return template.id; /* strings are allocated here, but used and freed by the widget */ -- cgit v1.2.3 From 4bb41984bf2f4cb8ed6ec1579d317790bd941788 Mon Sep 17 00:00:00 2001 From: Sathyanarayana Nujella Date: Sat, 28 Sep 2019 13:22:30 -0700 Subject: ASoC: max98373: check for device node before parsing Below Oops is caused in a system which uses ACPI instead of device node: of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]' BUG: kernel NULL pointer dereference, address: 0000000000000010 This patch avoids NULL pointer deferencing by adding a check before parsing and initializes to make reset-gpio pin as invalid. Signed-off-by: Sathyanarayana Nujella Signed-off-by: Jairaj Arava Link: https://lore.kernel.org/r/1569702150-11976-1-git-send-email-sathyanarayana.nujella@intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) -- cgit v1.2.3 From 9daf4fd0302b2559223cf90dae7dc510c6679047 Mon Sep 17 00:00:00 2001 From: Li Xu Date: Tue, 1 Oct 2019 14:09:11 +0100 Subject: ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region Fix potential NULL pointer dereference for alg_region in wm_adsp_buffer_parse_legacy. In practice this can never happen as loading the firmware should have failed at the wm_adsp2_setup_algs stage, however probably better for the code to be robust against future changes and this is more helpful for static analysis. Signed-off-by: Li Xu Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20191001130911.19238-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae28d9907c30..85396d920e0a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3697,11 +3697,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) u32 xmalg, addr, magic; int i, ret; + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + if (!alg_region) { + adsp_err(dsp, "No algorithm region found\n"); + return -EINVAL; + } + buf = wm_adsp_buffer_alloc(dsp); if (!buf) return -ENOMEM; - alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); -- cgit v1.2.3 From 798614885a0e1b867ceb0197c30c2d82575c73b0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 27 Sep 2019 15:05:26 -0500 Subject: ASoC: SOF: loader: fix kernel oops on firmware boot failure When we fail to boot the firmware, we encounter a kernel oops in hda_dsp_get_registers(), which is called conditionally in hda_dsp_dump() when the sdev_>boot_complete flag is set. Setting this flag _after_ dumping the data fixes the issue and does not change the programming flow. Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/loader.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index d7f32745fefe..9a9a381a908d 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -546,10 +546,10 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev) msecs_to_jiffies(sdev->boot_timeout)); if (ret == 0) { dev_err(sdev->dev, "error: firmware boot failure\n"); - /* after this point FW_READY msg should be ignored */ - sdev->boot_complete = true; snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX | SOF_DBG_TEXT | SOF_DBG_PCI); + /* after this point FW_READY msg should be ignored */ + sdev->boot_complete = true; return -EIO; } -- cgit v1.2.3 From 2e305a074061121220a2828f97a57d315cf8efba Mon Sep 17 00:00:00 2001 From: Keyon Jie Date: Fri, 27 Sep 2019 15:05:27 -0500 Subject: ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types We are using sof_parse_word_tokens() to parse tokens with bool/byte/short/word tuple types, here add the missing check, to fix the parsing failure at byte/bool tuple types. Signed-off-by: Keyon Jie Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index fc85efbad378..0aabb3190ddc 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -920,7 +920,9 @@ static void sof_parse_word_tokens(struct snd_soc_component *scomp, for (j = 0; j < count; j++) { /* match token type */ if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD || - tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT)) + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT || + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BYTE || + tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BOOL)) continue; /* match token id */ -- cgit v1.2.3 From e66e52c5b7422824cedf0084c0766602dea7e8a7 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 27 Sep 2019 15:05:30 -0500 Subject: ASoC: SOF: pcm: fix resource leak in hw_free Fix a bug in sof_pcm_hw_free() where some cleanup actions were skipped if STREAM_PCM_FREE IPC was already successfully sent to DSP when the stream was stopped or suspended. This is incorrect as hw_free should clean up also other resources, including pcm lib page allocations, period elapsed work queue and call to platform hw_free. Fixes: c29d96c3b9b4 ("ASoC: SOF: reset DMA state in prepare") Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index e3f6a6dc0f36..fa7769dd825c 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -244,7 +244,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream) snd_soc_rtdcom_lookup(rtd, DRV_NAME); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_sof_pcm *spcm; - int ret; + int ret, err = 0; /* nothing to do for BE */ if (rtd->dai_link->no_pcm) @@ -254,26 +254,26 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream) if (!spcm) return -EINVAL; - if (!spcm->prepared[substream->stream]) - return 0; - dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id, substream->stream); - ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (spcm->prepared[substream->stream]) { + ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (ret < 0) + err = ret; + } snd_pcm_lib_free_pages(substream); cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work); - if (ret < 0) - return ret; - ret = snd_sof_pcm_platform_hw_free(sdev, substream); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: platform hw free failed\n"); + err = ret; + } - return ret; + return err; } static int sof_pcm_prepare(struct snd_pcm_substream *substream) -- cgit v1.2.3 From 0a1b08345bc5d9214dc701f8ec5d67c473fab735 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Fri, 27 Sep 2019 15:05:31 -0500 Subject: ASoC: SOF: pcm: harden PCM STOP sequence The old STOP sequence is: 1. stop DMA 2. send STOP ipc If delay happen before the steps 1 and 2, the DMA buffer will be empty in short time and cause pipeline xrun then stop the pipeline. Then the step 2 ipc stop will return error as pipeline is already stopped. Suggested change to avoid the issue is to switch the order of steps 1 and 2 for the stop sequence. Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index fa7769dd825c..2b876d497447 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -323,6 +323,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct sof_ipc_stream stream; struct sof_ipc_reply reply; bool reset_hw_params = false; + bool ipc_first = false; int ret; /* nothing to do for BE */ @@ -343,6 +344,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_PAUSE_PUSH: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + ipc_first = true; break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; @@ -363,6 +365,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + ipc_first = true; reset_hw_params = true; break; default: @@ -370,12 +373,22 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return -EINVAL; } - snd_sof_pcm_platform_trigger(sdev, substream, cmd); + /* + * DMA and IPC sequence is different for start and stop. Need to send + * STOP IPC before stop DMA + */ + if (!ipc_first) + snd_sof_pcm_platform_trigger(sdev, substream, cmd); /* send IPC to the DSP */ ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream, sizeof(stream), &reply, sizeof(reply)); + /* need to STOP DMA even if STOP IPC failed */ + if (ipc_first) + snd_sof_pcm_platform_trigger(sdev, substream, cmd); + + /* free PCM if reset_hw_params is set and the STOP IPC is successful */ if (!ret && reset_hw_params) ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); -- cgit v1.2.3 From 4ff5f6439fe69624e8f7d559915e9b54a6477684 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Fri, 27 Sep 2019 15:05:35 -0500 Subject: ASoC: SOF: Intel: hda: fix warnings during FW load The "snd_pcm_substream" handle was not initialized properly in hda-loader.c for firmware load. When the HDA DMAs were used to load the firmware, the interrupts related to firmware load also triggered calls to snd_sof_pcm_period_elapsed() on a non-existent ALSA PCM stream. This caused runtime kernel warnings from pcm_lib.c:snd_pcm_period_elapsed(). Signed-off-by: Kai Vehmanen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-11-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 6427f0b3a2f1..65c2af3fcaab 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -44,6 +44,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format, return -ENODEV; } hstream = &dsp_stream->hstream; + hstream->substream = NULL; /* allocate DMA buffer */ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); -- cgit v1.2.3 From ff2be865633e6fa523cd2db3b73197d795dec991 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 27 Sep 2019 15:05:36 -0500 Subject: ASoC: SOF: Intel: initialise and verify FW crash dump data. FW mailbox offset was not set before use and HDR size was not validated. Fix this. Signed-off-by: Liam Girdwood Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-12-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/bdw.c | 7 +++++++ sound/soc/sof/intel/byt.c | 6 ++++++ sound/soc/sof/intel/hda.c | 7 +++++++ 3 files changed, 20 insertions(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index e282179263e8..80e2826fb447 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -37,6 +37,7 @@ #define MBOX_SIZE 0x1000 #define MBOX_DUMP_SIZE 0x30 #define EXCEPT_OFFSET 0x800 +#define EXCEPT_MAX_HDR_SIZE 0x400 /* DSP peripherals */ #define DMAC0_OFFSET 0xFE000 @@ -228,6 +229,11 @@ static void bdw_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info)); @@ -451,6 +457,7 @@ static int bdw_probe(struct snd_sof_dev *sdev) /* TODO: add offsets */ sdev->mmio_bar = BDW_DSP_BAR; sdev->mailbox_bar = BDW_DSP_BAR; + sdev->dsp_oops_offset = MBOX_OFFSET; /* PCI base */ mmio = platform_get_resource(pdev, IORESOURCE_MEM, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 5e7a6aaa627a..a1e514f71739 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -28,6 +28,7 @@ #define MBOX_OFFSET 0x144000 #define MBOX_SIZE 0x1000 #define EXCEPT_OFFSET 0x800 +#define EXCEPT_MAX_HDR_SIZE 0x400 /* DSP peripherals */ #define DMAC0_OFFSET 0x098000 @@ -126,6 +127,11 @@ static void byt_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info)); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c72e9a09eee1..06e84679087b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -35,6 +35,8 @@ #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) +#define EXCEPT_MAX_HDR_SIZE 0x400 + /* * Debug */ @@ -131,6 +133,11 @@ static void hda_dsp_get_registers(struct snd_sof_dev *sdev, /* note: variable AR register array is not read */ /* then get panic info */ + if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) { + dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n", + xoops->arch_hdr.totalsize); + return; + } offset += xoops->arch_hdr.totalsize; sof_block_read(sdev, sdev->mmio_bar, offset, panic_info, sizeof(*panic_info)); -- cgit v1.2.3 From 43b2ab9009b13bfff47fcc1893de9244b39bdd54 Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan Date: Fri, 27 Sep 2019 15:05:38 -0500 Subject: ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture There is a known issue on some Intel platforms which causes pause/release to run into xrun's during capture usecases. The suggested workaround to address the issue is to disable the entry of lower power L1 state in the physical DMI link when there is a capture stream open. Signed-off-by: Ranjani Sridharan Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927200538.660-14-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 10 +++++++++ sound/soc/sof/intel/hda-ctrl.c | 12 ++++------- sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++++++++++++++------- sound/soc/sof/intel/hda.h | 5 ++++- 4 files changed, 56 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..d62f51d33be1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -273,6 +273,16 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC Say Y if you want to enable HDAudio codecs with SOF. If unsure select "N". +config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1 + bool "SOF enable DMI Link L1" + help + This option enables DMI L1 for both playback and capture + and disables known workarounds for specific HDaudio platforms. + Only use to look into power optimizations on platforms not + affected by DMI L1 issues. This option is not recommended. + Say Y if you want to enable DMI Link L1 + If unsure, select "N". + endif ## SND_SOC_SOF_HDA_COMMON config SND_SOC_SOF_HDA_LINK_BASELINE diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index bc41028a7a01..df1909e1d950 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -139,20 +139,16 @@ void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable) */ int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable) { -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - struct hdac_bus *bus = sof_to_bus(sdev); -#endif u32 val; /* enable/disable audio dsp clock gating */ val = enable ? PCI_CGCTL_ADSPDCGE : 0; snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val); -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - /* enable/disable L1 support */ - val = enable ? SOF_HDA_VS_EM2_L1SEN : 0; - snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val); -#endif + /* enable/disable DMI Link L1 support */ + val = enable ? HDA_VS_INTEL_EM2_L1SEN : 0; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, val); /* enable/disable audio dsp power gating */ val = enable ? 0 : PCI_PGCTL_ADSPPGD; diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index ad8d41f22e92..2c7447188402 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -185,6 +185,17 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction) direction == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + /* + * Disable DMI Link L1 entry when capture stream is opened. + * Workaround to address a known issue with host DMA that results + * in xruns during pause/release in capture scenarios. + */ + if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (stream && direction == SNDRV_PCM_STREAM_CAPTURE) + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, 0); + return stream; } @@ -193,23 +204,43 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) { struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *s; + bool active_capture_stream = false; + bool found = false; spin_lock_irq(&bus->reg_lock); - /* find used stream */ + /* + * close stream matching the stream tag + * and check if there are any open capture streams. + */ list_for_each_entry(s, &bus->stream_list, list) { - if (s->direction == direction && - s->opened && s->stream_tag == stream_tag) { + if (!s->opened) + continue; + + if (s->direction == direction && s->stream_tag == stream_tag) { s->opened = false; - spin_unlock_irq(&bus->reg_lock); - return 0; + found = true; + } else if (s->direction == SNDRV_PCM_STREAM_CAPTURE) { + active_capture_stream = true; } } spin_unlock_irq(&bus->reg_lock); - dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); - return -ENODEV; + /* Enable DMI L1 entry if there are no capture streams open */ + if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1)) + if (!active_capture_stream) + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + HDA_VS_INTEL_EM2, + HDA_VS_INTEL_EM2_L1SEN, + HDA_VS_INTEL_EM2_L1SEN); + + if (!found) { + dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); + return -ENODEV; + } + + return 0; } int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 5591841a1b6f..23e430d3e056 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -39,7 +39,6 @@ #define SOF_HDA_WAKESTS 0x0E #define SOF_HDA_WAKESTS_INT_MASK ((1 << 8) - 1) #define SOF_HDA_RIRBSTS 0x5d -#define SOF_HDA_VS_EM2_L1SEN BIT(13) /* SOF_HDA_GCTL register bist */ #define SOF_HDA_GCTL_RESET BIT(0) @@ -228,6 +227,10 @@ #define HDA_DSP_REG_HIPCIE (HDA_DSP_IPC_BASE + 0x0C) #define HDA_DSP_REG_HIPCCTL (HDA_DSP_IPC_BASE + 0x10) +/* Intel Vendor Specific Registers */ +#define HDA_VS_INTEL_EM2 0x1030 +#define HDA_VS_INTEL_EM2_L1SEN BIT(13) + /* HIPCI */ #define HDA_DSP_REG_HIPCI_BUSY BIT(31) #define HDA_DSP_REG_HIPCI_MSG_MASK 0x7FFFFFFF -- cgit v1.2.3 From 4413adc4fd872579de87bedaecda633f999ef995 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 27 Sep 2019 15:14:06 -0500 Subject: ASoC: intel: sof_rt5682: use separate route map for dmic dmic map can only be added when dmic dai link is present. Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927201408.925-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 35 +++++++++++++++++++++++++++++++---- 1 file changed, 31 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index a437567b8cee..57b4ef75be15 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -308,6 +308,9 @@ static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; @@ -318,10 +321,6 @@ static const struct snd_soc_dapm_route sof_map[] = { /* other jacks */ { "IN1P", NULL, "Headset Mic" }, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, - }; static const struct snd_soc_dapm_route speaker_map[] = { @@ -329,6 +328,11 @@ static const struct snd_soc_dapm_route speaker_map[] = { { "Spk", NULL, "Speaker" }, }; +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -342,6 +346,28 @@ static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + /* sof audio machine driver for rt5682 codec */ static struct snd_soc_card sof_audio_card_rt5682 = { .name = "sof_rt5682", @@ -445,6 +471,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].name = "dmic01"; links[id].cpus = &cpus[id]; links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; if (dmic_be_num > 1) { /* set up 2 BE links at most */ links[id + 1].name = "dmic16k"; -- cgit v1.2.3 From a315e76fc544f09daf619530a7b2f85865e6b25e Mon Sep 17 00:00:00 2001 From: Jaska Uimonen Date: Fri, 27 Sep 2019 15:14:07 -0500 Subject: ASoC: rt5682: add NULL handler to set_jack function Implement NULL handler in set_jack function to disable irq's. Signed-off-by: Jaska Uimonen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927201408.925-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 1ef470700ed5..c50b75ce82e0 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -995,6 +995,16 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + rt5682->hs_jack = hs_jack; + + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + return 0; + } + switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2, @@ -1032,8 +1042,6 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, break; } - rt5682->hs_jack = hs_jack; - return 0; } -- cgit v1.2.3 From 6ba5041c23c1062d4e8287b2b76a1181538c6df1 Mon Sep 17 00:00:00 2001 From: Jaska Uimonen Date: Fri, 27 Sep 2019 15:14:08 -0500 Subject: ASoC: intel: sof_rt5682: add remove function to disable jack When removing sof module the rt5682 jack handler will oops if jack detection is not disabled. So add remove function, which disables the jack detection. Signed-off-by: Jaska Uimonen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927201408.925-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 57b4ef75be15..5ce643d62faf 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -648,8 +648,24 @@ static int sof_audio_probe(struct platform_device *pdev) &sof_audio_card_rt5682); } +static int sof_rt5682_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_component *component = NULL; + + for_each_card_components(card, component) { + if (!strcmp(component->name, rt5682_component[0].name)) { + snd_soc_component_set_jack(component, NULL, NULL); + break; + } + } + + return 0; +} + static struct platform_driver sof_audio = { .probe = sof_audio_probe, + .remove = sof_rt5682_remove, .driver = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, -- cgit v1.2.3 From 2bdf194e2030fce4f2e91300817338353414ab3b Mon Sep 17 00:00:00 2001 From: Jaska Uimonen Date: Fri, 27 Sep 2019 15:14:05 -0500 Subject: ASoC: intel: bytcr_rt5651: add null check to support_button_press When removing sof module the support_button_press function will oops because hp_jack pointer is not checked for NULL. So add a check to fix this. Signed-off-by: Jaska Uimonen Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190927201408.925-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5651.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 762595de956c..c506c9305043 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1770,6 +1770,9 @@ static int rt5651_detect_headset(struct snd_soc_component *component) static bool rt5651_support_button_press(struct rt5651_priv *rt5651) { + if (!rt5651->hp_jack) + return false; + /* Button press support only works with internal jack-detection */ return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) && rt5651->gpiod_hp_det == NULL; -- cgit v1.2.3 From 3ae7359c0e39f42a96284d6798fc669acff38140 Mon Sep 17 00:00:00 2001 From: Stuart Henderson Date: Wed, 2 Oct 2019 09:42:40 +0100 Subject: ASoC: wm_adsp: Don't generate kcontrols without READ flags User space always expects to be able to read ALSA controls, so ensure no kcontrols are generated without an appropriate READ flag. In the case of a read of such a control zeros will be returned. Signed-off-by: Stuart Henderson Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20191002084240.21589-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 85396d920e0a..9b8bb7bbe945 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) -- cgit v1.2.3 From b1e620e7d32f5aad5353cc3cfc13ed99fea65d3a Mon Sep 17 00:00:00 2001 From: Robin Murphy Date: Wed, 2 Oct 2019 16:30:37 +0100 Subject: ASoc: rockchip: i2s: Fix RPM imbalance If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait for an absent DMA channel, we return without disabling RPM, which makes subsequent re-probe attempts scream with errors about the unbalanced enable. Don't do that. Fixes: ebb75c0bdba2 ("ASoC: rockchip: i2s: Adjust devm usage") Signed-off-by: Robin Murphy Link: https://lore.kernel.org/r/bcb12a849a05437fb18372bc7536c649b94bdf07.1570029862.git.robin.murphy@arm.com Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af2d5a6124c8..61c984f10d8e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; -- cgit v1.2.3 From 1099f48457d06b816359fb43ac32a4a07e33219b Mon Sep 17 00:00:00 2001 From: Kai-Heng Feng Date: Thu, 3 Oct 2019 12:39:19 +0800 Subject: ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360 Headphone on XPS 9350/9360 produces a background white noise. The The noise level somehow correlates with "Headphone Mic Boost", when it sets to 1 the noise disappears. However, doing this has a side effect, which also decreases the overall headphone volume so I didn't send the patch upstream. The noise was bearable back then, but after commit 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") the noise exacerbates to a point it starts hurting ears. So let's use the workaround to set "Headphone Mic Boost" to 1 and lock it so it's not touchable by userspace. Fixes: 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") BugLink: https://bugs.launchpad.net/bugs/1654448 BugLink: https://bugs.launchpad.net/bugs/1845810 Signed-off-by: Kai-Heng Feng Link: https://lore.kernel.org/r/20191003043919.10960-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..b5c225a56b98 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5822,6 +5833,7 @@ enum { ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -6558,6 +6570,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7001,17 +7019,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), -- cgit v1.2.3 From 130bce3afbbbbe585cba8604f2124c28e8d86fb0 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Mon, 30 Sep 2019 09:29:45 -0500 Subject: ALSA: hdac: clear link output stream mapping Fix potential DMA hang upon starting playback on devices in HDA mode on Intel platforms (Gemini Lake/Whiskey Lake/Comet Lake/Ice Lake). It doesn't affect platforms before Gemini Lake or any Intel device in non-HDA mode. The reset value for the LOSDIV register is all output streams valid. Clear this register to invalidate non-existent streams when the bus is powered up. Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20190930142945.7805-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/sound/hda_register.h | 3 +++ sound/hda/ext/hdac_ext_controller.c | 5 +++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0fd39295b426..057d2a2d0bd0 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -264,6 +264,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define ML_LOSIDV_STREAM_MASK 0xFFFE + #define ML_LCTL_SCF_MASK 0xF #define AZX_MLCTL_SPA (0x1 << 16) #define AZX_MLCTL_CPA (0x1 << 23) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -270,6 +270,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); + /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery -- cgit v1.2.3 From 0632fa042541dbb3b8b960a8cd519eb9b6b584c0 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 5 Oct 2019 23:22:02 +0200 Subject: ASoC: core: Fix pcm code debugfs error We can have 2 dcpm-s with the same backend and frontend name (capture + playback pair), this causes the following debugfs error on Intel Bay Trail systems: [ 298.969049] debugfs: Directory 'SSP2-Codec' with parent 'Baytrail Audio Port' already present! This commit adds a ":playback" or ":capture" postfix to the debugfs dir name fixing this. Signed-off-by: Hans de Goede Link: https://lore.kernel.org/r/20191005212202.5206-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..b600d3eaaf5c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; + char *name; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state); + name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir(name, + fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } #endif return 1; } -- cgit v1.2.3 From bcab05880f9306e94531b0009c627421db110a74 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Wed, 9 Oct 2019 12:19:44 +0100 Subject: ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2 This patch adds missing MIX2 path on RX1/2 which take IIR1 and IIR2 as inputs. Without this patch sound card fails to intialize with below warning: ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1 ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1 Reported-by: Stephan Gerhold Signed-off-by: Srinivas Kandagatla Tested-by: Stephan Gerhold Link: https://lore.kernel.org/r/20191009111944.28069-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..58b2468fb2a7 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = { "ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3" }; +static const char * const rx_mix2_text[] = { + "ZERO", "IIR1", "IIR2" +}; + static const char *const dec_mux_text[] = { "ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2" }; @@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; +/* RX1 MIX2 */ +static const struct soc_enum rx_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL, + 0, 3, rx_mix2_text); + +/* RX2 MIX2 */ +static const struct soc_enum rx2_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL, + 0, 3, rx_mix2_text); + /* DEC */ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE( LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text); @@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]); static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]); +static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum); +static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); /* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); @@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { &rx3_mix1_inp2_mux), SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0, &rx3_mix1_inp3_mux), + SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx1_mix2_inp1_mux), + SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx2_mix2_inp1_mux), SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux), SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux), -- cgit v1.2.3 From 7571b6a17fcc5e4f6903f065a82d0e38011346ed Mon Sep 17 00:00:00 2001 From: Szabolcs Szőke Date: Fri, 11 Oct 2019 19:19:36 +0200 Subject: ALSA: usb-audio: Disable quirks for BOSS Katana amplifiers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit BOSS Katana amplifiers cannot be used for recording or playback if quirks are applied BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195223 Signed-off-by: Szabolcs Szőke Cc: Link: https://lore.kernel.org/r/20191011171937.8013-1-szszoke.code@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 33cd26763c0e..ff5ab24f3bd1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ + /* BOSS Katana amplifiers do not need quirks */ + return 0; } if (attr == USB_ENDPOINT_SYNC_ASYNC && -- cgit v1.2.3 From 8c8967a7dc01a25f57a0757fdca10987773cd1f2 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 17 Oct 2019 16:15:01 +0800 Subject: ALSA: hda/realtek - Enable headset mic on Asus MJ401TA On Asus MJ401TA (with Realtek ALC256), the headset mic is connected to pin 0x19, with default configuration value 0x411111f0 (indicating no physical connection). Enable this by quirking the pin. Mic jack detection was also tested and found to be working. This enables use of the headset mic on this product. Signed-off-by: Daniel Drake Cc: Link: https://lore.kernel.org/r/20191017081501.17135-1-drake@endlessm.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b5c225a56b98..ce4f11659765 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5881,6 +5881,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, @@ -6930,6 +6931,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7126,6 +7136,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- cgit v1.2.3 From 94989e318b2f11e217e86bee058088064fa9a2e9 Mon Sep 17 00:00:00 2001 From: Lukas Wunner Date: Thu, 17 Oct 2019 17:04:11 +0200 Subject: ALSA: hda - Force runtime PM on Nvidia HDMI codecs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Przemysław Kopa reports that since commit b516ea586d71 ("PCI: Enable NVIDIA HDA controllers"), the discrete GPU Nvidia GeForce GT 540M on his 2011 Samsung laptop refuses to runtime suspend, resulting in a power regression and excessive heat. Rivera Valdez witnesses the same issue with a GeForce GT 525M (GF108M) of the same era, as does another Arch Linux user named "R0AR" with a more recent GeForce GTX 1050 Ti (GP107M). The commit exposes the discrete GPU's HDA controller and all four codecs on the controller do not set the CLKSTOP and EPSS bits in the Supported Power States Response. They also do not set the PS-ClkStopOk bit in the Get Power State Response. hda_codec_runtime_suspend() therefore does not call snd_hdac_codec_link_down(), which prevents each codec and the PCI device from runtime suspending. The same issue is present on some AMD discrete GPUs and we addressed it by forcing runtime PM despite the bits not being set, see commit 57cb54e53bdd ("ALSA: hda - Force to link down at runtime suspend on ATI/AMD HDMI"). Do the same for Nvidia HDMI codecs. Fixes: b516ea586d71 ("PCI: Enable NVIDIA HDA controllers") Link: https://bbs.archlinux.org/viewtopic.php?pid=1865512 Link: https://bugs.freedesktop.org/show_bug.cgi?id=75985#c81 Reported-by: Przemysław Kopa Reported-by: Rivera Valdez Signed-off-by: Lukas Wunner Cc: Daniel Drake Cc: stable@vger.kernel.org # v5.3+ Link: https://lore.kernel.org/r/3086bc75135c1e3567c5bc4f3cc4ff5cbf7a56c2.1571324194.git.lukas@wunner.de Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bca5de78e9ad..795cbda32cbb 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); return 0; -- cgit v1.2.3 From 22e58665a01006d05f0239621f7d41cacca96cc4 Mon Sep 17 00:00:00 2001 From: Junya Monden Date: Wed, 16 Oct 2019 14:42:55 +0200 Subject: ASoC: rsnd: Reinitialize bit clock inversion flag for every format setting Unlike other format-related DAI parameters, rdai->bit_clk_inv flag is not properly re-initialized when setting format for new stream processing. The inversion, if requested, is then applied not to default, but to a previous value, which leads to SCKP bit in SSICR register being set incorrectly. Fix this by re-setting the flag to its initial value, determined by format. Fixes: 1a7889ca8aba3 ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior") Cc: Andrew Gabbasov Cc: Jiada Wang Cc: Timo Wischer Cc: stable@vger.kernel.org # v3.17+ Signed-off-by: Junya Monden Signed-off-by: Eugeniu Rosca Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/20191016124255.7442-1-erosca@de.adit-jv.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index bda5b958d0dc..e9596c2096cd 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -761,6 +761,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* set format */ + rdai->bit_clk_inv = 0; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: rdai->sys_delay = 0; -- cgit v1.2.3 From e5f0d490fb718254a884453e47fcd48493cd67ea Mon Sep 17 00:00:00 2001 From: Chuhong Yuan Date: Thu, 17 Oct 2019 10:50:44 +0800 Subject: ASoC: Intel: sof-rt5682: add a check for devm_clk_get sof_audio_probe misses a check for devm_clk_get and may cause problems. Add a check for it to fix the bug. Signed-off-by: Chuhong Yuan Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191017025044.31474-1-hslester96@gmail.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 5ce643d62faf..4f6e58c3954a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -603,6 +603,15 @@ static int sof_audio_probe(struct platform_device *pdev) /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + + dev_err(&pdev->dev, + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret); + return ret; + } + ret = clk_prepare_enable(ctx->mclk); if (ret < 0) { dev_err(&pdev->dev, -- cgit v1.2.3 From 9b7a7f921689d6c254e5acd670be631ebd82d54d Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Fri, 18 Oct 2019 10:20:40 +0200 Subject: ASoC: stm32: sai: fix sysclk management on shutdown The commit below, adds a call to sysclk callback on shutdown. This introduces a regression in stm32 SAI driver, as some clock services are called twice, leading to unbalanced calls. Move processing related to mclk from shutdown to sysclk callback. When requested frequency is 0, assume shutdown and release mclk. Fixes: 2458adb8f92a ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown") Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20191018082040.31022-1-olivier.moysan@st.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_sai_sub.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index d7501f88aaa6..a4060813bc74 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -505,10 +505,20 @@ static int stm32_sai_set_sysclk(struct snd_soc_dai *cpu_dai, if (dir == SND_SOC_CLOCK_OUT && sai->sai_mclk) { ret = regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - (unsigned int)~SAI_XCR1_NODIV); + freq ? 0 : SAI_XCR1_NODIV); if (ret < 0) return ret; + /* Assume shutdown if requested frequency is 0Hz */ + if (!freq) { + /* Release mclk rate only if rate was actually set */ + if (sai->mclk_rate) { + clk_rate_exclusive_put(sai->sai_mclk); + sai->mclk_rate = 0; + } + return 0; + } + /* If master clock is used, set parent clock now */ ret = stm32_sai_set_parent_clock(sai, freq); if (ret) @@ -1093,15 +1103,6 @@ static void stm32_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, STM_SAI_IMR_REGX, SAI_XIMR_MASK, 0); - regmap_update_bits(sai->regmap, STM_SAI_CR1_REGX, SAI_XCR1_NODIV, - SAI_XCR1_NODIV); - - /* Release mclk rate only if rate was actually set */ - if (sai->mclk_rate) { - clk_rate_exclusive_put(sai->sai_mclk); - sai->mclk_rate = 0; - } - clk_disable_unprepare(sai->sai_ck); spin_lock_irqsave(&sai->irq_lock, flags); -- cgit v1.2.3 From 95a32c98055f664f9b3f34c41e153d4dcedd0eff Mon Sep 17 00:00:00 2001 From: Dragos Tarcatu Date: Fri, 18 Oct 2019 07:38:06 -0500 Subject: ASoC: SOF: control: return true when kcontrol values change All the kcontrol put() functions are currently returning 0 when successful. This does not go well with alsamixer as it does not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks when values change for (some of) the sof kcontrols. This patch fixes that by returning true for volume, switch and enum type kcontrols when values do change in put(). Signed-off-by: Dragos Tarcatu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191018123806.18063-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/control.c | 26 ++++++++++++++++++-------- 1 file changed, 18 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index a4983f90ff5b..2b8711eda362 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -60,13 +60,16 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = - mixer_to_ipc(ucontrol->value.integer.value[i], + value = mixer_to_ipc(ucontrol->value.integer.value[i], scontrol->volume_table, sm->max + 1); + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -76,8 +79,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, SOF_CTRL_TYPE_VALUE_CHAN_GET, SOF_CTRL_CMD_VOLUME, true); - - return 0; + return change; } int snd_sof_switch_get(struct snd_kcontrol *kcontrol, @@ -105,11 +107,15 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.integer.value[i]; + value = ucontrol->value.integer.value[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of mixer updates */ @@ -120,7 +126,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_SWITCH, true); - return 0; + return change; } int snd_sof_enum_get(struct snd_kcontrol *kcontrol, @@ -148,11 +154,15 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, struct snd_sof_dev *sdev = scontrol->sdev; struct sof_ipc_ctrl_data *cdata = scontrol->control_data; unsigned int i, channels = scontrol->num_channels; + bool change = false; + u32 value; /* update each channel */ for (i = 0; i < channels; i++) { - cdata->chanv[i].value = ucontrol->value.enumerated.item[i]; + value = ucontrol->value.enumerated.item[i]; + change = change || (value != cdata->chanv[i].value); cdata->chanv[i].channel = i; + cdata->chanv[i].value = value; } /* notify DSP of enum updates */ @@ -163,7 +173,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, SOF_CTRL_CMD_ENUM, true); - return 0; + return change; } int snd_sof_bytes_get(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From 83629532ce45ef9df1f297b419b9ea112045685d Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 2 May 2019 16:03:26 +0800 Subject: ALSA: hda/realtek - Add support for ALC711 Support new codec ALC711. Signed-off-by: Kailang Yang Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce4f11659765..085a2f95e076 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -393,6 +393,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: alc_update_coef_idx(codec, 0x10, 1<<15, 0); break; case 0x10ec0662: @@ -8019,6 +8020,7 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: + case 0x10ec0711: spec->codec_variant = ALC269_TYPE_ALC700; spec->gen.mixer_nid = 0; /* ALC700 does not have any loopback mixer path */ alc_update_coef_idx(codec, 0x4a, 1 << 15, 0); /* Combo jack auto trigger control */ @@ -9233,6 +9235,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269), HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269), HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0711, "ALC711", patch_alc269), HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662), HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880), HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882), -- cgit v1.2.3 From ba8bf0967a154796be15c4983603aad0b05c3138 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Oct 2019 17:45:14 +0200 Subject: ALSA: usb-audio: Fix copy&paste error in the validator The recently introduced USB-audio descriptor validator had a stupid copy&paste error that may lead to an unexpected overlook of too short descriptors for processing and extension units. It's likely the cause of the report triggered by syzkaller fuzzer. Let's fix it. Fixes: 57f8770620e9 ("ALSA: usb-audio: More validations of descriptor units") Reported-by: syzbot+0620f79a1978b1133fd7@syzkaller.appspotmail.com Link: https://lore.kernel.org/r/s5hsgnkdbsl.wl-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/validate.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/validate.c b/sound/usb/validate.c index 3c8f73a0eb12..a5e584b60dcd 100644 --- a/sound/usb/validate.c +++ b/sound/usb/validate.c @@ -75,7 +75,7 @@ static bool validate_processing_unit(const void *p, if (d->bLength < sizeof(*d)) return false; - len = d->bLength < sizeof(*d) + d->bNrInPins; + len = sizeof(*d) + d->bNrInPins; if (d->bLength < len) return false; switch (v->protocol) { -- cgit v1.2.3 From 4750c212174892d26645cdf5ad73fb0e9d594ed3 Mon Sep 17 00:00:00 2001 From: Pan Xiuli Date: Tue, 22 Oct 2019 14:44:02 -0500 Subject: ALSA: hda: Add Tigerlake/Jasperlake PCI ID Add HD Audio Device PCI ID for the Intel Tigerlake and Jasperlake platform. Signed-off-by: Pan Xiuli Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20191022194402.23178-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a815bc811799 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2399,6 +2399,12 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Jasperlake */ + { PCI_DEVICE(0x8086, 0x38c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake */ + { PCI_DEVICE(0x8086, 0xa0c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, -- cgit v1.2.3 From e2995b95a914bbc6b5352be27d5d5f33ec802d2c Mon Sep 17 00:00:00 2001 From: Justin Song Date: Thu, 24 Oct 2019 12:27:14 +0200 Subject: ALSA: usb-audio: Add DSD support for Gustard U16/X26 USB Interface This patch adds native DSD support for Gustard U16/X26 USB Interface. Tested using VID and fp->dsd_raw method. Signed-off-by: Justin Song Cc: Link: https://lore.kernel.org/r/CA+9XP1ipsFn+r3bCBKRinQv-JrJ+EHOGBdZWZoMwxFv0R8Y1MQ@mail.gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fbfde996fee7..0bbe1201a6ac 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1657,6 +1657,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x23ba: /* Playback Designs */ case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ + case 0x292b: /* Gustard/Ess based devices */ case 0x2ab6: /* T+A devices */ case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ -- cgit v1.2.3 From f0778871a13889b86a65d4ad34bef8340af9d082 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 24 Oct 2019 15:13:32 +0800 Subject: ALSA: hda/realtek - Add support for ALC623 Support new codec ALC623. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/r/ed97b6a8bd9445ecb48bc763d9aaba7a@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 085a2f95e076..a0c237cc13d4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -409,6 +409,9 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0672: alc_update_coef_idx(codec, 0xd, 0, 1<<14); /* EAPD Ctrl */ break; + case 0x10ec0623: + alc_update_coef_idx(codec, 0x19, 1<<13, 0); + break; case 0x10ec0668: alc_update_coef_idx(codec, 0x7, 3<<13, 0); break; @@ -2920,6 +2923,7 @@ enum { ALC269_TYPE_ALC225, ALC269_TYPE_ALC294, ALC269_TYPE_ALC300, + ALC269_TYPE_ALC623, ALC269_TYPE_ALC700, }; @@ -2955,6 +2959,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC225: case ALC269_TYPE_ALC294: case ALC269_TYPE_ALC300: + case ALC269_TYPE_ALC623: case ALC269_TYPE_ALC700: ssids = alc269_ssids; break; @@ -8017,6 +8022,9 @@ static int patch_alc269(struct hda_codec *codec) spec->codec_variant = ALC269_TYPE_ALC300; spec->gen.mixer_nid = 0; /* no loopback on ALC300 */ break; + case 0x10ec0623: + spec->codec_variant = ALC269_TYPE_ALC623; + break; case 0x10ec0700: case 0x10ec0701: case 0x10ec0703: @@ -9218,6 +9226,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269), HDA_CODEC_ENTRY(0x10ec0300, "ALC300", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0623, "ALC623", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861), -- cgit v1.2.3 From 8a6c55d0f883e9a7e7c91841434f3b6bbf932bb2 Mon Sep 17 00:00:00 2001 From: Aaron Ma Date: Thu, 24 Oct 2019 19:44:39 +0800 Subject: ALSA: hda/realtek - Fix 2 front mics of codec 0x623 These 2 ThinkCentres installed a new realtek codec ID 0x623, it has 2 front mics with the same location on pin 0x18 and 0x19. Apply fixup ALC283_FIXUP_HEADSET_MIC to change 1 front mic location to right, then pulseaudio can handle them. One "Front Mic" and one "Mic" will be shown, and audio output works fine. Signed-off-by: Aaron Ma Cc: Link: https://lore.kernel.org/r/20191024114439.31522-1-aaron.ma@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0c237cc13d4..80f66ba85f87 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7221,6 +7221,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- cgit v1.2.3 From f2bbdbcb075f3977a53da3bdcb7cd460bc8ae5f2 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sat, 26 Oct 2019 12:06:20 +0900 Subject: ALSA: bebob: Fix prototype of helper function to return negative value A helper function of ALSA bebob driver returns negative value in a function which has a prototype to return unsigned value. This commit fixes it by changing the prototype. Fixes: eb7b3a056cd8 ("ALSA: bebob: Add commands and connections/streams management") Cc: # v3.16+ Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20191026030620.12077-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/bebob/bebob_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 73fee991bd75..6c1497d9f52b 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ end: return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; -- cgit v1.2.3 From 1a7f60b9df614bb36d14dc0c0bc898a31b2b506f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 28 Oct 2019 09:10:56 +0100 Subject: Revert "ALSA: hda: Flush interrupts on disabling" This reverts commit caa8422d01e983782548648e125fd617cadcec3f. It turned out that this commit caused a regression at shutdown / reboot, as the synchronize_irq() calls seems blocking the whole shutdown. Also another part of the change about shuffling the call order looks suspicious; the azx_stop_chip() call disables the CORB / RIRB while the others may still need the CORB/RIRB update. Since the original commit itself was a cargo-fix, let's revert the whole patch. Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205333 BugLinK: https://bugs.freedesktop.org/show_bug.cgi?id=111174 Signed-off-by: Takashi Iwai Cc: Chris Wilson Link: https://lore.kernel.org/r/20191028081056.22010-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_controller.c | 2 -- sound/pci/hda/hda_intel.c | 2 +- 2 files changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index d3999e7b0705..7e7be8e4dcf9 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -447,8 +447,6 @@ static void azx_int_disable(struct hdac_bus *bus) list_for_each_entry(azx_dev, &bus->stream_list, list) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_INT_MASK, 0); - synchronize_irq(bus->irq); - /* disable SIE for all streams */ snd_hdac_chip_writeb(bus, INTCTL, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a815bc811799..cf53fbd872ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1348,9 +1348,9 @@ static int azx_free(struct azx *chip) } if (bus->chip_init) { - azx_stop_chip(chip); azx_clear_irq_pending(chip); azx_stop_all_streams(chip); + azx_stop_chip(chip); } if (bus->irq >= 0) -- cgit v1.2.3