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[PATCH] ASoC: codecs: lpass-rx-macro: Remove unneeded semicolon
by Wan Jiabing 11 May '21

11 May '21

Fix the following coccicheck warning: ./sound/soc/codecs/lpass-rx-macro.c:2631:2-3: Unneeded semicolon Signed-off-by: Wan Jiabing <wanjiabing(a)vivo.com> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index b0ebfc8d180c..8dff72fc7bab 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2628,7 +2628,7 @@ static int rx_macro_enable_rx_path_clk(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } -- 2.25.1

2 1

[PATCH] ASoc: Fix unused define in jz4740-i2s.h
by Tang Bin 11 May '21

11 May '21

Delete unused define of JZ4740_I2S_BIT_CLK, because it is unused in any files. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/jz4740/jz4740-i2s.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.h b/sound/soc/jz4740/jz4740-i2s.h index 44f120160..4da14eac1 100644 --- a/sound/soc/jz4740/jz4740-i2s.h +++ b/sound/soc/jz4740/jz4740-i2s.h @@ -7,6 +7,4 @@ #define JZ4740_I2S_CLKSRC_EXT 0 #define JZ4740_I2S_CLKSRC_PLL 1 -#define JZ4740_I2S_BIT_CLK 0 - #endif -- 2.20.1.windows.1

2 1

[PATCH v2] ASoC: codecs: lpass-wsa-macro: handle unexpected input
by trix＠redhat.com 11 May '21

11 May '21

From: Tom Rix <trix(a)redhat.com> Static analysis reports this problem lpass-wsa-macro.c:1732:6: warning: Array subscript is undefined if (wsa->ec_hq[ec_tx]) { ^~~~~~~~~~~~~~~~~ The happens because 'ec_tx' is never initialized and there is no default in switch statement that sets ec_tx. Add a default case that returns an error before the array is accessed. Signed-off-by: Tom Rix <trix(a)redhat.com> --- v2: - handle the input error, keep switch --- sound/soc/codecs/lpass-wsa-macro.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 1a7fa5492f28..d3ac318fd6b6 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1727,6 +1727,10 @@ static int wsa_macro_enable_echo(struct snd_soc_dapm_widget *w, val = val & CDC_WSA_RX_MIX_TX1_SEL_MASK; ec_tx = (val >> CDC_WSA_RX_MIX_TX1_SEL_SHFT) - 1; break; + default: + dev_err(component->dev, "%s: Invalid shift %u\n", + __func__, w->shift); + return -EINVAL; } if (wsa->ec_hq[ec_tx]) { -- 2.26.3

3 2

[PATCH] ASoC: fsl_spdif: add support for enabling raw capture mode
by Shengjiu Wang 11 May '21

11 May '21

From: Viorel Suman <viorel.suman(a)nxp.com> Since i.MX8MM SPDIF interface is able to capture raw data. Add support in SPDIF driver for this functionality. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 67 +++++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_spdif.h | 1 + 2 files changed, 68 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index c631de325a6e..2a76714eb8e6 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -49,6 +49,7 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; * @imx: for imx platform * @shared_root_clock: flag of sharing a clock source with others; * so the driver shouldn't set root clock rate + * @raw_capture_mode: if raw capture mode support * @interrupts: interrupt number * @tx_burst: tx maxburst size * @rx_burst: rx maxburst size @@ -57,6 +58,7 @@ static u8 srpc_dpll_locked[] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0xa, 0xb }; struct fsl_spdif_soc_data { bool imx; bool shared_root_clock; + bool raw_capture_mode; u32 interrupts; u32 tx_burst; u32 rx_burst; @@ -136,6 +138,7 @@ struct fsl_spdif_priv { static struct fsl_spdif_soc_data fsl_spdif_vf610 = { .imx = false, .shared_root_clock = false, + .raw_capture_mode = false, .interrupts = 1, .tx_burst = FSL_SPDIF_TXFIFO_WML, .rx_burst = FSL_SPDIF_RXFIFO_WML, @@ -145,6 +148,7 @@ static struct fsl_spdif_soc_data fsl_spdif_vf610 = { static struct fsl_spdif_soc_data fsl_spdif_imx35 = { .imx = true, .shared_root_clock = false, + .raw_capture_mode = false, .interrupts = 1, .tx_burst = FSL_SPDIF_TXFIFO_WML, .rx_burst = FSL_SPDIF_RXFIFO_WML, @@ -154,6 +158,7 @@ static struct fsl_spdif_soc_data fsl_spdif_imx35 = { static struct fsl_spdif_soc_data fsl_spdif_imx6sx = { .imx = true, .shared_root_clock = true, + .raw_capture_mode = false, .interrupts = 1, .tx_burst = FSL_SPDIF_TXFIFO_WML, .rx_burst = FSL_SPDIF_RXFIFO_WML, @@ -164,12 +169,23 @@ static struct fsl_spdif_soc_data fsl_spdif_imx6sx = { static struct fsl_spdif_soc_data fsl_spdif_imx8qm = { .imx = true, .shared_root_clock = true, + .raw_capture_mode = false, .interrupts = 2, .tx_burst = 2, /* Applied for EDMA */ .rx_burst = 2, /* Applied for EDMA */ .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ }; +static struct fsl_spdif_soc_data fsl_spdif_imx8mm = { + .imx = true, + .shared_root_clock = false, + .raw_capture_mode = true, + .interrupts = 1, + .tx_burst = FSL_SPDIF_TXFIFO_WML, + .rx_burst = FSL_SPDIF_RXFIFO_WML, + .tx_formats = FSL_SPDIF_FORMATS_PLAYBACK, +}; + /* Check if clk is a root clock that does not share clock source with others */ static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk) { @@ -846,6 +862,39 @@ static int fsl_spdif_tx_vbit_put(struct snd_kcontrol *kcontrol, return 0; } +static int fsl_spdif_rx_rcm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val; + + regmap_read(regmap, REG_SPDIF_SCR, &val); + val = (val & SCR_RAW_CAPTURE_MODE) ? 1 : 0; + ucontrol->value.integer.value[0] = val; + + return 0; +} + +static int fsl_spdif_rx_rcm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regmap = spdif_priv->regmap; + u32 val = (ucontrol->value.integer.value[0] ? SCR_RAW_CAPTURE_MODE : 0); + + if (val) + cpu_dai->driver->capture.formats |= SNDRV_PCM_FMTBIT_S32_LE; + else + cpu_dai->driver->capture.formats &= ~SNDRV_PCM_FMTBIT_S32_LE; + + regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_RAW_CAPTURE_MODE, val); + + return 0; +} + /* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1029,6 +1078,19 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { }, }; +static struct snd_kcontrol_new fsl_spdif_ctrls_rcm[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Raw Capture Mode", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ctl_boolean_mono_info, + .get = fsl_spdif_rx_rcm_get, + .put = fsl_spdif_rx_rcm_put, + }, +}; + static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); @@ -1038,6 +1100,10 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); + if (spdif_private->soc->raw_capture_mode) + snd_soc_add_dai_controls(dai, fsl_spdif_ctrls_rcm, + ARRAY_SIZE(fsl_spdif_ctrls_rcm)); + /*Clear the val bit for Tx*/ regmap_update_bits(spdif_private->regmap, REG_SPDIF_SCR, SCR_VAL_MASK, SCR_VAL_CLEAR); @@ -1476,6 +1542,7 @@ static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,vf610-spdif", .data = &fsl_spdif_vf610, }, { .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, }, { .compatible = "fsl,imx8qm-spdif", .data = &fsl_spdif_imx8qm, }, + { .compatible = "fsl,imx8mm-spdif", .data = &fsl_spdif_imx8mm, }, {} }; MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index d5f1dfd58740..bff8290e71f2 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -63,6 +63,7 @@ #define SCR_TXFIFO_FSEL_IF4 (0x1 << SCR_TXFIFO_FSEL_OFFSET) #define SCR_TXFIFO_FSEL_IF8 (0x2 << SCR_TXFIFO_FSEL_OFFSET) #define SCR_TXFIFO_FSEL_IF12 (0x3 << SCR_TXFIFO_FSEL_OFFSET) +#define SCR_RAW_CAPTURE_MODE BIT(14) #define SCR_LOW_POWER (1 << 13) #define SCR_SOFT_RESET (1 << 12) #define SCR_TXFIFO_CTRL_OFFSET 10 -- 2.27.0

2 1

[PATCH] ASoC: amd: renoir: Remove redundant assignment to pdm_ctrl and pdm_enable and pdm_dma_enable
by Jiapeng Chong 11 May '21

11 May '21

Variable pdm_ctrl and pdm_enable and pdm_dma_enable are set to '0x00', but they are overwritten later on, so these are redundant assignments that can be removed. Clean up the following clang-analyzer warning: sound/soc/amd/renoir/acp3x-pdm-dma.c:148:2: warning: Value stored to 'pdm_dma_enable' is never read [clang-analyzer-deadcode.DeadStores]. sound/soc/amd/renoir/acp3x-pdm-dma.c:147:2: warning: Value stored to 'pdm_enable' is never read [clang-analyzer-deadcode.DeadStores]. sound/soc/amd/renoir/acp3x-pdm-dma.c:80:2: warning: Value stored to 'pdm_ctrl' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 4c2810e..bd20622b 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -77,7 +77,6 @@ static void enable_pdm_clock(void __iomem *acp_base) u32 pdm_clk_enable, pdm_ctrl; pdm_clk_enable = ACP_PDM_CLK_FREQ_MASK; - pdm_ctrl = 0x00; rn_writel(pdm_clk_enable, acp_base + ACP_WOV_CLK_CTRL); pdm_ctrl = rn_readl(acp_base + ACP_WOV_MISC_CTRL); @@ -144,9 +143,6 @@ static int stop_pdm_dma(void __iomem *acp_base) u32 pdm_enable, pdm_dma_enable; int timeout; - pdm_enable = 0x00; - pdm_dma_enable = 0x00; - pdm_enable = rn_readl(acp_base + ACP_WOV_PDM_ENABLE); pdm_dma_enable = rn_readl(acp_base + ACP_WOV_PDM_DMA_ENABLE); if (pdm_dma_enable & 0x01) { -- 1.8.3.1

2 1

[PATCH] ASoC: rt1019: Add non_legacy_dai_naming config
by derek.fang＠realtek.com 11 May '21

11 May '21

From: Derek Fang <derek.fang(a)realtek.com> Register the codec dai name as 'rt1019-aif' by adding non_legacy_dai_naming configuration. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index 10656a5927f1..2687ae60fd36 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -522,6 +522,7 @@ static const struct snd_soc_component_driver soc_component_dev_rt1019 = { .num_dapm_widgets = ARRAY_SIZE(rt1019_dapm_widgets), .dapm_routes = rt1019_dapm_routes, .num_dapm_routes = ARRAY_SIZE(rt1019_dapm_routes), + .non_legacy_dai_naming = 1, }; static const struct regmap_config rt1019_regmap = { -- 2.17.1

2 1

[PATCH] ASoC: jz4740-i2s: fix function name
by H. Nikolaus Schaller 11 May '21

11 May '21

This driver is not related to I2C protocol. s/_i2c_/_i2s_/ Signed-off-by: H. Nikolaus Schaller <hns(a)goldelico.com> --- sound/soc/jz4740/jz4740-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 47d955c0bb6a1..fe5b3a2b239c6 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -372,7 +372,7 @@ static int jz4740_i2s_resume(struct snd_soc_component *component) return 0; } -static void jz4740_i2c_init_pcm_config(struct jz4740_i2s *i2s) +static void jz4740_i2s_init_pcm_config(struct jz4740_i2s *i2s) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -397,7 +397,7 @@ static int jz4740_i2s_dai_probe(struct snd_soc_dai *dai) if (ret) return ret; - jz4740_i2c_init_pcm_config(i2s); + jz4740_i2s_init_pcm_config(i2s); snd_soc_dai_init_dma_data(dai, &i2s->playback_dma_data, &i2s->capture_dma_data); -- 2.26.2

3 2

[PATCH 0/4] ASoC: cppcheck fixes of the week
by Pierre-Louis Bossart 11 May '21

11 May '21

While running some checks on a rebased branch, I realized I missed a couple of trivial cases on newer code. Pierre-Louis Bossart (4): ASoC: codecs: mt6359-accdet: remove useless initialization ASoc: codecs: mt6359: remove useless initializations ASoC: codecs: rt1019: clarify expression ASoC: fsl: imx-pcm-rpmsg: remove useless initialization sound/soc/codecs/mt6359-accdet.c | 2 +- sound/soc/codecs/mt6359.c | 2 +- sound/soc/codecs/rt1019.c | 4 ++-- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: rt711-sdca: fix the function number of SDCA control for feature unit 0x1E
by shumingf＠realtek.com 11 May '21

11 May '21

From: Shuming Fan <shumingf(a)realtek.com> The function number should be FUNC_NUM_MIC_ARRAY(0x2) for the feature unit 0x1E. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index cc36739f7fcf..24a084e0b48a 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -683,13 +683,13 @@ static int rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) ch_r = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_r_mute) ? 0x01 : 0x00; err = regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); if (err < 0) return err; err = regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); if (err < 0) return err; -- 2.29.0

3 2

[PATCH 1/3] ASoC: cs42l42: Add support for set_jack calls
by Lucas Tanure 11 May '21

11 May '21

Replace the internal jack creation by set_jack call, so users can map buttons in their machine driver Also only enable jack detection IRQ after set_jack call Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 43 ++++++++++++++++++++++---------------- sound/soc/codecs/cs42l42.h | 2 +- 2 files changed, 26 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index bf982e145e94..2143957b95e6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -518,26 +518,33 @@ static const struct snd_soc_dapm_route cs42l42_audio_map[] = { { "SDOUT2", NULL, "ASP TX EN" }, }; +static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_jack *jk, void *d) +{ + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + + cs42l42->jack = jk; + + regmap_update_bits(cs42l42->regmap, CS42L42_TSRS_PLUG_INT_MASK, + CS42L42_RS_PLUG_MASK | CS42L42_RS_UNPLUG_MASK | + CS42L42_TS_PLUG_MASK | CS42L42_TS_UNPLUG_MASK, + (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | + (0 << CS42L42_TS_PLUG_SHIFT) | (0 << CS42L42_TS_UNPLUG_SHIFT)); + + return 0; +} + static int cs42l42_component_probe(struct snd_soc_component *component) { - struct cs42l42_private *cs42l42 = - (struct cs42l42_private *)snd_soc_component_get_drvdata(component); - struct snd_soc_card *crd = component->card; - int ret = 0; + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); cs42l42->component = component; - ret = snd_soc_card_jack_new(crd, "CS42L42 Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &cs42l42->jack, NULL, 0); - if (ret < 0) - dev_err(component->dev, "Cannot create CS42L42 Headset: %d\n", ret); - - return ret; + return 0; } static const struct snd_soc_component_driver soc_component_dev_cs42l42 = { .probe = cs42l42_component_probe, + .set_jack = cs42l42_set_jack, .dapm_widgets = cs42l42_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs42l42_dapm_widgets), .dapm_routes = cs42l42_audio_map, @@ -1410,11 +1417,11 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) switch(cs42l42->hs_type){ case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: - snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADSET, + snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); break; case CS42L42_PLUG_HEADPHONE: - snd_soc_jack_report(&cs42l42->jack, SND_JACK_HEADPHONE, + snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); break; default: @@ -1442,10 +1449,10 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) switch(cs42l42->hs_type){ case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: - snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADSET); + snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADSET); break; case CS42L42_PLUG_HEADPHONE: - snd_soc_jack_report(&cs42l42->jack, 0, SND_JACK_HEADPHONE); + snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADPHONE); break; default: break; @@ -1472,7 +1479,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) report = cs42l42_handle_button_press(cs42l42); } - snd_soc_jack_report(&cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | + snd_soc_jack_report(cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); } } @@ -1579,8 +1586,8 @@ static void cs42l42_set_interrupt_masks(struct cs42l42_private *cs42l42) CS42L42_TS_UNPLUG_MASK, (1 << CS42L42_RS_PLUG_SHIFT) | (1 << CS42L42_RS_UNPLUG_SHIFT) | - (0 << CS42L42_TS_PLUG_SHIFT) | - (0 << CS42L42_TS_UNPLUG_SHIFT)); + (1 << CS42L42_TS_PLUG_SHIFT) | + (1 << CS42L42_TS_UNPLUG_SHIFT)); } static void cs42l42_setup_hs_type_detect(struct cs42l42_private *cs42l42) diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 36b763f0d1a0..2e0d3836bd7e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -773,7 +773,7 @@ struct cs42l42_private { struct regulator_bulk_data supplies[CS42L42_NUM_SUPPLIES]; struct gpio_desc *reset_gpio; struct completion pdn_done; - struct snd_soc_jack jack; + struct snd_soc_jack *jack; int bclk; u32 sclk; u32 srate; -- 2.31.1

2 3

[PATCH] ASoC: rt286: Remove redundant assignment to d_len_code
by Jiapeng Chong 11 May '21

11 May '21

Variable d_len_code is set to zero, but this value is never read as it is overwritten or not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/soc/codecs/rt286.c:728:2: warning: Value stored to 'd_len_code' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rt286.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 802f485..caa7208 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -725,7 +725,6 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - d_len_code = 0; switch (params_width(params)) { /* bit 6:4 Bits per Sample */ case 16: -- 1.8.3.1

2 1

[RESEND PATCH] soundwire: add slave device to linked list after device_register()
by Bard Liao 11 May '21

11 May '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We currently add the slave device to a linked list before device_register(), and then remove it if device_register() fails. It's not clear why this sequence was necessary, this patch moves the linked list management to after the device_register(). Suggested-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/slave.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 0eed38a79c6d..ff01b96c6e56 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -65,9 +65,6 @@ int sdw_slave_add(struct sdw_bus *bus, for (i = 0; i < SDW_MAX_PORTS; i++) init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); - list_add_tail(&slave->node, &bus->slaves); - mutex_unlock(&bus->bus_lock); ret = device_register(&slave->dev); if (ret) { @@ -77,13 +74,15 @@ int sdw_slave_add(struct sdw_bus *bus, * On err, don't free but drop ref as this will be freed * when release method is invoked. */ - mutex_lock(&bus->bus_lock); - list_del(&slave->node); - mutex_unlock(&bus->bus_lock); put_device(&slave->dev); return ret; } + + mutex_lock(&bus->bus_lock); + list_add_tail(&slave->node, &bus->slaves); + mutex_unlock(&bus->bus_lock); + sdw_slave_debugfs_init(slave); return ret; -- 2.17.1

1 0

Re: broonie-sound/for-next bisection: baseline.bootrr.asoc-simple-card-probed on kontron-sl28-var3-ads2
by Guillaume Tucker 11 May '21

11 May '21

Hi Morimoto-san, Please see the bisection report below about the asoc-simple-card driver failing to probe on kontron-sl28-var3-ads2 in today's broonie-sound tree. I believe this has not reached linux-next yet. Reports aren't automatically sent to the public while we're trialing new bisection features on kernelci.org but this one looks valid. Here's the full boot log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… More details can be found here: https://linux.kernelci.org/test/case/id/608089974135ccea439b779c/ Please let us know if you need any help debugging the issue or to try a fix. Best wishes, Guillaume On 22/04/2021 04:06, KernelCI bot wrote: > * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * > * This automated bisection report was sent to you on the basis * > * that you may be involved with the breaking commit it has * > * found. No manual investigation has been done to verify it, * > * and the root cause of the problem may be somewhere else. * > * * > * If you do send a fix, please include this trailer: * > * Reported-by: "kernelci.org bot" <bot(a)kernelci.org> * > * * > * Hope this helps! * > * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * > > broonie-sound/for-next bisection: baseline.bootrr.asoc-simple-card-probed on kontron-sl28-var3-ads2 > > Summary: > Start: 80e0ab429149 Merge remote-tracking branch 'asoc/for-5.13' into asoc-next > Plain log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… > HTML log: https://storage.kernelci.org/broonie-sound/for-next/v5.12-rc8-542-g80e0ab42… > Result: 59c35c44a9cf ASoC: simple-card: add simple_parse_node() > > Checks: > revert: PASS > verify: PASS > > Parameters: > Tree: broonie-sound > URL: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git > Branch: for-next > Target: kontron-sl28-var3-ads2 > CPU arch: arm64 > Lab: lab-kontron > Compiler: gcc-8 > Config: defconfig+CONFIG_RANDOMIZE_BASE=y > Test case: baseline.bootrr.asoc-simple-card-probed > > Breaking commit found: > > ------------------------------------------------------------------------------- > commit 59c35c44a9cf89a83a92a8d26749e59497d0317d > Author: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Date: Wed Apr 21 14:45:48 2021 +0900 > > ASoC: simple-card: add simple_parse_node() > > Parse dai/tdm/clk are common for both CPU/Codec node. > This patch creates simple_parse_node() for it and share the code. > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Link: https://lore.kernel.org/r/87czuoi41f.wl-kuninori.morimoto.gx@renesas.com > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index a1373be4558f..91af4eca9c86 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c > @@ -93,12 +93,11 @@ static void simple_parse_convert(struct device *dev, > } > > static void simple_parse_mclk_fs(struct device_node *top, > - struct device_node *cpu, > - struct device_node *codec, > + struct device_node *np, > struct simple_dai_props *props, > char *prefix) > { > - struct device_node *node = of_get_parent(cpu); > + struct device_node *node = of_get_parent(np); > char prop[128]; > > snprintf(prop, sizeof(prop), "%smclk-fs", PREFIX); > @@ -106,12 +105,57 @@ static void simple_parse_mclk_fs(struct device_node *top, > > snprintf(prop, sizeof(prop), "%smclk-fs", prefix); > of_property_read_u32(node, prop, &props->mclk_fs); > - of_property_read_u32(cpu, prop, &props->mclk_fs); > - of_property_read_u32(codec, prop, &props->mclk_fs); > + of_property_read_u32(np, prop, &props->mclk_fs); > > of_node_put(node); > } > > +static int simple_parse_node(struct asoc_simple_priv *priv, > + struct device_node *np, > + struct link_info *li, > + char *prefix, > + int is_cpu) > +{ > + struct device *dev = simple_priv_to_dev(priv); > + struct device_node *top = dev->of_node; > + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > + struct snd_soc_dai_link_component *dlc; > + struct asoc_simple_dai *dai; > + int ret, single = 0; > + > + if (is_cpu) { > + dlc = asoc_link_to_cpu(dai_link, 0); > + dai = simple_props_to_dai_cpu(dai_props, 0); > + } else { > + dlc = asoc_link_to_codec(dai_link, 0); > + dai = simple_props_to_dai_codec(dai_props, 0); > + } > + > + simple_parse_mclk_fs(top, np, dai_props, prefix); > + > + ret = asoc_simple_parse_dai(np, dlc, &single); > + if (ret) > + return ret; > + > + ret = asoc_simple_parse_clk(dev, np, dai, dlc); > + if (ret) > + return ret; > + > + ret = asoc_simple_parse_tdm(np, dai); > + if (ret) > + return ret; > + > + if (is_cpu) { > + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > + > + asoc_simple_canonicalize_cpu(dlc, single); > + asoc_simple_canonicalize_platform(platforms, dlc); > + } > + > + return 0; > +} > + > static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > struct device_node *np, > struct device_node *codec, > @@ -121,10 +165,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > struct device *dev = simple_priv_to_dev(priv); > struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > - struct asoc_simple_dai *dai; > struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); > struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); > - struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > struct device_node *top = dev->of_node; > struct device_node *node = of_get_parent(np); > char *prefix = ""; > @@ -132,28 +174,18 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > > dev_dbg(dev, "link_of DPCM (%pOF)\n", np); > > - li->link++; > - > /* For single DAI link & old style of DT node */ > if (is_top) > prefix = PREFIX; > > if (li->cpu) { > - int is_single_links = 0; > - > /* Codec is dummy */ > > /* FE settings */ > dai_link->dynamic = 1; > dai_link->dpcm_merged_format = 1; > > - dai = simple_props_to_dai_cpu(dai_props, 0); > - > - ret = asoc_simple_parse_dai(np, cpus, &is_single_links); > - if (ret) > - goto out_put_node; > - > - ret = asoc_simple_parse_clk(dev, np, dai, cpus); > + ret = simple_parse_node(priv, np, li, prefix, 1); > if (ret < 0) > goto out_put_node; > > @@ -162,9 +194,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > cpus->dai_name); > if (ret < 0) > goto out_put_node; > - > - asoc_simple_canonicalize_cpu(cpus, is_single_links); > - asoc_simple_canonicalize_platform(platforms, cpus); > } else { > struct snd_soc_codec_conf *cconf; > > @@ -174,14 +203,9 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > dai_link->no_pcm = 1; > dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; > > - dai = simple_props_to_dai_codec(dai_props, 0); > cconf = simple_props_to_codec_conf(dai_props, 0); > > - ret = asoc_simple_parse_dai(np, codecs, NULL); > - if (ret < 0) > - goto out_put_node; > - > - ret = asoc_simple_parse_clk(dev, np, dai, codecs); > + ret = simple_parse_node(priv, np, li, prefix, 0); > if (ret < 0) > goto out_put_node; > > @@ -201,11 +225,6 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > } > > simple_parse_convert(dev, np, &dai_props->adata); > - simple_parse_mclk_fs(top, np, codec, dai_props, prefix); > - > - ret = asoc_simple_parse_tdm(np, dai); > - if (ret) > - goto out_put_node; > > ret = asoc_simple_parse_daifmt(dev, node, codec, > prefix, &dai_link->dai_fmt); > @@ -218,6 +237,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, > dai_link->init = asoc_simple_dai_init; > > out_put_node: > + li->link++; > + > of_node_put(node); > return ret; > } > @@ -230,23 +251,18 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > { > struct device *dev = simple_priv_to_dev(priv); > struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); > - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); > - struct asoc_simple_dai *cpu_dai = simple_props_to_dai_cpu(dai_props, 0); > - struct asoc_simple_dai *codec_dai = simple_props_to_dai_codec(dai_props, 0); > struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); > struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); > struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); > - struct device_node *top = dev->of_node; > struct device_node *cpu = NULL; > struct device_node *node = NULL; > struct device_node *plat = NULL; > char prop[128]; > char *prefix = ""; > - int ret, single_cpu = 0; > + int ret; > > cpu = np; > node = of_get_parent(np); > - li->link++; > > dev_dbg(dev, "link_of (%pOF)\n", node); > > @@ -262,13 +278,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > if (ret < 0) > goto dai_link_of_err; > > - simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); > - > - ret = asoc_simple_parse_dai(cpu, cpus, &single_cpu); > + ret = simple_parse_node(priv, cpu, li, prefix, 1); > if (ret < 0) > goto dai_link_of_err; > > - ret = asoc_simple_parse_dai(codec, codecs, NULL); > + ret = simple_parse_node(priv, codec, li, prefix, 0); > if (ret < 0) > goto dai_link_of_err; > > @@ -276,22 +290,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > if (ret < 0) > goto dai_link_of_err; > > - ret = asoc_simple_parse_tdm(cpu, cpu_dai); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_tdm(codec, codec_dai); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_clk(dev, cpu, cpu_dai, cpus); > - if (ret < 0) > - goto dai_link_of_err; > - > - ret = asoc_simple_parse_clk(dev, codec, codec_dai, codecs); > - if (ret < 0) > - goto dai_link_of_err; > - > ret = asoc_simple_set_dailink_name(dev, dai_link, > "%s-%s", > cpus->dai_name, > @@ -302,13 +300,12 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, > dai_link->ops = &simple_ops; > dai_link->init = asoc_simple_dai_init; > > - asoc_simple_canonicalize_cpu(cpus, single_cpu); > - asoc_simple_canonicalize_platform(platforms, cpus); > - > dai_link_of_err: > of_node_put(plat); > of_node_put(node); > > + li->link++; > + > return ret; > } > ------------------------------------------------------------------------------- > > > Git bisection log: > > ------------------------------------------------------------------------------- > git bisect start > # good: [bf05bf16c76bb44ab5156223e1e58e26dfe30a88] Linux 5.12-rc8 > git bisect good bf05bf16c76bb44ab5156223e1e58e26dfe30a88 > # bad: [80e0ab4291498248157d2801c994dfaa09ef3082] Merge remote-tracking branch 'asoc/for-5.13' into asoc-next > git bisect bad 80e0ab4291498248157d2801c994dfaa09ef3082 > # good: [d479f00b795ac62b24ef90f4ec421e65c3178ca7] ASoC: soc-pcm: indicate error message at dpcm_path_get() > git bisect good d479f00b795ac62b24ef90f4ec421e65c3178ca7 > # good: [a96d2ba2d8248d5e8170f2f44f98d4a33329b08a] ASoC: codecs: tlv320aic3x: move I2C to separated file > git bisect good a96d2ba2d8248d5e8170f2f44f98d4a33329b08a > # good: [8577bf61a6359bf2387f85a2fda713a4f05185c3] Merge series "ASoC: rsnd: tidyup Renesas sound" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: > git bisect good 8577bf61a6359bf2387f85a2fda713a4f05185c3 > # good: [87143bfdb9f7ddc14b129fda610e114d29077596] Merge series "ASoC: remove more cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: > git bisect good 87143bfdb9f7ddc14b129fda610e114d29077596 > # good: [23b16df6c9c91f70df070be43c5b13ef4016c1e7] Merge series "ASoC: audio-graph: cleanups" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: > git bisect good 23b16df6c9c91f70df070be43c5b13ef4016c1e7 > # good: [1fa27f35ee23b52e0bd708d00c272c5df805afc8] Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward(a)gatech.edu>: > git bisect good 1fa27f35ee23b52e0bd708d00c272c5df805afc8 > # bad: [434392271afcff350fe11730f12b831fffaf33eb] ASoC: simple-card: add simple_link_init() > git bisect bad 434392271afcff350fe11730f12b831fffaf33eb > # good: [e51237b8d3052251421770468903fa6e4446d158] ASoC: audio-graph: add graph_link_init() > git bisect good e51237b8d3052251421770468903fa6e4446d158 > # bad: [59c35c44a9cf89a83a92a8d26749e59497d0317d] ASoC: simple-card: add simple_parse_node() > git bisect bad 59c35c44a9cf89a83a92a8d26749e59497d0317d > # first bad commit: [59c35c44a9cf89a83a92a8d26749e59497d0317d] ASoC: simple-card: add simple_parse_node() > ------------------------------------------------------------------------------- > > > -=-=-=-=-=-=-=-=-=-=-=- > Groups.io Links: You receive all messages sent to this group. > View/Reply Online (#10236): https://groups.io/g/kernelci-results/message/10236 > Mute This Topic: https://groups.io/mt/82277947/924702 > Group Owner: kernelci-results+owner(a)groups.io > Unsubscribe: https://groups.io/g/kernelci-results/unsub [guillaume.tucker(a)collabora.com] > -=-=-=-=-=-=-=-=-=-=-=- > >

4 13

[PATCH] ALSA: usb-audio: Validate MS endpoint descriptors
by Takashi Iwai 10 May '21

10 May '21

snd_usbmidi_get_ms_info() may access beyond the border when a malformed descriptor is passed. This patch adds the sanity checks of the given MS endpoint descriptors, and skips invalid ones. Reported-by: syzbot+6bb23a5d5548b93c94aa(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index a10ac75969a8..649eb8d1ab7d 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1956,8 +1956,12 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_ep = find_usb_ms_endpoint_descriptor(hostep); if (!ms_ep) continue; + if (ms_ep->bLength <= sizeof(*ms_ep)) + continue; if (ms_ep->bNumEmbMIDIJack > 0x10) continue; + if (ms_ep->bLength < sizeof(*ms_ep) + ms_ep->bNumEmbMIDIJack) + continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { -- 2.26.2

1 0

[syzbot] KASAN: slab-out-of-bounds Read in snd_usbmidi_get_ms_info
by syzbot 10 May '21

10 May '21

Hello, syzbot found the following issue on: HEAD commit: 31a85035 usb: gadget: fsl_qe_udc: fix implicit-fallthrough.. git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing console output: https://syzkaller.appspot.com/x/log.txt?x=109f5a63d00000 kernel config: https://syzkaller.appspot.com/x/.config?x=fbebe41cc7d12c56 dashboard link: https://syzkaller.appspot.com/bug?extid=6bb23a5d5548b93c94aa syz repro: https://syzkaller.appspot.com/x/repro.syz?x=10693995d00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=10e23c65d00000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+6bb23a5d5548b93c94aa(a)syzkaller.appspotmail.com usb 1-1: config 199 interface 0 altsetting 0 endpoint 0x5 has an invalid bInterval 0, changing to 7 usb 1-1: New USB device found, idVendor=7300, idProduct=128c, bcdDevice=ed.06 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 usb 1-1: string descriptor 0 read error: -71 usb 1-1: MIDIStreaming interface descriptor not found ================================================================== BUG: KASAN: slab-out-of-bounds in snd_usbmidi_get_ms_info+0xeaf/0x10e0 sound/usb/midi.c:1982 Read of size 1 at addr ffff888119b80722 by task kworker/0:1/7 CPU: 0 PID: 7 Comm: kworker/0:1 Not tainted 5.12.0-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x143/0x1db lib/dump_stack.c:120 print_address_description.constprop.0.cold+0x5b/0x2f8 mm/kasan/report.c:233 __kasan_report mm/kasan/report.c:419 [inline] kasan_report.cold+0x7c/0xd8 mm/kasan/report.c:436 snd_usbmidi_get_ms_info+0xeaf/0x10e0 sound/usb/midi.c:1982 __snd_usbmidi_create+0x393/0x1ba0 sound/usb/midi.c:2485 snd_usb_create_quirk+0xa5/0xe0 sound/usb/quirks.c:604 usb_audio_probe+0xcf0/0x2c90 sound/usb/card.c:796 usb_probe_interface+0x315/0x7f0 drivers/usb/core/driver.c:396 really_probe+0x291/0xf60 drivers/base/dd.c:576 driver_probe_device+0x298/0x410 drivers/base/dd.c:763 __device_attach_driver+0x203/0x2c0 drivers/base/dd.c:870 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4b0 drivers/base/dd.c:938 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbe0/0x2100 drivers/base/core.c:3319 usb_set_configuration+0x113f/0x1910 drivers/usb/core/message.c:2164 usb_generic_driver_probe+0xba/0x100 drivers/usb/core/generic.c:238 usb_probe_device+0xd9/0x2c0 drivers/usb/core/driver.c:293 really_probe+0x291/0xf60 drivers/base/dd.c:576 driver_probe_device+0x298/0x410 drivers/base/dd.c:763 __device_attach_driver+0x203/0x2c0 drivers/base/dd.c:870 bus_for_each_drv+0x15f/0x1e0 drivers/base/bus.c:431 __device_attach+0x228/0x4b0 drivers/base/dd.c:938 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:491 device_add+0xbe0/0x2100 drivers/base/core.c:3319 usb_new_device.cold+0x721/0x1058 drivers/usb/core/hub.c:2556 hub_port_connect drivers/usb/core/hub.c:5276 [inline] hub_port_connect_change drivers/usb/core/hub.c:5416 [inline] port_event drivers/usb/core/hub.c:5562 [inline] hub_event+0x2357/0x4330 drivers/usb/core/hub.c:5644 process_one_work+0x98d/0x1580 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x38c/0x460 kernel/kthread.c:313 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 Allocated by task 7: kasan_save_stack+0x1b/0x40 mm/kasan/common.c:38 kasan_set_track mm/kasan/common.c:46 [inline] set_alloc_info mm/kasan/common.c:428 [inline] ____kasan_kmalloc mm/kasan/common.c:507 [inline] __kasan_kmalloc+0x7c/0x90 mm/kasan/common.c:516 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x321/0x3d60 drivers/usb/core/config.c:919 usb_enumerate_device drivers/usb/core/hub.c:2389 [inline] usb_new_device+0x42c/0x7a0 drivers/usb/core/hub.c:2525 hub_port_connect drivers/usb/core/hub.c:5276 [inline] hub_port_connect_change drivers/usb/core/hub.c:5416 [inline] port_event drivers/usb/core/hub.c:5562 [inline] hub_event+0x2357/0x4330 drivers/usb/core/hub.c:5644 process_one_work+0x98d/0x1580 kernel/workqueue.c:2275 worker_thread+0x64c/0x1120 kernel/workqueue.c:2421 kthread+0x38c/0x460 kernel/kthread.c:313 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:294 The buggy address belongs to the object at ffff888119b80700 which belongs to the cache kmalloc-64 of size 64 The buggy address is located 34 bytes inside of 64-byte region [ffff888119b80700, ffff888119b80740) The buggy address belongs to the page: page:ffffea000466e000 refcount:1 mapcount:0 mapping:0000000000000000 index:0xffff888119b80680 pfn:0x119b80 flags: 0x200000000000200(slab|node=0|zone=2) raw: 0200000000000200 ffffea0004206f00 0000000900000009 ffff888100041640 raw: ffff888119b80680 000000008020001d 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected page_owner tracks the page as allocated page last allocated via order 0, migratetype Unmovable, gfp_mask 0x12cc0(GFP_KERNEL|__GFP_NOWARN|__GFP_NORETRY), pid 2653, ts 10410979522, free_ts 0 prep_new_page mm/page_alloc.c:2358 [inline] get_page_from_freelist+0xc97/0x26a0 mm/page_alloc.c:3994 __alloc_pages+0x1b2/0x4f0 mm/page_alloc.c:5200 alloc_pages+0x18c/0x2a0 mm/mempolicy.c:2272 alloc_slab_page mm/slub.c:1644 [inline] allocate_slab+0x2c5/0x4c0 mm/slub.c:1784 new_slab mm/slub.c:1847 [inline] new_slab_objects mm/slub.c:2593 [inline] ___slab_alloc+0x476/0x7b0 mm/slub.c:2756 __slab_alloc+0x68/0x80 mm/slub.c:2796 slab_alloc_node mm/slub.c:2878 [inline] slab_alloc mm/slub.c:2920 [inline] kmem_cache_alloc_trace+0x27e/0x2a0 mm/slub.c:2937 kmalloc include/linux/slab.h:556 [inline] kzalloc include/linux/slab.h:686 [inline] allocate_cgrp_cset_links+0x19c/0x240 kernel/cgroup/cgroup.c:1119 find_css_set+0x6d3/0x1a40 kernel/cgroup/cgroup.c:1197 cgroup_migrate_prepare_dst+0x105/0x830 kernel/cgroup/cgroup.c:2641 cgroup_attach_task+0x39f/0x760 kernel/cgroup/cgroup.c:2747 __cgroup1_procs_write.constprop.0+0x3a9/0x490 kernel/cgroup/cgroup-v1.c:519 cgroup_file_write+0x1ec/0x780 kernel/cgroup/cgroup.c:3711 kernfs_fop_write_iter+0x342/0x500 fs/kernfs/file.c:296 call_write_iter include/linux/fs.h:2114 [inline] new_sync_write+0x426/0x650 fs/read_write.c:518 vfs_write+0x743/0x9e0 fs/read_write.c:605 page_owner free stack trace missing Memory state around the buggy address: ffff888119b80600: fa fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff888119b80680: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >ffff888119b80700: 00 00 00 00 02 fc fc fc fc fc fc fc fc fc fc fc ^ ffff888119b80780: fa fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff888119b80800: fa fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[PATCH] ALSA: hda: generic: Remove redundant assignment to dac and dacs[i]
by Jiapeng Chong 10 May '21

10 May '21

Variable dac and dacs[i] is set to zero, but this value is never read as it is overwritten or not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/pci/hda/hda_generic.c:1436:4: warning: Value stored to 'dac' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/hda/hda_generic.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b638fc2..cce1bf7 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1433,7 +1433,6 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, path = snd_hda_add_new_path(codec, dac, pin, 0); } if (!path) { - dac = dacs[i] = 0; badness += bad->no_dac; } else { /* print_nid_path(codec, "output", path); */ -- 1.8.3.1

2 1

[PATCH] ASoC:q6dsp:q6afe-dai: Fix a typo in module description
by Masanari Iida 10 May '21

10 May '21

This patch fixes a spelling typo in MODULE_DESCRIPTION in q6afe-dai.c Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index b539af86e8f7..2166fc305c8f 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1706,5 +1706,5 @@ static struct platform_driver q6afe_dai_platform_driver = { }; module_platform_driver(q6afe_dai_platform_driver); -MODULE_DESCRIPTION("Q6 Audio Fronend dai driver"); +MODULE_DESCRIPTION("Q6 Audio Frontend dai driver"); MODULE_LICENSE("GPL v2"); -- 2.25.0

2 1

[PATCH 1/2] ASoC: imx-akcodec: Add imx-akcodec machine driver
by Shengjiu Wang 08 May '21

08 May '21

Add machine driver for i.MX boards that have AK4458/AK5558/AK4497/AK5552 DAC/ADC attached to SAI interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Adrian Alonso <adrian.alonso(a)nxp.com> --- sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-akcodec.c | 868 ++++++++++++++++++++++++++++++++++++ 3 files changed, 882 insertions(+) create mode 100644 sound/soc/fsl/imx-akcodec.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0917d65d6921..5720ad36494f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -350,6 +350,18 @@ config SND_SOC_IMX_RPMSG Say Y if you want to add support for SoC audio on an i.MX board with a rpmsg devices. +config SND_SOC_IMX_AKCODEC + tristate "SoC Audio support for i.MX boards with AKCODEC" + depends on OF && I2C + select SND_SOC_AK4458 + select SND_SOC_AK5558 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with AK4458/AK4497/AK5558/AK5552 + Say Y if you want to add support for SoC audio on an i.MX board with + an AK4458/AK4497/AK5558/AK5552 DAC/ADC. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f146ce464acd..ee7caed0c758 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -71,6 +71,7 @@ snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-audmix-objs := imx-audmix.o snd-soc-imx-hdmi-objs := imx-hdmi.o snd-soc-imx-rpmsg-objs := imx-rpmsg.o +snd-soc-imx-akcodec-objs := imx-akcodec.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o @@ -79,3 +80,4 @@ obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o obj-$(CONFIG_SND_SOC_IMX_HDMI) += snd-soc-imx-hdmi.o obj-$(CONFIG_SND_SOC_IMX_RPMSG) += snd-soc-imx-rpmsg.o +obj-$(CONFIG_SND_SOC_IMX_AKCODEC) += snd-soc-imx-akcodec.o diff --git a/sound/soc/fsl/imx-akcodec.c b/sound/soc/fsl/imx-akcodec.c new file mode 100644 index 000000000000..606365f79a54 --- /dev/null +++ b/sound/soc/fsl/imx-akcodec.c @@ -0,0 +1,868 @@ +// SPDX-License-Identifier: GPL-2.0+ +// Copyright 2017-2021 NXP + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/pcm.h> +#include <sound/soc-dapm.h> + +#include "fsl_sai.h" + +#define MAX_CODECS 2 + +enum codec_type { + CODEC_AK5558 = 1, + CODEC_AK4458, + CODEC_AK4497, + CODEC_AK5552, +}; + +/* + * Mapping LRCK fs and frame width, table 3 & 4 in datasheet + * @rmin: min rate + * @rmax: max rate + * @wmin: min frame ratio + * @wmax: max frame ratio + */ +struct imx_akcodec_fs_mul { + unsigned int rmin; + unsigned int rmax; + unsigned int wmin; + unsigned int wmax; +}; + +/* + * Mapping TDM mode and frame width + */ +struct imx_akcodec_tdm_fs_mul { + unsigned int min; + unsigned int max; + unsigned int mul; +}; + +/* + * struct imx_akcodec_plat_data - specific info for codecs + * + * @fs_mul: ratio of mclk/fs for normal mode + * @tdm_fs_mul: ratio of mclk/fs for tdm mode + * @support_rates: supported sample rate + * @support_tdm_rates: supported sample rate for tdm mode + * @support_channels: supported channels + * @support_tdm_channels: supported channels for tdm mode + * @num_fs_mul: ARRAY_SIZE of fs_mul + * @num_tdm_fs_mul: ARRAY_SIZE of tdm_fs_mul + * @num_rates: ARRAY_SIZE of support_rates + * @num_tdm_rates: ARRAY_SIZE of support_tdm_rates + * @num_channels: ARRAY_SIZE of support_channels + * @num_tdm_channels: ARRAY_SIZE of support_tdm_channels + * @num_codecs: codec number, maximum is MAX_CODECS + * @playback_only: only playback supported + * @capture_only: only capture supported + * @one2one_ratio: if mclk/bclk = 1 or not + * @tdm_is_support: if this codec support tdm + * @type: codec type + * @dai_name: codec dai name + */ +struct imx_akcodec_plat_data { + struct imx_akcodec_fs_mul *fs_mul; + struct imx_akcodec_tdm_fs_mul *tdm_fs_mul; + const u32 *support_rates; + const u32 *support_tdm_rates; + const u32 *support_channels; + const u32 *support_tdm_channels; + unsigned int num_fs_mul; + unsigned int num_tdm_fs_mul; + unsigned int num_rates; + unsigned int num_tdm_rates; + unsigned int num_channels; + unsigned int num_tdm_channels; + unsigned int num_codecs; + unsigned int playback_only; + unsigned int capture_only; + bool one2one_ratio; + bool tdm_is_support; + enum codec_type type; + const char *dai_name; +}; + +/* + * struct imx_akcodec_data - platform device data + * + * @card: card instance + * @codec_conf: pointer of codec conf + * @plat_data: pointer of imx_akcodec_plat_data + * @dai_link: snd_soc_dai_link + * @codecs: codec dai link component + * @dapm_routes: pointer of dapm_routes + * @asrc_pdev: pointer of asrc_pdev + * @num_codec_conf: number of codec conf + * @num_dapm_routes: number of dapm_routes + * @slots: slots value + * @slot_width: slot width value + * @tdm_mode: tdm mode enabled or not + * @asrc_rate: asrc rates + * @asrc_format: asrc format + */ +struct imx_akcodec_data { + struct snd_soc_card card; + struct snd_soc_codec_conf *codec_conf; + const struct imx_akcodec_plat_data *plat_data; + struct snd_soc_dai_link dai_link[3]; + struct snd_soc_dai_link_component codecs[MAX_CODECS]; + struct snd_soc_dapm_route *dapm_routes; + struct platform_device *asrc_pdev; + int num_codec_conf; + int num_dapm_routes; + unsigned int slots; + unsigned int slot_width; + bool tdm_mode; + u32 asrc_rate; + u32 asrc_format; +}; + +struct imx_akcodec_fs_mul ak4458_fs_mul[] = { + /* Normal, < 32kHz */ + { .rmin = 8000, .rmax = 24000, .wmin = 1024, .wmax = 1024, }, + /* Normal, 32kHz */ + { .rmin = 32000, .rmax = 32000, .wmin = 256, .wmax = 1024, }, + /* Normal */ + { .rmin = 44100, .rmax = 48000, .wmin = 256, .wmax = 768, }, + /* Double */ + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 512, }, + /* Quad */ + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 256, }, + /* Oct */ + { .rmin = 352800, .rmax = 384000, .wmin = 32, .wmax = 128, }, + /* Hex */ + { .rmin = 705600, .rmax = 768000, .wmin = 16, .wmax = 64, }, +}; + +struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { + /* + * Table 13 - Audio Interface Format + * For TDM mode, MCLK should is set to + * obtained from 2 * slots * slot_width + */ + { .min = 128, .max = 128, .mul = 256 }, /* TDM128 */ + { .min = 256, .max = 256, .mul = 512 }, /* TDM256 */ + { .min = 512, .max = 512, .mul = 1024 }, /* TDM512 */ +}; + +struct imx_akcodec_fs_mul ak4497_fs_mul[] = { + /** + * Table 7 - mapping multiplier and speed mode + * Tables 8 & 9 - mapping speed mode and LRCK fs + */ + { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, /* Normal, <= 32kHz */ + { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, /* Normal */ + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, /* Double */ + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 128, }, /* Quad */ + { .rmin = 352800, .rmax = 384000, .wmin = 128, .wmax = 128, }, /* Oct */ + { .rmin = 705600, .rmax = 768000, .wmin = 64, .wmax = 64, }, /* Hex */ +}; + +/* + * Auto MCLK selection based on LRCK for Normal Mode + * (Table 4 from datasheet) + */ +struct imx_akcodec_fs_mul ak5558_fs_mul[] = { + { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, + { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 128, }, + { .rmin = 352800, .rmax = 384000, .wmin = 64, .wmax = 64, }, + { .rmin = 705600, .rmax = 768000, .wmin = 32, .wmax = 32, }, +}; + +/* + * MCLK and BCLK selection based on TDM mode + * because of SAI we also add the restriction: MCLK >= 2 * BCLK + * (Table 9 from datasheet) + */ +struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { + { .min = 128, .max = 128, .mul = 256 }, + { .min = 256, .max = 256, .mul = 512 }, + { .min = 512, .max = 512, .mul = 1024 }, +}; + +static const u32 akcodec_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 352800, 384000, 705600, 768000, +}; + +static const u32 akcodec_tdm_rates[] = { + 8000, 16000, 32000, 48000, 96000, +}; + +static const u32 ak4458_channels[] = { + 1, 2, 4, 6, 8, 10, 12, 14, 16, +}; + +static const u32 ak4458_tdm_channels[] = { + 1, 2, 3, 4, 5, 6, 7, 8, 16, +}; + +static const u32 ak5558_channels[] = { + 1, 2, 4, 6, 8, +}; + +static const u32 ak5558_tdm_channels[] = { + 1, 2, 3, 4, 5, 6, 7, 8, +}; + +static const struct imx_akcodec_plat_data imx8mq_ak4458 = { + .fs_mul = ak4458_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak4458_fs_mul), + .tdm_fs_mul = ak4458_tdm_fs_mul, + .num_tdm_fs_mul = ARRAY_SIZE(ak4458_tdm_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .support_channels = ak4458_channels, + .num_channels = ARRAY_SIZE(ak4458_channels), + .support_tdm_channels = ak4458_tdm_channels, + .num_tdm_channels = ARRAY_SIZE(ak4458_tdm_channels), + .one2one_ratio = false, + .dai_name = "ak4458-aif", + .num_codecs = MAX_CODECS, + .tdm_is_support = true, + .playback_only = 1, + .capture_only = 0, + .type = CODEC_AK4458, +}; + +static const struct imx_akcodec_plat_data imx_ak4458 = { + .fs_mul = ak4458_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak4458_fs_mul), + .tdm_fs_mul = ak4458_tdm_fs_mul, + .num_tdm_fs_mul = ARRAY_SIZE(ak4458_tdm_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .support_channels = ak4458_channels, + .num_channels = ARRAY_SIZE(ak4458_channels), + .support_tdm_channels = ak4458_tdm_channels, + .num_tdm_channels = ARRAY_SIZE(ak4458_tdm_channels), + .one2one_ratio = true, + .dai_name = "ak4458-aif", + .num_codecs = MAX_CODECS, + .tdm_is_support = true, + .playback_only = 1, + .capture_only = 0, + .type = CODEC_AK4458, +}; + +static const struct imx_akcodec_plat_data imx8mq_ak4497 = { + .fs_mul = ak4497_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak4497_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_channels = ak4458_channels, + .num_channels = ARRAY_SIZE(ak4458_channels), + .one2one_ratio = false, + .dai_name = "ak4497-aif", + .num_codecs = 1, + .tdm_is_support = false, + .playback_only = 1, + .capture_only = 0, + .type = CODEC_AK4497, +}; + +static const struct imx_akcodec_plat_data imx_ak4497 = { + .fs_mul = ak4497_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak4497_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_channels = ak4458_channels, + .num_channels = ARRAY_SIZE(ak4458_channels), + .one2one_ratio = true, + .dai_name = "ak4497-aif", + .num_codecs = 1, + .tdm_is_support = false, + .playback_only = 1, + .capture_only = 0, + .type = CODEC_AK4497, +}; + +static const struct imx_akcodec_plat_data imx8mq_ak5558 = { + .fs_mul = ak5558_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak5558_fs_mul), + .tdm_fs_mul = ak5558_tdm_fs_mul, + .num_tdm_fs_mul = ARRAY_SIZE(ak5558_tdm_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .support_channels = ak5558_channels, + .num_channels = ARRAY_SIZE(ak5558_channels), + .support_tdm_channels = ak5558_tdm_channels, + .num_tdm_channels = ARRAY_SIZE(ak5558_tdm_channels), + .one2one_ratio = false, + .dai_name = "ak5558-aif", + .num_codecs = 1, + .tdm_is_support = true, + .playback_only = 0, + .capture_only = 1, + .type = CODEC_AK5558, +}; + +static const struct imx_akcodec_plat_data imx_ak5558 = { + .fs_mul = ak5558_fs_mul, + .num_fs_mul = ARRAY_SIZE(ak5558_fs_mul), + .tdm_fs_mul = ak5558_tdm_fs_mul, + .num_tdm_fs_mul = ARRAY_SIZE(ak5558_tdm_fs_mul), + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .support_channels = ak5558_channels, + .num_channels = ARRAY_SIZE(ak5558_channels), + .support_tdm_channels = ak5558_tdm_channels, + .num_tdm_channels = ARRAY_SIZE(ak5558_tdm_channels), + .one2one_ratio = true, + .dai_name = "ak5558-aif", + .num_codecs = 1, + .tdm_is_support = true, + .playback_only = 0, + .capture_only = 1, + .type = CODEC_AK5558, +}; + +static bool format_is_dsd(struct snd_pcm_hw_params *params) +{ + snd_pcm_format_t format = params_format(params); + + switch (format) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + return true; + default: + return false; + } +} + +static struct snd_soc_dapm_widget imx_akcodec_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(rtd->card); + const struct imx_akcodec_plat_data *plat_data = data->plat_data; + unsigned int width = data->slots * data->slot_width; + unsigned int rate = params_rate(params); + int i; + + if (data->tdm_mode) { + for (i = 0; i < plat_data->num_tdm_fs_mul; i++) { + /* min = max = slots * slots_width */ + if (width != plat_data->tdm_fs_mul[i].min) + continue; + return rate * plat_data->tdm_fs_mul[i].mul; + } + } else { + for (i = 0; i < plat_data->num_fs_mul; i++) { + if (rate >= plat_data->fs_mul[i].rmin && + rate <= plat_data->fs_mul[i].rmax) { + width = max(width, plat_data->fs_mul[i].wmin); + width = min(width, plat_data->fs_mul[i].wmax); + + /* Adjust SAI bclk:mclk ratio */ + width *= plat_data->one2one_ratio ? 1 : 2; + + return rate * width; + } + } + } + + /* Let DAI manage clk frequency by default */ + return 0; +} + +static int imx_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + unsigned int fmt = SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + bool is_dsd = format_is_dsd(params); + struct device *dev = card->dev; + unsigned long mclk_freq; + int ret, i; + + if (is_dsd) { + data->slots = 1; + data->slot_width = params_width(params); + fmt |= SND_SOC_DAIFMT_PDM; + } else if (data->tdm_mode) { + data->slots = 8; + data->slot_width = 32; + fmt |= SND_SOC_DAIFMT_DSP_B; + } else { + data->slots = 2; + data->slot_width = params_physical_width(params); + fmt |= SND_SOC_DAIFMT_I2S; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_tdm_slot(cpu_dai, + BIT(data->slots) - 1, + BIT(data->slots) - 1, + data->slots, data->slot_width); + if (ret) { + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + return ret; + } + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, i); + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret) { + dev_err(dev, "failed to set codec dai[%d] fmt: %d\n", i, ret); + return ret; + } + ret = snd_soc_dai_set_tdm_slot(codec_dai, + BIT(data->slots) - 1, + BIT(data->slots) - 1, + data->slots, data->slot_width); + if (ret) { + dev_err(dev, "failed to set codec dai[%d] tdm slot: %d\n", i, ret); + return ret; + } + } + + /* Set MCLK freq */ + mclk_freq = akcodec_get_mclk_rate(substream, params); + /* Use the maximum freq from DSD512 (512*44100 = 22579200) */ + if (is_dsd) + mclk_freq = 22579200; + + ret = snd_soc_dai_set_sysclk(cpu_dai, FSL_SAI_CLK_MAST1, mclk_freq, + SND_SOC_CLOCK_OUT); + if (ret < 0) + dev_err(dev, "failed to set cpui dai mclk1 rate (%lu): %d\n", + mclk_freq, ret); + return ret; +} + +static int ak5558_hw_rule_rate(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *r) +{ + struct imx_akcodec_data *data = r->private; + struct snd_interval t = { .min = 8000, .max = 8000, }; + unsigned long mclk_freq; + unsigned int fs; + int i; + + fs = hw_param_interval(p, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + fs *= data->tdm_mode ? 8 : 2; + + /* Identify maximum supported rate */ + for (i = 0; i < ARRAY_SIZE(akcodec_rates); i++) { + mclk_freq = fs * akcodec_rates[i]; + /* Adjust SAI bclk:mclk ratio */ + mclk_freq *= data->plat_data->one2one_ratio ? 1 : 2; + + /* Skip rates for which MCLK is beyond supported value */ + if (mclk_freq > 36864000) + continue; + + if (t.max < akcodec_rates[i]) + t.max = akcodec_rates[i]; + } + + return snd_interval_refine(hw_param_interval(p, r->var), &t); +} + +static int imx_aif_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + static struct snd_pcm_hw_constraint_list constraint_rates; + static struct snd_pcm_hw_constraint_list constraint_channels; + int ret = 0; + + if (data->tdm_mode) { + constraint_channels.list = data->plat_data->support_tdm_channels; + constraint_channels.count = data->plat_data->num_tdm_channels; + constraint_rates.list = data->plat_data->support_tdm_rates; + constraint_rates.count = data->plat_data->num_tdm_rates; + } else { + constraint_channels.list = data->plat_data->support_channels; + constraint_channels.count = data->plat_data->num_channels; + constraint_rates.list = data->plat_data->support_rates; + constraint_rates.count = data->plat_data->num_rates; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraint_channels); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraint_rates); + if (ret) + return ret; + + if (data->plat_data->type == CODEC_AK5558) + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ak5558_hw_rule_rate, data, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + + return ret; +} + +static int imx_aif_hw_params_be(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + unsigned int slots, slot_width; + unsigned long mclk_freq; + unsigned int fmt; + int ret; + + /* In order to support odd channels, force tdm mode for FE-BE case */ + if (data->plat_data->tdm_is_support) { + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + /* support TDM256 (8 slots * 32 bits/per slot) */ + slots = 8; + slot_width = 32; + } else { + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + slots = 2; + slot_width = 32; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, BIT(slots) - 1, BIT(slots) - 1, + slots, slot_width); + if (ret) { + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, BIT(slots) - 1, BIT(slots) - 1, + slots, slot_width); + if (ret) { + dev_err(dev, "failed to set codec dai fmt: %d\n", ret); + return ret; + } + + mclk_freq = akcodec_get_mclk_rate(substream, params); + ret = snd_soc_dai_set_sysclk(cpu_dai, FSL_SAI_CLK_MAST1, mclk_freq, + SND_SOC_CLOCK_OUT); + if (ret < 0) + dev_err(dev, "failed to set cpu_dai mclk1 rate %lu\n", + mclk_freq); + + return ret; +} + +static struct snd_soc_ops imx_aif_ops = { + .hw_params = imx_aif_hw_params, + .startup = imx_aif_startup, +}; + +static struct snd_soc_ops imx_aif_ops_be = { + .hw_params = imx_aif_hw_params_be, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + struct snd_interval *rate; + struct snd_mask *mask; + + if (!data->asrc_pdev) + return -EINVAL; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = data->asrc_rate; + rate->min = data->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, data->asrc_format); + + return 0; +} + +static int imx_akcodec_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link_component *dlc; + struct device_node *codec_np[MAX_CODECS]; + struct device_node *cpu_np = NULL; + struct imx_akcodec_data *data; + struct device_node *asrc_np = NULL; + char tmp[MAX_CODECS][8]; + int ret, i; + u32 width; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->plat_data = of_device_get_match_data(&pdev->dev); + + dlc = devm_kzalloc(&pdev->dev, 9 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + cpu_np = of_parse_phandle(pdev->dev.of_node, "audio-cpu", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "audio dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + /* Parse codec node */ + for (i = 0; i < data->plat_data->num_codecs; i++) { + codec_np[i] = of_parse_phandle(pdev->dev.of_node, "audio-codec", i); + if (!codec_np[i]) { + dev_err(&pdev->dev, "audio codec phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + } + + asrc_np = of_parse_phandle(pdev->dev.of_node, "audio-asrc", 0); + if (asrc_np) + data->asrc_pdev = of_find_device_by_node(asrc_np); + + if (of_find_property(pdev->dev.of_node, "fsl,tdm", NULL) && + data->plat_data->tdm_is_support) + data->tdm_mode = true; + + data->num_codec_conf = data->plat_data->num_codecs; + data->codec_conf = devm_kzalloc(&pdev->dev, + data->num_codec_conf * sizeof(struct snd_soc_codec_conf), + GFP_KERNEL); + if (!data->codec_conf) { + ret = -ENOMEM; + goto fail; + } + + data->num_dapm_routes = data->plat_data->num_codecs + 1; + data->dapm_routes = devm_kcalloc(&pdev->dev, data->num_dapm_routes, + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!data->dapm_routes) { + ret = -ENOMEM; + goto fail; + } + + data->dai_link[0].name = "akcodec"; + data->dai_link[0].stream_name = "Audio"; + data->dai_link[0].cpus = &dlc[0]; + data->dai_link[0].platforms = &dlc[1]; + data->dai_link[0].codecs = data->codecs; + data->dai_link[0].num_cpus = 1; + data->dai_link[0].num_platforms = 1; + data->dai_link[0].num_codecs = data->plat_data->num_codecs; + data->dai_link[0].cpus->of_node = cpu_np; + data->dai_link[0].platforms->of_node = cpu_np; + + for (i = 0; i < data->plat_data->num_codecs; i++) { + sprintf(tmp[i], "%d", i); + data->codec_conf[i].name_prefix = tmp[i]; + data->codec_conf[i].dlc.of_node = codec_np[i]; + data->codecs[i].of_node = codec_np[i]; + data->codecs[i].dai_name = data->plat_data->dai_name; + + if (data->plat_data->playback_only) { + data->dapm_routes[i].sink = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + tmp[i], "Playback"); + data->dapm_routes[i].source = "CPU-Playback"; + } else { + data->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s %s", + tmp[i], "Capture"); + data->dapm_routes[i].sink = "CPU-Capture"; + } + } + + if (data->plat_data->playback_only) { + data->dapm_routes[i].sink = "CPU-Playback"; + data->dapm_routes[i].source = "ASRC-Playback"; + } else { + data->dapm_routes[i].sink = "ASRC-Capture"; + data->dapm_routes[i].source = "CPU-Capture"; + } + + data->dai_link[0].ops = &imx_aif_ops; + data->dai_link[0].ignore_pmdown_time = 1; + data->dai_link[0].playback_only = data->plat_data->playback_only; + data->dai_link[0].capture_only = data->plat_data->capture_only; + + data->card.codec_conf = data->codec_conf; + data->card.num_configs = data->num_codec_conf; + data->card.dai_link = &data->dai_link[0]; + data->card.num_links = 1; + data->card.dev = &pdev->dev; + data->card.owner = THIS_MODULE; + data->card.dapm_widgets = imx_akcodec_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_akcodec_dapm_widgets); + + /*if there is no asrc controller, we only enable one device*/ + if (data->asrc_pdev) { + data->dai_link[1].name = "HiFi-ASRC-FE"; + data->dai_link[1].stream_name = "HiFi-ASRC-FE"; + data->dai_link[1].dynamic = 1; + data->dai_link[1].ignore_pmdown_time = 1; + data->dai_link[1].dpcm_playback = data->plat_data->playback_only; + data->dai_link[1].dpcm_capture = data->plat_data->capture_only; + data->dai_link[1].dpcm_merged_chan = 1; + + data->dai_link[1].cpus = &dlc[2]; + data->dai_link[1].platforms = &dlc[3]; + data->dai_link[1].codecs = &dlc[4]; + data->dai_link[1].num_cpus = 1; + data->dai_link[1].num_platforms = 1; + data->dai_link[1].num_codecs = 1; + data->dai_link[1].cpus->of_node = asrc_np; + data->dai_link[1].platforms->of_node = asrc_np; + data->dai_link[1].codecs->name = "snd-soc-dummy"; + data->dai_link[1].codecs->dai_name = "snd-soc-dummy-dai"; + + data->dai_link[2].cpus = &dlc[5]; + data->dai_link[2].platforms = &dlc[6]; + data->dai_link[2].codecs = data->codecs; + data->dai_link[2].num_cpus = 1; + data->dai_link[2].num_platforms = 1; + data->dai_link[2].num_codecs = data->plat_data->num_codecs; + + data->dai_link[2].name = "HiFi-ASRC-BE"; + data->dai_link[2].stream_name = "HiFi-ASRC-BE"; + data->dai_link[2].cpus->of_node = cpu_np; + data->dai_link[2].platforms->name = "snd-soc-dummy"; + data->dai_link[2].ops = &imx_aif_ops_be; + data->dai_link[2].be_hw_params_fixup = be_hw_params_fixup; + data->dai_link[2].no_pcm = 1; + data->dai_link[2].ignore_pmdown_time = 1; + data->dai_link[2].dpcm_playback = data->plat_data->playback_only; + data->dai_link[2].dpcm_capture = data->plat_data->capture_only; + + data->card.dapm_routes = data->dapm_routes; + data->card.num_dapm_routes = data->num_dapm_routes; + data->card.num_links = 3; + + ret = of_property_read_u32(asrc_np, "fsl,asrc-rate", + &data->asrc_rate); + if (ret) { + dev_err(&pdev->dev, "failed to get output rate\n"); + ret = -EINVAL; + goto fail; + } + + ret = of_property_read_u32(asrc_np, "fsl,asrc-format", + &data->asrc_format); + if (ret) { + /* Fallback to old binding; translate to asrc_format */ + ret = of_property_read_u32(asrc_np, "fsl,asrc-width", + &width); + if (ret) { + dev_err(&pdev->dev, + "failed to decide output format\n"); + goto fail; + } + + if (width == 24) + data->asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + data->asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + } + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + + ret = 0; +fail: + if (cpu_np) + of_node_put(cpu_np); + + for (i = 0; i < data->plat_data->num_codecs; i++) + if (codec_np[i]) + of_node_put(codec_np[i]); + + return ret; +} + +static const struct of_device_id imx_akcodec_dt_ids[] = { + { .compatible = "fsl,imx8mq-audio-ak4458", .data = &imx8mq_ak4458, }, + { .compatible = "fsl,imx8mq-audio-ak4497", .data = &imx8mq_ak4497, }, + { .compatible = "fsl,imx8mq-audio-ak5558", .data = &imx8mq_ak5558, }, + { .compatible = "fsl,imx-audio-ak4458", .data = &imx_ak4458, }, + { .compatible = "fsl,imx-audio-ak4497", .data = &imx_ak4497, }, + { .compatible = "fsl,imx-audio-ak5558", .data = &imx_ak5558, }, + { .compatible = "fsl,imx-audio-ak5552", .data = &imx_ak5558, }, + { }, +}; +MODULE_DEVICE_TABLE(of, imx_akcodec_dt_ids); + +static struct platform_driver imx_akcodec_driver = { + .driver = { + .name = "imx-akcodec", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_akcodec_dt_ids, + }, + .probe = imx_akcodec_probe, +}; +module_platform_driver(imx_akcodec_driver); + +MODULE_DESCRIPTION("Freescale i.MX AK CODEC ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-akcodec"); -- 2.17.1

4 5

ASUS UM425IA shows headphones and speakers despite having no headphone jack
by Julian Sikorski 08 May '21

08 May '21

Hello, on ASUS UM425IA Pipewire reports having both speakers and headphones available. This machine has no headphone jack, it comes with a USB-C dongle (which I have not tested yet). Switching between the two changes the volume a bit - it is louder on speakers - but other than that audio plays regardless of what is selected. Headphones were selected by default. I have saved alsa-info.sh output here: http://alsa-project.org/db/?f=c0a5c7eb7dbe5bf6550ba61740c7f1588b8db16e Thank you for your support in advance. Best regards, Julian

1 0

[PATCH] ASoC: cs42l42: make HSBIAS_SENSE_EN optional
by Lucas Tanure 07 May '21

07 May '21

From: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> HSBIAS_SENSE_EN configures HSBIAS output current sense through the external 2.21-k resistor. HSBIAS_SENSE is hardware feature to reduce the potential pop noise during the headset plug out slowly. But on some platforms ESD voltage will affect it causing test to fail, especially with CTIA headset type. For different hardware setups, a designer might want to tweak default behavior. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- .../devicetree/bindings/sound/cs42l42.txt | 10 ++++++++++ include/dt-bindings/sound/cs42l42.h | 3 +++ sound/soc/codecs/cs42l42.c | 20 ++++++++++++++++++- sound/soc/codecs/cs42l42.h | 1 + 4 files changed, 33 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs42l42.txt b/Documentation/devicetree/bindings/sound/cs42l42.txt index 7dfaa2ab906f..e75f191d2945 100644 --- a/Documentation/devicetree/bindings/sound/cs42l42.txt +++ b/Documentation/devicetree/bindings/sound/cs42l42.txt @@ -81,6 +81,16 @@ Optional properties: < x1 x2 x3 x4 > Default = < 15 8 4 1> + - cirrus,hs-bias-sense-en: Configures HSBIAS output current sense through + the external 2.21-k resistor. HSBIAS_SENSE is hardware feature to reduce + the potential pop noise during the headset plug out slowly. But on some + platforms ESD voltage will affect it causing test to fail, especially + with CTIA headset type. For different hardware setups, a designer might + want to tweak default behavior. + + 0 - disabled + 1 - enabled + Default = 1 Example: diff --git a/include/dt-bindings/sound/cs42l42.h b/include/dt-bindings/sound/cs42l42.h index f25d83c6188b..788e1598b15f 100644 --- a/include/dt-bindings/sound/cs42l42.h +++ b/include/dt-bindings/sound/cs42l42.h @@ -66,4 +66,7 @@ #define CS42L42_HSBIAS_RAMP_TIME2 90 #define CS42L42_HSBIAS_RAMP_TIME3 170 +#define CS42L42_HSBIAS_SENSE_ON 1 +#define CS42L42_HSBIAS_SENSE_OFF 0 + #endif /* __DT_CS42L42_H */ diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d9f8da7a68d0..139245c6b720 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1032,7 +1032,7 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) CS42L42_AUTO_HSBIAS_HIZ_MASK | CS42L42_TIP_SENSE_EN_MASK | CS42L42_HSBIAS_SENSE_TRIP_MASK, - (1 << CS42L42_HSBIAS_SENSE_EN_SHIFT) | + (cs42l42->hs_bias_sense_en << CS42L42_HSBIAS_SENSE_EN_SHIFT) | (1 << CS42L42_AUTO_HSBIAS_HIZ_SHIFT) | (0 << CS42L42_TIP_SENSE_EN_SHIFT) | (3 << CS42L42_HSBIAS_SENSE_TRIP_SHIFT)); @@ -1807,6 +1807,24 @@ static int cs42l42_handle_device_data(struct device *dev, (cs42l42->hs_bias_ramp_rate << CS42L42_HSBIAS_RAMP_SHIFT)); + ret = device_property_read_u32(dev, "cirrus,hs-bias-sense-en", &val); + if (!ret) { + switch (val) { + case CS42L42_HSBIAS_SENSE_OFF: + case CS42L42_HSBIAS_SENSE_ON: + cs42l42->hs_bias_sense_en = val; + break; + default: + dev_err(dev, + "Wrong cirrus,hs-bias-sense-en DT value %d\n", + val); + cs42l42->hs_bias_sense_en = CS42L42_HSBIAS_SENSE_ON; + break; + } + } else { + cs42l42->hs_bias_sense_en = CS42L42_HSBIAS_SENSE_ON; + } + return 0; } diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 2e0d3836bd7e..24f7be228d5f 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -787,6 +787,7 @@ struct cs42l42_private { u8 bias_thresholds[CS42L42_NUM_BIASES]; u8 hs_bias_ramp_rate; u8 hs_bias_ramp_time; + u8 hs_bias_sense_en; u8 stream_use; }; -- 2.31.1

3 3

[PATCH alsa-ucm-conf 0/3] byt/cht: Add support for controlling mute LEDs using the new snd_ctl_led kmod
by Hans de Goede 07 May '21

07 May '21

Hi All, This series adds support for controlling spk/mic mute LEDs one some Bay- and Cherry-Trail devices using the new snd_ctl_led kmod functionality. This requires the latest alsa-lib and alsa-utils git code and when running on Fedora it also requires putting selinux in permissive mode for now. I've filed a bug against the selinux-policy to get this fixed: https://bugzilla.redhat.com/show_bug.cgi?id=1958210 Note the first patch is mostly unrelated, it is a resend of an earlier patch which seems to have fallen through the cracks. Regards, Hans Hans de Goede (3): cht-bsw-rt5672: Add support for the components string cht-bsw-rt5672: Add support for controlling speaker- and mic-mute LEDs bytcr-rt5640: Add support for controlling a speaker-mute LED ucm2/bytcr-rt5640/bytcr-rt5640.conf | 6 ++++ ucm2/cht-bsw-rt5672/HiFi.conf | 37 +++++++++++++++++++++++-- ucm2/cht-bsw-rt5672/cht-bsw-rt5672.conf | 6 ++++ 3 files changed, 47 insertions(+), 2 deletions(-) -- 2.31.1

1 0

[BUG] ALSA: korg1212: Potential NULL pointer dereference in snd_korg1212_interrupt()
by Подгорный Алексей Олегович 07 May '21

07 May '21

snd_korg1212_create() makes the following steps during initialization of the card: 1) registers an interrupt handler (lines 2230-2232) 2) allocates and initializes korg1212->sharedBufferPtr (lines 2280-2287) 3) reboots the card via snd_korg1212_Send1212Command() (line 2358) 2145 static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, struct snd_korg1212 **rchip) 2147 { ... 2230 err = request_irq(pci->irq, snd_korg1212_interrupt, 2231 IRQF_SHARED, 2232 KBUILD_MODNAME, korg1212); ... 2280 if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &pci->dev, 2281 sizeof(struct KorgSharedBuffer), &korg1212->dma_shared) < 0){ 2282 snd_printk(KERN_ERR "korg1212: can not allocate shared buffer memory (%zdbytes)\n", sizeof(struct KorgSharedBuffer)); 2283 snd_korg1212_free(korg1212); 2284 return -ENOMEM; 2285 } 2286 korg1212->sharedBufferPtr = (struct KorgSharedBuffer*)korg1212->dma_shared.area; 2287 korg1212->sharedBufferPhy = korg1212->dma_shared.addr; ... 2358 rc = snd_korg1212_Send1212Command(korg1212, K1212_DB_RebootCard, 0, 0, 0, 0); ... 2412 } But if interrupt happens when snd_korg1212_create() is still within lines 2233-2286, snd_korg1212_interrupt() may dereference korg1212->sharedBufferPtr before it was initialized without any checks (line 1149): 1098 static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) 1099 { ... 1116 switch (doorbellValue) { ... 1145 case K1212_DB_CARDSTOPPED: 1146 K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ CSTP count - %ld, %x, [%s].\n", 1147 korg1212->irqcount, doorbellValue, 1148 stateName[korg1212->cardState]); 1149 korg1212->sharedBufferPtr->cardCommand = 0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ 1150 break; ... 1185 } Should we be sure that such interrupt cannot happen or should we move the registration of the interrupt handler after korg1212->sharedBufferPtr is initialized? Found by Linux Driver Verification project (linuxtesting.org).

2 1

[PATCH] ALSA: hda/realtek: reset eapd coeff to default value for alc287
by Hui Wang 07 May '21

07 May '21

Ubuntu users reported an audio bug on the Lenovo Yoga Slim 7 14IIL05, he installed dual OS (Windows + Linux), if he booted to the Linux from Windows, the Speaker can't work well, it has crackling noise, if he poweroff the machine first after Windows, the Speaker worked well. Before rebooting or shutdown from Windows, the Windows changes the codec eapd coeff value, but the BIOS doesn't re-initialize its value, when booting into the Linux from Windows, the eapd coeff value is not correct. To fix it, set the codec default value to that coeff register in the alsa driver. BugLink: http://bugs.launchpad.net/bugs/1925057 Suggested-by: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d58f24c9702..a5f3e78ec04e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -395,7 +395,6 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0282: case 0x10ec0283: case 0x10ec0286: - case 0x10ec0287: case 0x10ec0288: case 0x10ec0285: case 0x10ec0298: @@ -406,6 +405,10 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0275: alc_update_coef_idx(codec, 0xe, 0, 1<<0); break; + case 0x10ec0287: + alc_update_coef_idx(codec, 0x10, 1<<9, 0); + alc_write_coef_idx(codec, 0x8, 0x4ab7); + break; case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; -- 2.25.1

2 1

[PATCH v2 0/2] ASoC: meson: g12a-toacodec: add support for SM1
by Neil Armstrong 07 May '21

07 May '21

This patchset adds Amlogic SM1 support on the TOACODEC driver, first by switching to regmap fields for some bit fields to avoid code duplication, and then by adding the corresponding bits & struct for the SM1 changed bits. Changes since v1 at [1]: - switch to regmap field [1] https://lore.kernel.org/r/20210429170147.3615883-1-narmstrong@baylibre.com Neil Armstrong (2): ASoC: meson: g12a-toacodec: use regmap fields to prepare SM1 support ASoC: meson: g12a-toacodec: add support for SM1 TOACODEC sound/soc/meson/g12a-toacodec.c | 144 +++++++++++++++++++++++++++----- 1 file changed, 125 insertions(+), 19 deletions(-) -- 2.25.1

2 5

[PATCH v5] ASoC: topology: Add support for multiple kcontrol types to a widget
by Jaska Uimonen 07 May '21

07 May '21

Current dapm widget has a single variable to describe its kcontrol's type. As there can be many kcontrols in one widget it is inherently presumed that the types are the same. Lately there has been use cases where different types of kcontrols would be needed for a single widget. Thus add pointer to dapm widget to hold an array for different kcontrol types and modify the kcontrol creation to operate in a loop based on individual kcontrol type. Change control creation and deletion to use individual kcontrol types in SOF driver. This is done in the same patch for not breaking bisect. SOF driver is also currently the only one using the dapm widget kcontrol_type. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Hi, This is a patch for enabling multiple different types of kcontrols for a dapm widget. It refactors the dapm kcontrol parsing by lifting the creation loop and memory allocation up one level and making the type variable to hold multiple type values. Sof driver is modified to use this type information in the same patch as it is quite cumbersome to split the changes nicely. changes since v4: - change title from dapm -> topology - remove cover letter - cc maintainers, me learning list behavior, apologies for the noise changes since v3: - fix sparse warnings for not using le32_to_cpu changes since v2: - only 1 patch for not breaking bisect - sof internal reviews -> minor changes + reviewed by tags - no upstream comments after v2 -> assuming I'm on the right track here Changes since v1: - don't add kcontrol type to core i.e. to kcontrol and kcontrol_news - extend existing dapm widget kcontrol_type into holding an array of types include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 467 +++++++++++++++++------------------ sound/soc/sof/topology.c | 15 +- 3 files changed, 234 insertions(+), 250 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 328cf763d9b4..4afd667e124c 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -54,7 +54,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ + unsigned int *kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 73076d425efb..2a10bf2cfb00 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1203,249 +1203,216 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, return ret; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_dmixer_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { - struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct snd_soc_tplg_mixer_control *mc; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (kc == NULL) - return NULL; - - for (i = 0; i < num_kcontrols; i++) { - mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; - - /* validate kcontrol */ - if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_sm; + int err; - sm = devm_kzalloc(tplg->dev, sizeof(*sm), GFP_KERNEL); - if (sm == NULL) - goto err_sm; + mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + - le32_to_cpu(mc->priv.size)); + /* validate kcontrol */ + if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", - mc->hdr.name, i); + sm = devm_kzalloc(tplg->dev, sizeof(*sm), GFP_KERNEL); + if (!sm) + return -ENOMEM; - kc[i].private_value = (long)sm; - kc[i].name = devm_kstrdup(tplg->dev, mc->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_sm; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(mc->hdr.access); + tplg->pos += sizeof(struct snd_soc_tplg_mixer_control) + + le32_to_cpu(mc->priv.size); - /* we only support FL/FR channel mapping atm */ - sm->reg = tplc_chan_get_reg(tplg, mc->channel, - SNDRV_CHMAP_FL); - sm->rreg = tplc_chan_get_reg(tplg, mc->channel, - SNDRV_CHMAP_FR); - sm->shift = tplc_chan_get_shift(tplg, mc->channel, - SNDRV_CHMAP_FL); - sm->rshift = tplc_chan_get_shift(tplg, mc->channel, - SNDRV_CHMAP_FR); + dev_dbg(tplg->dev, " adding DAPM widget mixer control %s\n", + mc->hdr.name); - sm->max = le32_to_cpu(mc->max); - sm->min = le32_to_cpu(mc->min); - sm->invert = le32_to_cpu(mc->invert); - sm->platform_max = le32_to_cpu(mc->platform_max); - sm->dobj.index = tplg->index; - INIT_LIST_HEAD(&sm->dobj.list); - - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&mc->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &mc->hdr, mc->hdr.name); - goto err_sm; - } + kc->private_value = (long)sm; + kc->name = devm_kstrdup(tplg->dev, mc->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(mc->hdr.access); + + /* we only support FL/FR channel mapping atm */ + sm->reg = tplc_chan_get_reg(tplg, mc->channel, + SNDRV_CHMAP_FL); + sm->rreg = tplc_chan_get_reg(tplg, mc->channel, + SNDRV_CHMAP_FR); + sm->shift = tplc_chan_get_shift(tplg, mc->channel, + SNDRV_CHMAP_FL); + sm->rshift = tplc_chan_get_shift(tplg, mc->channel, + SNDRV_CHMAP_FR); + + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); + sm->dobj.index = tplg->index; + INIT_LIST_HEAD(&sm->dobj.list); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&mc->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &mc->hdr, mc->hdr.name); + return err; + } - /* create any TLV data */ - err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", - mc->hdr.name); - goto err_sm; - } + /* create any TLV data */ + err = soc_tplg_create_tlv(tplg, kc, &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + return err; + } - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)mc); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - mc->hdr.name); - goto err_sm; - } + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)mc); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + mc->hdr.name); + return err; } - return kc; -err_sm: - return NULL; + return 0; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_denum_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { - struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (kc == NULL) - return NULL; - - for (i = 0; i < num_kcontrols; i++) { - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_se; + int err; - se = devm_kzalloc(tplg->dev, sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err_se; + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - le32_to_cpu(ec->priv.size)); + se = devm_kzalloc(tplg->dev, sizeof(*se), GFP_KERNEL); + if (!se) + return -ENOMEM; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + le32_to_cpu(ec->priv.size)); - kc[i].private_value = (long)se; - kc[i].name = devm_kstrdup(tplg->dev, ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_se; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(ec->hdr.access); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, - SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, - SNDRV_CHMAP_FR); + kc->private_value = (long)se; + kc->name = devm_kstrdup(tplg->dev, ec->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(ec->hdr.access); - se->items = le32_to_cpu(ec->items); - se->mask = le32_to_cpu(ec->mask); - se->dobj.index = tplg->index; + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - switch (le32_to_cpu(ec->hdr.ops.info)) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(tplg, se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); - goto err_se; - } - fallthrough; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(tplg, se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", - ec->hdr.name); - goto err_se; - } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); + se->dobj.index = tplg->index; - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; + switch (le32_to_cpu(ec->hdr.ops.info)) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(tplg, se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + return err; } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)ec); + fallthrough; + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(tplg, se, ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", ec->hdr.name); - goto err_se; + return err; } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); + return -EINVAL; } - return kc; + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + return err; + } -err_se: - return NULL; + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + ec->hdr.name); + return err; + } + + return 0; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_dbytes_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { struct snd_soc_tplg_bytes_control *be; struct soc_bytes_ext *sbe; - struct snd_kcontrol_new *kc; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (!kc) - return NULL; + int err; - for (i = 0; i < num_kcontrols; i++) { - be = (struct snd_soc_tplg_bytes_control *)tplg->pos; + be = (struct snd_soc_tplg_bytes_control *)tplg->pos; - /* validate kcontrol */ - if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_sbe; + /* validate kcontrol */ + if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - sbe = devm_kzalloc(tplg->dev, sizeof(*sbe), GFP_KERNEL); - if (sbe == NULL) - goto err_sbe; + sbe = devm_kzalloc(tplg->dev, sizeof(*sbe), GFP_KERNEL); + if (!sbe) + return -ENOMEM; - tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + - le32_to_cpu(be->priv.size)); + tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + + le32_to_cpu(be->priv.size)); - dev_dbg(tplg->dev, - "ASoC: adding bytes kcontrol %s with access 0x%x\n", - be->hdr.name, be->hdr.access); + dev_dbg(tplg->dev, + "ASoC: adding bytes kcontrol %s with access 0x%x\n", + be->hdr.name, be->hdr.access); - kc[i].private_value = (long)sbe; - kc[i].name = devm_kstrdup(tplg->dev, be->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_sbe; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(be->hdr.access); + kc->private_value = (long)sbe; + kc->name = devm_kstrdup(tplg->dev, be->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(be->hdr.access); - sbe->max = le32_to_cpu(be->max); - INIT_LIST_HEAD(&sbe->dobj.list); + sbe->max = le32_to_cpu(be->max); + INIT_LIST_HEAD(&sbe->dobj.list); - /* map standard io handlers and check for external handlers */ - err = soc_tplg_kcontrol_bind_io(&be->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &be->hdr, be->hdr.name); - goto err_sbe; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)be); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - be->hdr.name); - goto err_sbe; - } + /* map standard io handlers and check for external handlers */ + err = soc_tplg_kcontrol_bind_io(&be->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &be->hdr, be->hdr.name); + return err; } - return kc; - -err_sbe: + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)be); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + be->hdr.name); + return err; + } - return NULL; + return 0; } static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, @@ -1455,8 +1422,13 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; - unsigned int kcontrol_type; + unsigned int *kcontrol_type; + struct snd_kcontrol_new *kc; + int mixer_count = 0; + int bytes_count = 0; + int enum_count = 0; int ret = 0; + int i; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) @@ -1499,7 +1471,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, le32_to_cpu(w->priv.size)); if (w->num_kcontrols == 0) { - kcontrol_type = 0; template.num_kcontrols = 0; goto widget; } @@ -1508,57 +1479,66 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", w->name, w->num_kcontrols, control_hdr->type); - switch (le32_to_cpu(control_hdr->ops.info)) { - case SND_SOC_TPLG_CTL_VOLSW: - case SND_SOC_TPLG_CTL_STROBE: - case SND_SOC_TPLG_CTL_VOLSW_SX: - case SND_SOC_TPLG_CTL_VOLSW_XR_SX: - case SND_SOC_TPLG_CTL_RANGE: - case SND_SOC_TPLG_DAPM_CTL_VOLSW: - kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_dmixer_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; - goto hdr_err; - } - break; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; - goto hdr_err; - } - break; - case SND_SOC_TPLG_CTL_BYTES: - kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_dbytes_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; + template.num_kcontrols = le32_to_cpu(w->num_kcontrols); + kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); + if (!kc) + goto err; + + kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), + GFP_KERNEL); + if (!kcontrol_type) + goto err; + + for (i = 0; i < w->num_kcontrols; i++) { + control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + switch (le32_to_cpu(control_hdr->ops.info)) { + case SND_SOC_TPLG_CTL_VOLSW: + case SND_SOC_TPLG_CTL_STROBE: + case SND_SOC_TPLG_CTL_VOLSW_SX: + case SND_SOC_TPLG_CTL_VOLSW_XR_SX: + case SND_SOC_TPLG_CTL_RANGE: + case SND_SOC_TPLG_DAPM_CTL_VOLSW: + /* volume mixer */ + kc[i].index = mixer_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_MIXER; + mixer_count++; + ret = soc_tplg_dapm_widget_dmixer_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + /* enumerated mixer */ + kc[i].index = enum_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_ENUM; + enum_count++; + ret = soc_tplg_dapm_widget_denum_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + case SND_SOC_TPLG_CTL_BYTES: + /* bytes control */ + kc[i].index = bytes_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_BYTES; + bytes_count++; + ret = soc_tplg_dapm_widget_dbytes_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + default: + dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", + control_hdr->ops.get, control_hdr->ops.put, + le32_to_cpu(control_hdr->ops.info)); + ret = -EINVAL; goto hdr_err; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", - control_hdr->ops.get, control_hdr->ops.put, - le32_to_cpu(control_hdr->ops.info)); - ret = -EINVAL; - goto hdr_err; } + template.kcontrol_news = kc; + widget: ret = soc_tplg_widget_load(tplg, &template, w); if (ret < 0) @@ -1595,6 +1575,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, snd_soc_dapm_free_widget(widget); hdr_err: kfree(template.sname); + kfree(kc); err: kfree(template.name); return ret; diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 59abcfc9bd55..92d346bbd357 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1063,6 +1063,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, scontrol->min_volume_step = le32_to_cpu(mc->min); scontrol->max_volume_step = le32_to_cpu(mc->max); scontrol->num_channels = le32_to_cpu(mc->num_channels); + scontrol->control_data->index = kc->index; /* set cmd for mixer control */ if (le32_to_cpu(mc->max) == 1) { @@ -1140,7 +1141,7 @@ static int sof_control_load_enum(struct snd_soc_component *scomp, scontrol->comp_id = sdev->next_comp_id; scontrol->num_channels = le32_to_cpu(ec->num_channels); - + scontrol->control_data->index = kc->index; scontrol->cmd = SOF_CTRL_CMD_ENUM; dev_dbg(scomp->dev, "tplg: load kcontrol index %d chans %d comp_id %d\n", @@ -1188,6 +1189,7 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, scontrol->comp_id = sdev->next_comp_id; scontrol->cmd = SOF_CTRL_CMD_BINARY; + scontrol->control_data->index = kc->index; dev_dbg(scomp->dev, "tplg: load kcontrol index %d chans %d\n", scontrol->comp_id, scontrol->num_channels); @@ -2133,7 +2135,7 @@ static int sof_get_control_data(struct snd_soc_component *scomp, for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; - switch (widget->dobj.widget.kcontrol_type) { + switch (widget->dobj.widget.kcontrol_type[i]) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; wdata[i].control = sm->dobj.private; @@ -2147,8 +2149,8 @@ static int sof_get_control_data(struct snd_soc_component *scomp, wdata[i].control = se->dobj.private; break; default: - dev_err(scomp->dev, "error: unknown kcontrol type %d in widget %s\n", - widget->dobj.widget.kcontrol_type, + dev_err(scomp->dev, "error: unknown kcontrol type %u in widget %s\n", + widget->dobj.widget.kcontrol_type[i], widget->name); return -EINVAL; } @@ -2164,7 +2166,8 @@ static int sof_get_control_data(struct snd_soc_component *scomp, return -EINVAL; /* make sure data is valid - data can be updated at runtime */ - if (wdata[i].pdata->magic != SOF_ABI_MAGIC) + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && + wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; *size += wdata[i].pdata->size; @@ -2605,7 +2608,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, } for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; - switch (dobj->widget.kcontrol_type) { + switch (widget->dobj.widget.kcontrol_type[i]) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; scontrol = sm->dobj.private; -- 2.24.1

2 1

[PATCH v8 0/2] hardware-privacy-implementation-for-dell-laptop
by Perry Yuan 06 May '21

06 May '21

From: Perry Yuan <perry_yuan(a)dell.com> Hi All, This patch set is a new driver for Dell mobile platform that has the hardware privacy feature including micmute,camera mute. For micmute led control, the hotkey is Fn + F4, it is a hardware based mute state, and new privacy will prevent micphone void input from hardware layer, any application cannot get voice data when micmute activated. Camera mute use a new hardware design to control the camrea shutter. When video is muted, no OS application should be functionally able to capture images of the person/environment in front of the system Older history: [1]https://patchwork.kernel.org/project/platform-driver-x86/patch/2020122813… [2]https://patchwork.kernel.org/project/alsa-devel/patch/20201103125859.8759… [3]https://www.spinics.net/lists/alsa-devel/msg120537.html [4]https://github.com/thesofproject/linux/pull/2660 [5]https://github.com/thesofproject/linux/issues/2496 Perry Yuan (2): platform/x86: dell-privacy: Add support for Dell hardware privacy ASoC: rt715:add micmute led state control supports .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 13 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 394 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 25 ++ drivers/platform/x86/dell/dell-wmi.c | 9 +- sound/soc/codecs/rt715-sdca.c | 42 ++ sound/soc/codecs/rt715.c | 42 ++ 9 files changed, 591 insertions(+), 4 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h -- 2.25.1

1 0

[PATCH v4 0/1] Enable using multiple different type kcontrols for widgets
by Jaska Uimonen 06 May '21

06 May '21

Hi, This is a patch v4 for enabling multiple different types of kcontrols for a dapm widget. Currently asoc allows to define and parse multiple same type kcontrols for single widget. So you can have for example two volume controls in a widget but not one byte and one enum control. Lately we've seen couple of cases where different types of controls would be useful and alsa topology actually allows you to create these. This patch refactors the dapm kcontrol parsing by lifting the creation loop and memory allocation up one level and making the type variable to hold multiple type values. Sof driver is modified to use this type information in the same patch as it is quite cumbersome to split the changes nicely. changes since v3: - fix sparse warnings for not using le32_to_cpu Jaska Uimonen (1): ASoC: dapm: Add support for multiple kcontrol types to a widget include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 467 +++++++++++++++++------------------ sound/soc/sof/topology.c | 15 +- 3 files changed, 234 insertions(+), 250 deletions(-) -- 2.24.1

1 1

[PATCH] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Hui Wang 06 May '21

06 May '21

Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3998e1771805..b638fc2ef6f7 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1204,11 +1204,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.25.1

2 3

[PATCH AUTOSEL 4.4 16/19] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index af2ed774b552..63ed5b38b11f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1117,12 +1117,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1133,7 +1132,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1149,14 +1148,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1180,8 +1180,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 06/19] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index c253bdf92e36..e5611ee9f2ae 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 05/19] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 1a0c0d16a279..f4b164f19d30 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6912,7 +6912,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 04/19] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index dd6c9e6a1d53..4128c04fbfde 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5314,7 +5314,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 16/22] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 7899a2cdeb42..4a0ab620983d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1119,12 +1119,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1135,7 +1134,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1151,14 +1150,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1182,8 +1182,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1222,7 +1222,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 06/22] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index a76b1f147660..67bd75fbdc7e 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 05/22] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 9899ef4c7efa..a88a81fc638a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6912,7 +6912,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 04/22] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index b044dea3c815..9843954698f4 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5314,7 +5314,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 19/25] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index af6325c78292..17a66b4a46a2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1119,12 +1119,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1135,7 +1134,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1151,14 +1150,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1182,8 +1182,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1222,7 +1222,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 06/25] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index edd765e22377..f82fa5be7d33 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 05/25] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 343f533906ba..5bbbbba0817b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6913,7 +6913,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 04/25] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e41bb4100306..edd359772f1f 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5343,7 +5343,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 24/32] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 0b0f748bffbe..7e44ccae3bc8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1118,12 +1118,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1134,7 +1133,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1150,14 +1149,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1181,8 +1181,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1221,7 +1221,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 15/32] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 875c40eadf6ac6644c0f71842a4f30dd9968d281 ] The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210325221054.22714-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 62b4187e9f44..4ebc023f1507 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -509,6 +509,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 08/32] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index edd765e22377..f82fa5be7d33 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1761,7 +1761,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 07/32] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 11b5b5e0e058..5dfddade1bae 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6913,7 +6913,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 06/32] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index ba99ff0e93e0..a0797fc17d95 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5343,7 +5343,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 03/32] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit b7c7203a1f751348f35fc4bcb157572d303f7573 ] The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210312114850.13832-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d63d99776384..62b4187e9f44 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -473,6 +473,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.2

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[PATCH AUTOSEL 5.4 33/46] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..03e3e0aa25a2 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1115,12 +1115,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1131,7 +1130,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1147,14 +1146,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1178,8 +1178,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1218,7 +1218,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

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[PATCH AUTOSEL 5.4 29/46] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Sasha Levin 05 May '21

05 May '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 0c37e2eb6b83e375e8a654d01598292d5591fc65 ] When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to use "dev.power.power_state.event" instead of ALSA specific card power state variable. BugLink: https://github.com/thesofproject/linux/issues/2825 Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ce38b5d4670d..f620b402b309 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2567,7 +2567,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2772,7 +2772,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) -- 2.30.2

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[PATCH AUTOSEL 5.4 23/46] ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init()
by Sasha Levin 05 May '21

05 May '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit a122a116fc6d8fcf2f202dcd185173a54268f239 ] Current rsnd needs to call .prepare (P) for clock settings, .trigger for playback start (S) and stop (E). It should be called as below from SSI point of view. P -> S -> E -> P -> S -> E -> ... But, if you used MIXer, below case might happen (2) 1: P -> S ---> E -> ... 2: P ----> S -> ... (1) (3) P(1) setups clock, but E(2) resets it. and starts playback (3). In such case, it will reports "SSI parent/child should use same rate". rsnd_ssi_master_clk_start() which is the main function at (P) was called from rsnd_ssi_init() (= S) before, but was moved by below patch to rsnd_soc_dai_prepare() (= P) to avoid using clk_get_rate() which shouldn't be used under atomic context. commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Because of above patch, rsnd_ssi_master_clk_start() is now called at (P) which is for non atomic context. But (P) is assuming that spin lock is *not* used. One issue now is rsnd_ssi_master_clk_start() is checking ssi->xxx which should be protected by spin lock. After above patch, adg.c had below patch for other reasons. commit 06e8f5c842f2d ("ASoC: rsnd: don't call clk_get_rate() under atomic context") clk_get_rate() is used at probe() timing by this patch. In other words, rsnd_ssi_master_clk_start() is no longer using clk_get_rate() any more. This means we can call it from rsnd_ssi_init() (= S) again which is protected by spin lock. This patch re-move it to under spin lock, and solves 1. checking ssi->xxx without spin lock issue. 2. clk setting / device start / device stop race condition. Reported-by: Linh Phung T. Y. <linh.phung.jy(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/875z0x1jt5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 47d5ddb526f2..8926dd69e8b8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -507,10 +507,15 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + int ret; if (!rsnd_ssi_is_run_mods(mod, io)) return 0; + ret = rsnd_ssi_master_clk_start(mod, io); + if (ret < 0) + return ret; + ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -1060,13 +1065,6 @@ static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_prepare(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - return rsnd_ssi_master_clk_start(mod, io); -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, @@ -1079,7 +1077,6 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; @@ -1166,7 +1163,6 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .pcm_new = rsnd_ssi_pcm_new, .fallback = rsnd_ssi_fallback, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; -- 2.30.2

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[PATCH AUTOSEL 5.4 17/46] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 875c40eadf6ac6644c0f71842a4f30dd9968d281 ] The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210325221054.22714-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 006cf1e8b602..46a81d4f0b2d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -512,6 +512,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

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[PATCH AUTOSEL 5.4 10/46] ALSA: bebob: enable to deliver MIDI messages for multiple ports
by Sasha Levin 05 May '21

05 May '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d2b6f15bc18ac8fbce25398290774c21f5b2cd44 ] Current implementation of bebob driver doesn't correctly handle the case that the device has multiple MIDI ports. The cause is the number of MIDI conformant data channels is passed to AM824 data block processing layer. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210321032831.340278-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index ce07ea0d4e71..3935e90c8e8f 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -534,20 +534,22 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { - struct snd_bebob_stream_formation *formation; + unsigned int pcm_channels; + unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { - formation = bebob->tx_stream_formations + index; + pcm_channels = bebob->tx_stream_formations[index].pcm; + midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { - formation = bebob->rx_stream_formations + index; + pcm_channels = bebob->rx_stream_formations[index].pcm; + midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } - err = amdtp_am824_set_parameters(stream, rate, formation->pcm, - formation->midi, false); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; -- 2.30.2

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[PATCH AUTOSEL 5.4 09/46] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 4c851f8dcaf8..73ad6e74aac9 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1745,7 +1745,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.4 08/46] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 81a6f4b2bd3c..e34f07c9ff47 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6889,7 +6889,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.4 07/46] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 5cbdc9be9c7e..c7b3e76ea2d2 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5326,7 +5326,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.4 05/46] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by Sasha Levin 05 May '21

05 May '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> [ Upstream commit 19c6a63ced5e07e40f3a5255cb1f0fe0d3be7b14 ] snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Link: https://lore.kernel.org/r/1615870055-13954-1-git-send-email-mikhail_durnev@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a6c1cf987e6e..df8d7b53b760 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1426,8 +1426,75 @@ static int rsnd_hw_params(struct snd_pcm_substream *substream, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } ret = rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 03/46] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit b7c7203a1f751348f35fc4bcb157572d303f7573 ] The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210312114850.13832-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index cfd307717473..006cf1e8b602 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -476,6 +476,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 61/85] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 5fb9653d9131..8ae2e2eaad3d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1117,12 +1117,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1133,7 +1132,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1149,14 +1148,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1180,8 +1180,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 57/85] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Sasha Levin 05 May '21

05 May '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 0c37e2eb6b83e375e8a654d01598292d5591fc65 ] When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to use "dev.power.power_state.event" instead of ALSA specific card power state variable. BugLink: https://github.com/thesofproject/linux/issues/2825 Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8c6f10cbced3..6d2a4dfcfe43 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2653,7 +2653,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2839,7 +2839,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 56/85] ASoC: Intel: sof_sdw: add quirk for new ADL-P Rvp
by Sasha Levin 05 May '21

05 May '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> [ Upstream commit d25bbe80485f8bcbbeb91a2a6cd8798c124b27b7 ] Add quirks for jack detection, rt711 DAI and DMIC Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210415175013.192862-6-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1d7677376e74..9dc982c2c776 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -187,6 +187,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + /* AlderLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alder Lake Client Platform"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | + SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC), + }, {} }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 55/85] ALSA: hda/realtek: Add quirk for Lenovo Ideapad S740
by Sasha Levin 05 May '21

05 May '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 26928ca1f06aab4361eb5adbe7ef3b5c82f13cf2 ] Lenovo Ideapad S740 requires quite a few COEF setups to make its speakers working. The verb table was provided from Ryan Prescott as the result of investigation via qemu: https://github.com/ryanprescott/realtek-verb-tools/wiki/How-to-sniff-verbs-… BugLink: https://github.com/thesofproject/linux/issues/2748 Tested-by: Ryan Prescott <ryan(a)cousinscomputers.net> Link: https://lore.kernel.org/r/20210416081211.20059-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/ideapad_s740_helper.c | 492 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 11 + 2 files changed, 503 insertions(+) create mode 100644 sound/pci/hda/ideapad_s740_helper.c diff --git a/sound/pci/hda/ideapad_s740_helper.c b/sound/pci/hda/ideapad_s740_helper.c new file mode 100644 index 000000000000..564b9086e52d --- /dev/null +++ b/sound/pci/hda/ideapad_s740_helper.c @@ -0,0 +1,492 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Ideapad S740, to be included from codec driver */ + +static const struct hda_verb alc285_ideapad_s740_coefs[] = { +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0320 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{} +}; + +static void alc285_fixup_ideapad_s740_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, alc285_ideapad_s740_coefs); + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..68201bd05ee5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6232,6 +6232,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc285_fixup_ideapad_s740_coef() */ +#include "ideapad_s740_helper.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6427,6 +6430,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_IDEAPAD_S740_COEF, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7905,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_IDEAPAD_S740_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8244,6 +8254,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), 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), -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 49/85] ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init()
by Sasha Levin 05 May '21

05 May '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit a122a116fc6d8fcf2f202dcd185173a54268f239 ] Current rsnd needs to call .prepare (P) for clock settings, .trigger for playback start (S) and stop (E). It should be called as below from SSI point of view. P -> S -> E -> P -> S -> E -> ... But, if you used MIXer, below case might happen (2) 1: P -> S ---> E -> ... 2: P ----> S -> ... (1) (3) P(1) setups clock, but E(2) resets it. and starts playback (3). In such case, it will reports "SSI parent/child should use same rate". rsnd_ssi_master_clk_start() which is the main function at (P) was called from rsnd_ssi_init() (= S) before, but was moved by below patch to rsnd_soc_dai_prepare() (= P) to avoid using clk_get_rate() which shouldn't be used under atomic context. commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Because of above patch, rsnd_ssi_master_clk_start() is now called at (P) which is for non atomic context. But (P) is assuming that spin lock is *not* used. One issue now is rsnd_ssi_master_clk_start() is checking ssi->xxx which should be protected by spin lock. After above patch, adg.c had below patch for other reasons. commit 06e8f5c842f2d ("ASoC: rsnd: don't call clk_get_rate() under atomic context") clk_get_rate() is used at probe() timing by this patch. In other words, rsnd_ssi_master_clk_start() is no longer using clk_get_rate() any more. This means we can call it from rsnd_ssi_init() (= S) again which is protected by spin lock. This patch re-move it to under spin lock, and solves 1. checking ssi->xxx without spin lock issue. 2. clk setting / device start / device stop race condition. Reported-by: Linh Phung T. Y. <linh.phung.jy(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/875z0x1jt5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d0ded427a836..a2f8138d40c7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -507,10 +507,15 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + int ret; if (!rsnd_ssi_is_run_mods(mod, io)) return 0; + ret = rsnd_ssi_master_clk_start(mod, io); + if (ret < 0) + return ret; + ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -1060,13 +1065,6 @@ static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_prepare(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - return rsnd_ssi_master_clk_start(mod, io); -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, @@ -1079,7 +1077,6 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; @@ -1166,7 +1163,6 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .pcm_new = rsnd_ssi_pcm_new, .fallback = rsnd_ssi_fallback, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 33/85] ASoC: rt5670: Add a quirk for the Dell Venue 10 Pro 5055
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 84cb0d5581b6a7bd5d96013f67e9f2eb0c7b4378 ] Add a quirk with the jack-detect and dmic settings necessary to make jack-detect and the builtin mic work on Dell Venue 10 Pro 5055 tablets. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210402140747.174716-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a0c8f58d729b..47ce074289ca 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2908,6 +2908,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_GPIO1_IS_IRQ | RT5670_JD_MODE3), }, + { + .callback = rt5670_quirk_cb, + .ident = "Dell Venue 10 Pro 5055", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 10 Pro 5055"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_GPIO1_IS_IRQ | + RT5670_JD_MODE1), + }, { .callback = rt5670_quirk_cb, .ident = "Aegex 10 tablet (RU2)", -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 29/85] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 875c40eadf6ac6644c0f71842a4f30dd9968d281 ] The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210325221054.22714-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 2d887406ca85..1ef0464249d1 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -514,6 +514,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 17/85] ALSA: bebob: enable to deliver MIDI messages for multiple ports
by Sasha Levin 05 May '21

05 May '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d2b6f15bc18ac8fbce25398290774c21f5b2cd44 ] Current implementation of bebob driver doesn't correctly handle the case that the device has multiple MIDI ports. The cause is the number of MIDI conformant data channels is passed to AM824 data block processing layer. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210321032831.340278-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index bbae04793c50..c18017e0a3d9 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -517,20 +517,22 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { - struct snd_bebob_stream_formation *formation; + unsigned int pcm_channels; + unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { - formation = bebob->tx_stream_formations + index; + pcm_channels = bebob->tx_stream_formations[index].pcm; + midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { - formation = bebob->rx_stream_formations + index; + pcm_channels = bebob->rx_stream_formations[index].pcm; + midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } - err = amdtp_am824_set_parameters(stream, rate, formation->pcm, - formation->midi, false); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 16/85] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 7ab10028d9fa..8def24673f35 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1740,7 +1740,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 15/85] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4a1f576dd9cf..51c3c6a08a1c 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6891,7 +6891,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 14/85] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..4aee30db034d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5321,7 +5321,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 10/85] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by Sasha Levin 05 May '21

05 May '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> [ Upstream commit 19c6a63ced5e07e40f3a5255cb1f0fe0d3be7b14 ] snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Link: https://lore.kernel.org/r/1615870055-13954-1-git-send-email-mikhail_durnev@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3e92a0..289928d4c0c9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,75 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.30.2

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[PATCH AUTOSEL 5.10 07/85] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit b7c7203a1f751348f35fc4bcb157572d303f7573 ] The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210312114850.13832-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d5812e73eb63..2d887406ca85 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -478,6 +478,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 080/104] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 05 May '21

05 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 5fb9653d9131..8ae2e2eaad3d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1117,12 +1117,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1133,7 +1132,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1149,14 +1148,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1180,8 +1180,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 073/104] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Sasha Levin 05 May '21

05 May '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 0c37e2eb6b83e375e8a654d01598292d5591fc65 ] When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to use "dev.power.power_state.event" instead of ALSA specific card power state variable. BugLink: https://github.com/thesofproject/linux/issues/2825 Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d6387106619f..7b0d9d7a1c38 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2650,7 +2650,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2836,7 +2836,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) -- 2.30.2

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[PATCH AUTOSEL 5.11 072/104] ASoC: Intel: sof_sdw: add quirk for new ADL-P Rvp
by Sasha Levin 05 May '21

05 May '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> [ Upstream commit d25bbe80485f8bcbbeb91a2a6cd8798c124b27b7 ] Add quirks for jack detection, rt711 DAI and DMIC Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210415175013.192862-6-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1d7677376e74..9dc982c2c776 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -187,6 +187,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + /* AlderLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alder Lake Client Platform"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | + SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC), + }, {} }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 071/104] ALSA: hda/realtek: Add quirk for Lenovo Ideapad S740
by Sasha Levin 05 May '21

05 May '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 26928ca1f06aab4361eb5adbe7ef3b5c82f13cf2 ] Lenovo Ideapad S740 requires quite a few COEF setups to make its speakers working. The verb table was provided from Ryan Prescott as the result of investigation via qemu: https://github.com/ryanprescott/realtek-verb-tools/wiki/How-to-sniff-verbs-… BugLink: https://github.com/thesofproject/linux/issues/2748 Tested-by: Ryan Prescott <ryan(a)cousinscomputers.net> Link: https://lore.kernel.org/r/20210416081211.20059-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/ideapad_s740_helper.c | 492 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 11 + 2 files changed, 503 insertions(+) create mode 100644 sound/pci/hda/ideapad_s740_helper.c diff --git a/sound/pci/hda/ideapad_s740_helper.c b/sound/pci/hda/ideapad_s740_helper.c new file mode 100644 index 000000000000..564b9086e52d --- /dev/null +++ b/sound/pci/hda/ideapad_s740_helper.c @@ -0,0 +1,492 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Ideapad S740, to be included from codec driver */ + +static const struct hda_verb alc285_ideapad_s740_coefs[] = { +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0320 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{} +}; + +static void alc285_fixup_ideapad_s740_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, alc285_ideapad_s740_coefs); + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..68201bd05ee5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6232,6 +6232,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc285_fixup_ideapad_s740_coef() */ +#include "ideapad_s740_helper.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6427,6 +6430,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_IDEAPAD_S740_COEF, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7905,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_IDEAPAD_S740_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8244,6 +8254,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), 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), -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 065/104] ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init()
by Sasha Levin 05 May '21

05 May '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit a122a116fc6d8fcf2f202dcd185173a54268f239 ] Current rsnd needs to call .prepare (P) for clock settings, .trigger for playback start (S) and stop (E). It should be called as below from SSI point of view. P -> S -> E -> P -> S -> E -> ... But, if you used MIXer, below case might happen (2) 1: P -> S ---> E -> ... 2: P ----> S -> ... (1) (3) P(1) setups clock, but E(2) resets it. and starts playback (3). In such case, it will reports "SSI parent/child should use same rate". rsnd_ssi_master_clk_start() which is the main function at (P) was called from rsnd_ssi_init() (= S) before, but was moved by below patch to rsnd_soc_dai_prepare() (= P) to avoid using clk_get_rate() which shouldn't be used under atomic context. commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Because of above patch, rsnd_ssi_master_clk_start() is now called at (P) which is for non atomic context. But (P) is assuming that spin lock is *not* used. One issue now is rsnd_ssi_master_clk_start() is checking ssi->xxx which should be protected by spin lock. After above patch, adg.c had below patch for other reasons. commit 06e8f5c842f2d ("ASoC: rsnd: don't call clk_get_rate() under atomic context") clk_get_rate() is used at probe() timing by this patch. In other words, rsnd_ssi_master_clk_start() is no longer using clk_get_rate() any more. This means we can call it from rsnd_ssi_init() (= S) again which is protected by spin lock. This patch re-move it to under spin lock, and solves 1. checking ssi->xxx without spin lock issue. 2. clk setting / device start / device stop race condition. Reported-by: Linh Phung T. Y. <linh.phung.jy(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/875z0x1jt5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d0ded427a836..a2f8138d40c7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -507,10 +507,15 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + int ret; if (!rsnd_ssi_is_run_mods(mod, io)) return 0; + ret = rsnd_ssi_master_clk_start(mod, io); + if (ret < 0) + return ret; + ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -1060,13 +1065,6 @@ static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_prepare(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - return rsnd_ssi_master_clk_start(mod, io); -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, @@ -1079,7 +1077,6 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; @@ -1166,7 +1163,6 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .pcm_new = rsnd_ssi_pcm_new, .fallback = rsnd_ssi_fallback, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 048/104] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by Sasha Levin 05 May '21

05 May '21

From: Gyeongtaek Lee <gt82.lee(a)samsung.com> [ Upstream commit 45475bf60cc1d42da229a0aa757180c88bab8d22 ] If panic_on_warn=1 is added in bootargs and compress offload playback with DPCM is started, kernel panic would be occurred because rtd->card->pcm_mutex isn't held in soc_compr_open_fe() and soc_compr_free_fe() and it generates lockdep warning in the following code. void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { struct snd_soc_dai *dai; int i; lockdep_assert_held(&rtd->card->pcm_mutex); To prevent lockdep warning but minimize side effect by adding mutex, pcm_mutex is held just before snd_soc_runtime_activate() and snd_soc_runtime_deactivate() and is released right after them. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Link: https://lore.kernel.org/r/1891546521.01617772502282.JavaMail.epsvc@epcpadp3 Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 246a5e32e22a..b4810266f5e5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -153,7 +153,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -181,7 +183,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 039/104] ASoC: rt5670: Add a quirk for the Dell Venue 10 Pro 5055
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 84cb0d5581b6a7bd5d96013f67e9f2eb0c7b4378 ] Add a quirk with the jack-detect and dmic settings necessary to make jack-detect and the builtin mic work on Dell Venue 10 Pro 5055 tablets. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210402140747.174716-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index a0c8f58d729b..47ce074289ca 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2908,6 +2908,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_GPIO1_IS_IRQ | RT5670_JD_MODE3), }, + { + .callback = rt5670_quirk_cb, + .ident = "Dell Venue 10 Pro 5055", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 10 Pro 5055"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_GPIO1_IS_IRQ | + RT5670_JD_MODE1), + }, { .callback = rt5670_quirk_cb, .ident = "Aegex 10 tablet (RU2)", -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 034/104] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 875c40eadf6ac6644c0f71842a4f30dd9968d281 ] The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210325221054.22714-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 07730ed7ab3c..d45f43290653 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -514,6 +514,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 022/104] ALSA: bebob: enable to deliver MIDI messages for multiple ports
by Sasha Levin 05 May '21

05 May '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d2b6f15bc18ac8fbce25398290774c21f5b2cd44 ] Current implementation of bebob driver doesn't correctly handle the case that the device has multiple MIDI ports. The cause is the number of MIDI conformant data channels is passed to AM824 data block processing layer. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210321032831.340278-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index bbae04793c50..c18017e0a3d9 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -517,20 +517,22 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { - struct snd_bebob_stream_formation *formation; + unsigned int pcm_channels; + unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { - formation = bebob->tx_stream_formations + index; + pcm_channels = bebob->tx_stream_formations[index].pcm; + midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { - formation = bebob->rx_stream_formations + index; + pcm_channels = bebob->rx_stream_formations[index].pcm; + midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } - err = amdtp_am824_set_parameters(stream, rate, formation->pcm, - formation->midi, false); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; -- 2.30.2

1 0

[PATCH AUTOSEL 5.11 021/104] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 012fbec5e6a7..0f4ab86a29f6 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1733,7 +1733,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.11 020/104] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 04e878a0f773..49fee31ad905 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6884,7 +6884,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.11 019/104] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..4aee30db034d 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5321,7 +5321,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.11 015/104] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by Sasha Levin 05 May '21

05 May '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> [ Upstream commit 19c6a63ced5e07e40f3a5255cb1f0fe0d3be7b14 ] snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Link: https://lore.kernel.org/r/1615870055-13954-1-git-send-email-mikhail_durnev@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6e670b3e92a0..289928d4c0c9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,75 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.30.2

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[PATCH AUTOSEL 5.11 012/104] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit b7c7203a1f751348f35fc4bcb157572d303f7573 ] The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210312114850.13832-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 21d2e1cba380..07730ed7ab3c 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -478,6 +478,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.2

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[PATCH AUTOSEL 5.11 002/104] ALSA: usb-audio: Add Pioneer DJM-850 to quirks-table
by Sasha Levin 05 May '21

05 May '21

From: Nicolas MURE <nicolas.mure2019(a)gmail.com> [ Upstream commit a3c30b0cb6d05f5bf66d1a5d42e876f31753a447 ] Declare the Pioneer DJM-850 interfaces for capture and playback. See https://github.com/nm2107/Pioneer-DJM-850-driver-reverse-engineering/blob/1… for the complete device spec. Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com> Link: https://lore.kernel.org/r/20210301152729.18094-2-nicolas.mure2019@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 63 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1165a5ac60f2..9716a9f7c095 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3817,6 +3817,69 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-850 + * 8 channels playback and 8 channels capture @ 44.1/48/96kHz S24LE + * Playback on EP 0x05 + * Capture on EP 0x86 + */ + USB_DEVICE_VENDOR_SPEC(0x08e4, 0x0163), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_DATA, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_DATA, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-450 -- 2.30.2

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[PATCH AUTOSEL 5.12 082/116] ALSA: hda/hdmi: fix race in handling acomp ELD notification at resume
by Sasha Levin 05 May '21

05 May '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 0c37e2eb6b83e375e8a654d01598292d5591fc65 ] When snd-hda-codec-hdmi is used with ASoC HDA controller like SOF (acomp used for ELD notifications), display connection change done during suspend, can be lost due to following sequence of events: 1. system in S3 suspend 2. DP/HDMI receiver connected 3. system resumed 4. HDA controller resumed, but card->deferred_resume_work not complete 5. acomp eld_notify callback 6. eld_notify ignored as power state is not CTL_POWER_D0 7. HDA resume deferred work completed, power state set to CTL_POWER_D0 This results in losing the notification, and the jack state reported to user-space is not correct. The check on step 6 was added in commit 8ae743e82f0b ("ALSA: hda - Skip ELD notification during system suspend"). It would seem with the deferred resume logic in ASoC core, this check is not safe. Fix the issue by modifying the check to use "dev.power.power_state.event" instead of ALSA specific card power state variable. BugLink: https://github.com/thesofproject/linux/issues/2825 Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20210416131157.1881366-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5de3666a7101..4b2cc8cb55c4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2654,7 +2654,7 @@ static void generic_acomp_pin_eld_notify(void *audio_ptr, int port, int dev_id) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) @@ -2840,7 +2840,7 @@ static void intel_pin_eld_notify(void *audio_ptr, int port, int pipe) /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ - if (snd_power_get_state(codec->card) != SNDRV_CTL_POWER_D0) + if (codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND) return; /* ditto during suspend/resume process itself */ if (snd_hdac_is_in_pm(&codec->core)) -- 2.30.2

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[PATCH AUTOSEL 5.12 081/116] ASoC: Intel: sof_sdw: add quirk for new ADL-P Rvp
by Sasha Levin 05 May '21

05 May '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> [ Upstream commit d25bbe80485f8bcbbeb91a2a6cd8798c124b27b7 ] Add quirks for jack detection, rt711 DAI and DMIC Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210415175013.192862-6-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 8adce6417b02..ecd3f90f4bbe 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -187,6 +187,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + /* AlderLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Alder Lake Client Platform"), + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD1 | + SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC), + }, {} }; -- 2.30.2

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[PATCH AUTOSEL 5.12 080/116] ALSA: hda/realtek: Add quirk for Lenovo Ideapad S740
by Sasha Levin 05 May '21

05 May '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 26928ca1f06aab4361eb5adbe7ef3b5c82f13cf2 ] Lenovo Ideapad S740 requires quite a few COEF setups to make its speakers working. The verb table was provided from Ryan Prescott as the result of investigation via qemu: https://github.com/ryanprescott/realtek-verb-tools/wiki/How-to-sniff-verbs-… BugLink: https://github.com/thesofproject/linux/issues/2748 Tested-by: Ryan Prescott <ryan(a)cousinscomputers.net> Link: https://lore.kernel.org/r/20210416081211.20059-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/ideapad_s740_helper.c | 492 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 11 + 2 files changed, 503 insertions(+) create mode 100644 sound/pci/hda/ideapad_s740_helper.c diff --git a/sound/pci/hda/ideapad_s740_helper.c b/sound/pci/hda/ideapad_s740_helper.c new file mode 100644 index 000000000000..564b9086e52d --- /dev/null +++ b/sound/pci/hda/ideapad_s740_helper.c @@ -0,0 +1,492 @@ +// SPDX-License-Identifier: GPL-2.0 +/* Fixes for Lenovo Ideapad S740, to be included from codec driver */ + +static const struct hda_verb alc285_ideapad_s740_coefs[] = { +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x10 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0320 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0041 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001d }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004e }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0042 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x007f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x003c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0011 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x002a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x000f }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0046 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0044 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0003 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0009 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x004c }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001b }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0019 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0025 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0018 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0037 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x001a }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0040 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0016 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0076 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0017 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0010 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0015 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0007 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0086 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0001 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x29 }, +{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0002 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0x0000 }, +{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, +{} +}; + +static void alc285_fixup_ideapad_s740_coef(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_add_verbs(codec, alc285_ideapad_s740_coefs); + break; + } +} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..68201bd05ee5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6232,6 +6232,9 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" +/* for alc285_fixup_ideapad_s740_coef() */ +#include "ideapad_s740_helper.c" + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6427,6 +6430,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_IDEAPAD_S740_COEF, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7905,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_IDEAPAD_S740_COEF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8244,6 +8254,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), + SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), 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), -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 073/116] ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init()
by Sasha Levin 05 May '21

05 May '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit a122a116fc6d8fcf2f202dcd185173a54268f239 ] Current rsnd needs to call .prepare (P) for clock settings, .trigger for playback start (S) and stop (E). It should be called as below from SSI point of view. P -> S -> E -> P -> S -> E -> ... But, if you used MIXer, below case might happen (2) 1: P -> S ---> E -> ... 2: P ----> S -> ... (1) (3) P(1) setups clock, but E(2) resets it. and starts playback (3). In such case, it will reports "SSI parent/child should use same rate". rsnd_ssi_master_clk_start() which is the main function at (P) was called from rsnd_ssi_init() (= S) before, but was moved by below patch to rsnd_soc_dai_prepare() (= P) to avoid using clk_get_rate() which shouldn't be used under atomic context. commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic") Because of above patch, rsnd_ssi_master_clk_start() is now called at (P) which is for non atomic context. But (P) is assuming that spin lock is *not* used. One issue now is rsnd_ssi_master_clk_start() is checking ssi->xxx which should be protected by spin lock. After above patch, adg.c had below patch for other reasons. commit 06e8f5c842f2d ("ASoC: rsnd: don't call clk_get_rate() under atomic context") clk_get_rate() is used at probe() timing by this patch. In other words, rsnd_ssi_master_clk_start() is no longer using clk_get_rate() any more. This means we can call it from rsnd_ssi_init() (= S) again which is protected by spin lock. This patch re-move it to under spin lock, and solves 1. checking ssi->xxx without spin lock issue. 2. clk setting / device start / device stop race condition. Reported-by: Linh Phung T. Y. <linh.phung.jy(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/875z0x1jt5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index d0ded427a836..a2f8138d40c7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -507,10 +507,15 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, struct rsnd_priv *priv) { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); + int ret; if (!rsnd_ssi_is_run_mods(mod, io)) return 0; + ret = rsnd_ssi_master_clk_start(mod, io); + if (ret < 0) + return ret; + ssi->usrcnt++; rsnd_mod_power_on(mod); @@ -1060,13 +1065,6 @@ static int rsnd_ssi_pio_pointer(struct rsnd_mod *mod, return 0; } -static int rsnd_ssi_prepare(struct rsnd_mod *mod, - struct rsnd_dai_stream *io, - struct rsnd_priv *priv) -{ - return rsnd_ssi_master_clk_start(mod, io); -} - static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .name = SSI_NAME, .probe = rsnd_ssi_common_probe, @@ -1079,7 +1077,6 @@ static struct rsnd_mod_ops rsnd_ssi_pio_ops = { .pointer = rsnd_ssi_pio_pointer, .pcm_new = rsnd_ssi_pcm_new, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; @@ -1166,7 +1163,6 @@ static struct rsnd_mod_ops rsnd_ssi_dma_ops = { .pcm_new = rsnd_ssi_pcm_new, .fallback = rsnd_ssi_fallback, .hw_params = rsnd_ssi_hw_params, - .prepare = rsnd_ssi_prepare, .get_status = rsnd_ssi_get_status, }; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 051/116] ASoC: soc-compress: lock pcm_mutex to resolve lockdep error
by Sasha Levin 05 May '21

05 May '21

From: Gyeongtaek Lee <gt82.lee(a)samsung.com> [ Upstream commit 45475bf60cc1d42da229a0aa757180c88bab8d22 ] If panic_on_warn=1 is added in bootargs and compress offload playback with DPCM is started, kernel panic would be occurred because rtd->card->pcm_mutex isn't held in soc_compr_open_fe() and soc_compr_free_fe() and it generates lockdep warning in the following code. void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { struct snd_soc_dai *dai; int i; lockdep_assert_held(&rtd->card->pcm_mutex); To prevent lockdep warning but minimize side effect by adding mutex, pcm_mutex is held just before snd_soc_runtime_activate() and snd_soc_runtime_deactivate() and is released right after them. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Link: https://lore.kernel.org/r/1891546521.01617772502282.JavaMail.epsvc@epcpadp3 Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 246a5e32e22a..b4810266f5e5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -153,7 +153,9 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_activate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); mutex_unlock(&fe->card->mutex); @@ -181,7 +183,9 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + mutex_lock_nested(&fe->card->pcm_mutex, fe->card->pcm_subclass); snd_soc_runtime_deactivate(fe, stream); + mutex_unlock(&fe->card->pcm_mutex); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 042/116] ASoC: rt5670: Add a quirk for the Dell Venue 10 Pro 5055
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 84cb0d5581b6a7bd5d96013f67e9f2eb0c7b4378 ] Add a quirk with the jack-detect and dmic settings necessary to make jack-detect and the builtin mic work on Dell Venue 10 Pro 5055 tablets. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210402140747.174716-5-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 4063aac2a443..dd69d874bad2 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2980,6 +2980,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_GPIO1_IS_IRQ | RT5670_JD_MODE3), }, + { + .callback = rt5670_quirk_cb, + .ident = "Dell Venue 10 Pro 5055", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), + DMI_MATCH(DMI_PRODUCT_NAME, "Venue 10 Pro 5055"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_GPIO1_IS_IRQ | + RT5670_JD_MODE1), + }, { .callback = rt5670_quirk_cb, .ident = "Aegex 10 tablet (RU2)", -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 037/116] ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 875c40eadf6ac6644c0f71842a4f30dd9968d281 ] The Chuwi Hi8 tablet is using an analog mic on IN1 and has its jack-detect connected to JD2_IN4N, instead of using the default IN3 for its internal mic and JD1_IN4P for jack-detect. It also only has 1 speaker. Add a quirk applying the correct settings for this configuration. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20210325221054.22714-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 19faababb78d..22912cab5e63 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -518,6 +518,23 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + /* Chuwi Hi8 (CWI509) */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Hampoo"), + DMI_MATCH(DMI_BOARD_NAME, "BYT-PA03C"), + DMI_MATCH(DMI_SYS_VENDOR, "ilife"), + DMI_MATCH(DMI_PRODUCT_NAME, "S806"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Circuitco"), -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 030/116] ALSA: hda/hdmi: fix max DP-MST dev_num for Intel TGL+ platforms
by Sasha Levin 05 May '21

05 May '21

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit e839fbed26e8b8713803b8ac73da92fd2b0c7594 ] Increase the device select range to 4 on platforms supporting 4 concurrent displays. This fixes a problem in scenario where total of 4 displays are active, and 3 of these are audio capable DP receivers and connected to a DP-MST hub. Due to incorrect range for device select, audio could not be played to the 3rd monitor in DP-MST hub. BugLink: https://github.com/thesofproject/linux/issues/2798 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210324172337.51730-1-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 25 +++++++++++-------------- 1 file changed, 11 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 45ae845e82df..5de3666a7101 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1848,16 +1848,12 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) */ if (spec->intel_hsw_fixup) { /* - * On Intel platforms, device entries number is - * changed dynamically. If there is a DP MST - * hub connected, the device entries number is 3. - * Otherwise, it is 1. - * Here we manually set dev_num to 3, so that - * we can initialize all the device entries when - * bootup statically. + * On Intel platforms, device entries count returned + * by AC_PAR_DEVLIST_LEN is dynamic, and depends on + * the type of receiver that is connected. Allocate pin + * structures based on worst case. */ - dev_num = 3; - spec->dev_num = 3; + dev_num = spec->dev_num; } else if (spec->dyn_pcm_assign && codec->dp_mst) { dev_num = snd_hda_get_num_devices(codec, pin_nid) + 1; /* @@ -2942,7 +2938,7 @@ static int parse_intel_hdmi(struct hda_codec *codec) /* Intel Haswell and onwards; audio component with eld notifier */ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, - const int *port_map, int port_num) + const int *port_map, int port_num, int dev_num) { struct hdmi_spec *spec; int err; @@ -2957,6 +2953,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->port_map = port_map; spec->port_num = port_num; spec->intel_hsw_fixup = true; + spec->dev_num = dev_num; intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); @@ -2982,12 +2979,12 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, static int patch_i915_hsw_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x08, NULL, 0); + return intel_hsw_common_init(codec, 0x08, NULL, 0, 3); } static int patch_i915_glk_hdmi(struct hda_codec *codec) { - return intel_hsw_common_init(codec, 0x0b, NULL, 0); + return intel_hsw_common_init(codec, 0x0b, NULL, 0, 3); } static int patch_i915_icl_hdmi(struct hda_codec *codec) @@ -2998,7 +2995,7 @@ static int patch_i915_icl_hdmi(struct hda_codec *codec) */ static const int map[] = {0x0, 0x4, 0x6, 0x8, 0xa, 0xb}; - return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + return intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 3); } static int patch_i915_tgl_hdmi(struct hda_codec *codec) @@ -3010,7 +3007,7 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) static const int map[] = {0x4, 0x6, 0x8, 0xa, 0xb, 0xc, 0xd, 0xe, 0xf}; int ret; - ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map)); + ret = intel_hsw_common_init(codec, 0x02, map, ARRAY_SIZE(map), 4); if (!ret) { struct hdmi_spec *spec = codec->spec; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 022/116] ALSA: bebob: enable to deliver MIDI messages for multiple ports
by Sasha Levin 05 May '21

05 May '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d2b6f15bc18ac8fbce25398290774c21f5b2cd44 ] Current implementation of bebob driver doesn't correctly handle the case that the device has multiple MIDI ports. The cause is the number of MIDI conformant data channels is passed to AM824 data block processing layer. This commit fixes the bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210321032831.340278-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/bebob/bebob_stream.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index bbae04793c50..c18017e0a3d9 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -517,20 +517,22 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) static int keep_resources(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate, unsigned int index) { - struct snd_bebob_stream_formation *formation; + unsigned int pcm_channels; + unsigned int midi_ports; struct cmp_connection *conn; int err; if (stream == &bebob->tx_stream) { - formation = bebob->tx_stream_formations + index; + pcm_channels = bebob->tx_stream_formations[index].pcm; + midi_ports = bebob->midi_input_ports; conn = &bebob->out_conn; } else { - formation = bebob->rx_stream_formations + index; + pcm_channels = bebob->rx_stream_formations[index].pcm; + midi_ports = bebob->midi_output_ports; conn = &bebob->in_conn; } - err = amdtp_am824_set_parameters(stream, rate, formation->pcm, - formation->midi, false); + err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports, false); if (err < 0) return err; -- 2.30.2

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[PATCH AUTOSEL 5.12 021/116] ALSA: rme9652: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit f57a741874bb6995089020e97a1dcdf9b165dcbe ] rme9652 wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.751595] snd_rme9652 0000:00:03.0: disabling already-disabled device [ 1.751605] WARNING: CPU: 0 PID: 174 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.759968] Call Trace: [ 1.760145] snd_rme9652_card_free+0x76/0xa0 [snd_rme9652] [ 1.760434] release_card_device+0x4b/0x80 [snd] [ 1.760679] device_release+0x3b/0xa0 [ 1.760874] kobject_put+0x94/0x1b0 [ 1.761059] put_device+0x13/0x20 [ 1.761235] snd_card_free+0x61/0x90 [snd] [ 1.761454] snd_rme9652_probe+0x3be/0x700 [snd_rme9652] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-4-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/rme9652.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 4df992e846f2..7a4d395abcee 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -1731,7 +1731,8 @@ static int snd_rme9652_free(struct snd_rme9652 *rme9652) if (rme9652->port) pci_release_regions(rme9652->pci); - pci_disable_device(rme9652->pci); + if (pci_is_enabled(rme9652->pci)) + pci_disable_device(rme9652->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 020/116] ALSA: hdspm: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 790f5719b85e12e10c41753b864e74249585ed08 ] hdspm wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.786391] snd_hdspm 0000:00:03.0: disabling already-disabled device [ 1.786400] WARNING: CPU: 0 PID: 182 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.795181] Call Trace: [ 1.795320] snd_hdspm_card_free+0x58/0xa0 [snd_hdspm] [ 1.795595] release_card_device+0x4b/0x80 [snd] [ 1.795860] device_release+0x3b/0xa0 [ 1.796072] kobject_put+0x94/0x1b0 [ 1.796260] put_device+0x13/0x20 [ 1.796438] snd_card_free+0x61/0x90 [snd] [ 1.796659] snd_hdspm_probe+0x97b/0x1440 [snd_hdspm] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-3-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdspm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8d900c132f0f..97a0bff96b28 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6883,7 +6883,8 @@ static int snd_hdspm_free(struct hdspm * hdspm) if (hdspm->port) pci_release_regions(hdspm->pci); - pci_disable_device(hdspm->pci); + if (pci_is_enabled(hdspm->pci)) + pci_disable_device(hdspm->pci); return 0; } -- 2.30.2

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[PATCH AUTOSEL 5.12 019/116] ALSA: hdsp: don't disable if not enabled
by Sasha Levin 05 May '21

05 May '21

From: Tong Zhang <ztong0001(a)gmail.com> [ Upstream commit 507cdb9adba006a7798c358456426e1aea3d9c4f ] hdsp wants to disable a not enabled pci device, which makes kernel throw a warning. Make sure the device is enabled before calling disable. [ 1.758292] snd_hdsp 0000:00:03.0: disabling already-disabled device [ 1.758327] WARNING: CPU: 0 PID: 180 at drivers/pci/pci.c:2146 pci_disable_device+0x91/0xb0 [ 1.766985] Call Trace: [ 1.767121] snd_hdsp_card_free+0x94/0xf0 [snd_hdsp] [ 1.767388] release_card_device+0x4b/0x80 [snd] [ 1.767639] device_release+0x3b/0xa0 [ 1.767838] kobject_put+0x94/0x1b0 [ 1.768027] put_device+0x13/0x20 [ 1.768207] snd_card_free+0x61/0x90 [snd] [ 1.768430] snd_hdsp_probe+0x524/0x5e0 [snd_hdsp] Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Tong Zhang <ztong0001(a)gmail.com> Link: https://lore.kernel.org/r/20210321153840.378226-2-ztong0001@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/rme9652/hdsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4cf879c42dc4..720297cbdf87 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5390,7 +5390,8 @@ static int snd_hdsp_free(struct hdsp *hdsp) if (hdsp->port) pci_release_regions(hdsp->pci); - pci_disable_device(hdsp->pci); + if (pci_is_enabled(hdsp->pci)) + pci_disable_device(hdsp->pci); return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 015/116] ASoC: rsnd: core: Check convert rate in rsnd_hw_params
by Sasha Levin 05 May '21

05 May '21

From: Mikhail Durnev <mikhail_durnev(a)mentor.com> [ Upstream commit 19c6a63ced5e07e40f3a5255cb1f0fe0d3be7b14 ] snd_pcm_hw_params_set_rate_near can return incorrect sample rate in some cases, e.g. when the backend output rate is set to some value higher than 48000 Hz and the input rate is 8000 Hz. So passing the value returned by snd_pcm_hw_params_set_rate_near to snd_pcm_hw_params will result in "FSO/FSI ratio error" and playing no audio at all while the userland is not properly notified about the issue. If SRC is unable to convert the requested sample rate to the sample rate the backend is using, then the requested sample rate should be adjusted in rsnd_hw_params. The userland will be notified about that change in the returned hw_params structure. Signed-off-by: Mikhail Durnev <mikhail_durnev(a)mentor.com> Link: https://lore.kernel.org/r/1615870055-13954-1-git-send-email-mikhail_durnev@… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/core.c | 69 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1029d8d9d800..d2b4632d9c2a 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1428,8 +1428,75 @@ static int rsnd_hw_params(struct snd_soc_component *component, } if (io->converted_chan) dev_dbg(dev, "convert channels = %d\n", io->converted_chan); - if (io->converted_rate) + if (io->converted_rate) { + /* + * SRC supports convert rates from params_rate(hw_params)/k_down + * to params_rate(hw_params)*k_up, where k_up is always 6, and + * k_down depends on number of channels and SRC unit. + * So all SRC units can upsample audio up to 6 times regardless + * its number of channels. And all SRC units can downsample + * 2 channel audio up to 6 times too. + */ + int k_up = 6; + int k_down = 6; + int channel; + struct rsnd_mod *src_mod = rsnd_io_to_mod_src(io); + dev_dbg(dev, "convert rate = %d\n", io->converted_rate); + + channel = io->converted_chan ? io->converted_chan : + params_channels(hw_params); + + switch (rsnd_mod_id(src_mod)) { + /* + * SRC0 can downsample 4, 6 and 8 channel audio up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 4 channel audio + * up to 4 times. + * SRC1, SRC3 and SRC4 can downsample 6 and 8 channel audio + * no more than twice. + */ + case 1: + case 3: + case 4: + if (channel > 4) { + k_down = 2; + break; + } + fallthrough; + case 0: + if (channel > 2) + k_down = 4; + break; + + /* Other SRC units do not support more than 2 channels */ + default: + if (channel > 2) + return -EINVAL; + } + + if (params_rate(hw_params) > io->converted_rate * k_down) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + io->converted_rate * k_down; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + io->converted_rate * k_down; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } else if (params_rate(hw_params) * k_up < io->converted_rate) { + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->min = + (io->converted_rate + k_up - 1) / k_up; + hw_param_interval(hw_params, SNDRV_PCM_HW_PARAM_RATE)->max = + (io->converted_rate + k_up - 1) / k_up; + hw_params->cmask |= SNDRV_PCM_HW_PARAM_RATE; + } + + /* + * TBD: Max SRC input and output rates also depend on number + * of channels and SRC unit: + * SRC1, SRC3 and SRC4 do not support more than 128kHz + * for 6 channel and 96kHz for 8 channel audio. + * Perhaps this function should return EINVAL if the input or + * the output rate exceeds the limitation. + */ + } } return rsnd_dai_call(hw_params, io, substream, hw_params); -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 012/116] ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF
by Sasha Levin 05 May '21

05 May '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit b7c7203a1f751348f35fc4bcb157572d303f7573 ] The Asus T100TAF uses the same jack-detect settings as the T100TA, this has been confirmed on actual hardware. Add these settings to the T100TAF quirks to enable jack-detect support on the T100TAF. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210312114850.13832-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5d48cc359c3d..19faababb78d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -482,6 +482,9 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "T100TAF"), }, .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MONO_SPEAKER | BYT_RT5640_DIFF_MIC | BYT_RT5640_SSP0_AIF2 | -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 002/116] ALSA: usb-audio: Add Pioneer DJM-850 to quirks-table
by Sasha Levin 05 May '21

05 May '21

From: Nicolas MURE <nicolas.mure2019(a)gmail.com> [ Upstream commit a3c30b0cb6d05f5bf66d1a5d42e876f31753a447 ] Declare the Pioneer DJM-850 interfaces for capture and playback. See https://github.com/nm2107/Pioneer-DJM-850-driver-reverse-engineering/blob/1… for the complete device spec. Signed-off-by: Nicolas MURE <nicolas.mure2019(a)gmail.com> Link: https://lore.kernel.org/r/20210301152729.18094-2-nicolas.mure2019@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 63 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1165a5ac60f2..9716a9f7c095 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3817,6 +3817,69 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-850 + * 8 channels playback and 8 channels capture @ 44.1/48/96kHz S24LE + * Playback on EP 0x05 + * Capture on EP 0x86 + */ + USB_DEVICE_VENDOR_SPEC(0x08e4, 0x0163), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_DATA, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_DATA, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { 44100, 48000, 96000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-450 -- 2.30.2

1 0

Setting volume of headphones seems to apply to Speakers as well
by GitHub issues - edited 05 May '21

05 May '21

alsa-project/alsa-utils issue #87 was edited from roboknight: When using amixer in the following way: amixer -c 1 set Headphone 0% mute I receive the following partial output from scontents: Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback 0 [0%] [-65.25dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] which looks good. However, it appears that the volume is applied across both the Headphone control and Speaker control (or maybe even just the Master volume) as there is no output from the speakers after this even though it would appear they should have full volume and are not muted. I am using the Ubuntu Groovy Gorilla update, so Linux kernel 5.11. The default installation uses pulseaudio, but I've switched things to pipewire and both cause the same issue presumably because both use ALSA. I started here because amixer seems to at least allow me to identify the problem. When I change the Volume for the Headphones to 100% using the following: amixer -c 1 set Headphone 100% mute the audio from the speakers is then fine. amixer appears to be showing the correct information (as per above) according to the audio, but the Headphone volume seems to be applying to the Speakers as well. Maybe someone can point me in a direction that I can find a solution to this, as it would appear that anytime headphones are plugged in, removing the headphones causes the same situation: no sound. I suspect amixer is working fine, but wherever the volume from amixer is applied, it does not appear to be set appropriately. Issue URL : https://github.com/alsa-project/alsa-utils/issues/87 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Setting volume of headphones seems to apply to Speakers as well
by GitHub issues - opened 05 May '21

05 May '21

alsa-project/alsa-utils issue #87 was opened from roboknight: When using amixer in the following way: amixer -c 1 set Headphone 0% mute I receive the following partial output from scontents: Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 0 [0%] [-65.25dB] [off] Front Right: Playback 0 [0%] [-65.25dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 87 Mono: Front Left: Playback 87 [100%] [0.00dB] [on] Front Right: Playback 87 [100%] [0.00dB] [on] which looks good. However, it appears that the volume is applied across both the Headphone control and Speaker control (or maybe even just the Master volume) as there is no output from the speakers after this even though it would appear they should have full volume and are not muted. I am using the Ubuntu Groovy Gorilla update, so Linux kernel 5.11. The default installation uses pulseaudio, but I've switched things to pipewire and both cause the same issue presumable because both use ALSA. I started here because amixer seems to at least allow me to identify the problem. When I change the Volume for the Headphones to 100% using the following: amixer -c 1 set Headphone 100% mute the audio from the speakers is then fine. amixer appears to be showing the correct information (as per above) according to the audio, but the Headphone volume seems to be applying to the Speakers as well. Maybe someone can point me in a direction that I can find a solution to this, as it would appear that anytime headphones are plugged in, removing the headphones causes the same situation: no sound. I suspect amixer is working fine, but wherever the volume from amixer is applied, it does not appear to be set appropriately. Issue URL : https://github.com/alsa-project/alsa-utils/issues/87 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[a52] Strange behavior with 5.1 surround on ALC1220
by GitHub issues - edited 05 May '21

05 May '21

alsa-project/alsa-plugins issue #21 was edited from davidebeatrici: I originally encountered the issue ~1 year ago, when I started using a motherboard with an ALC1220 on it. The out-of-box experience was flawless: I simply connected the optical SPDIF cable to my amplifier/receiver and stereo sound immediately worked. Once I switched to 5.1 surround, audio output became extremely fast; even videos were affected by it, to the point that they were being reproduced in seconds instead of minutes. After a few days of research and experiments I managed to workaround the issue by replacing the content of `60-a52-encoder.conf` with: ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 card $CARD } rate 48000 } } pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } ``` Based on https://help.ubuntu.com/community/DigitalAC-3Pulseaudio#Configuring_asound.…. After that, I wanted to investigate the cause and report my findings to either ALSA or PulseAudio (depending on the results). Unfortunately that never happened and I kinda forgot about it, as [libasound2-plugins](https://packages.debian.org/bullseye/libasound2-plugins) was updated a single time in the past year and thus I only had to replace the file's content once more. Fast-forward to today: I switch from PulseAudio to PipeWire and while troubleshooting https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1143 I decided to try to restore the original file. Interestingly, the behavior is different compared to PulseAudio: audio, instead of being fast, is crackling. It's basically as if a frame every two is being skipped. Unfortunately PipeWire reports no errors whatsoever. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/21 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[a52] Strange behavior with 5.1 surround on ALC1220
by GitHub issues - edited 05 May '21

05 May '21

alsa-project/alsa-plugins issue #21 was edited from davidebeatrici: I originally encountered the issue ~1 year ago, when I started using a motherboard with an ALC1120 on it. The out-of-box experience was flawless: I simply connected the optical SPDIF cable to my amplifier/receiver and stereo sound immediately worked. Once I switched to 5.1 surround, audio output became extremely fast; even videos were affected by it, to the point that they were being reproduced in seconds instead of minutes. After a few days of research and experiments I managed to workaround the issue by replacing the content of `60-a52-encoder.conf` with: ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 card $CARD } rate 48000 } } pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } ``` Based on https://help.ubuntu.com/community/DigitalAC-3Pulseaudio#Configuring_asound.…. After that, I wanted to investigate the cause and report my findings to either ALSA or PulseAudio (depending on the results). Unfortunately that never happened and I kinda forgot about it, as [libasound2-plugins](https://packages.debian.org/bullseye/libasound2-plugins) was updated a single time in the past year and thus I only had to replace the file's content once more. Fast-forward to today: I switch from PulseAudio to PipeWire and while troubleshooting https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1143 I decided to try to restore the original file. Interestingly, the behavior is different compared to PulseAudio: audio, instead of being fast, is crackling. It's basically as if a frame every two is being skipped. Unfortunately PipeWire reports no errors whatsoever. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/21 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[a52] Strange behavior with 5.1 surround on ALC1120
by GitHub issues - opened 05 May '21

05 May '21

alsa-project/alsa-plugins issue #21 was opened from davidebeatrici: I originally encountered the issue ~1 year ago, when I started using a motherboard with an ALC1120 on it. The out-of-box experience was flawless: I simply connected the optical SPDIF cable to my amplifier/receiver and stereo sound immediately worked. Once I switched to 5.1 surround, audio output became extremely fast; even videos were affected by it, to the point that they were being reproduced in seconds instead of minutes. After a few days of research and experiments I managed to workaround the issue by replacing the content of `60-a52-encoder.conf` with: ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 card $CARD } rate 48000 } } pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } ``` Based on https://help.ubuntu.com/community/DigitalAC-3Pulseaudio#Configuring_asound.…. After that, I wanted to investigate the cause and report my findings to either ALSA or PulseAudio (depending on the results). Unfortunately that never happened and I kinda forgot about it, as [libasound2-plugins](https://packages.debian.org/bullseye/libasound2-plugins) was updated a single time in the past year and thus I only had to replace the file's content once more. Fast-forward to today: I switch from PulseAudio to PipeWire and while troubleshooting https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1143 I decided to try to restore the original file. Interestingly, the behavior is different compared to PulseAudio: audio, instead of being fast, is crackling. It's basically as if a frame every two is being skipped. Unfortunately PipeWire reports no errors whatsoever. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/21 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 0/9] ASoC: Intel: add new TGL/ADL configurations
by Pierre-Louis Bossart 04 May '21

04 May '21

Very little code but quite a few descriptors to add TigerLake (TGL) /AlderLake (ADL) ACPI match tables for I2S and SoundWire devices, new dailinks for Bluetooth offload. Some day this will be read from platform firmware. Also clarify how microphones are handled for SoundWire devices, and create modules to avoid linking the same code multiple times. Pierre-Louis Bossart (5): ASoC: Intel: soc-acpi: add ADL SoundWire base configurations ASoC: Intel: soc-acpi: add ADL jack-less SoundWire configurations ASoC: Intel: sof_sdw: add mutual exclusion between PCH DMIC and RT715 ASoC: Intel: boards: handle hda-dsp-common as a module ASoC: Intel: boards: create sof-maxim-common module Vamshi Krishna Gopal (3): ASoC: Intel: soc-acpi: add entries for i2s machines in ADL match table ASoC: Intel: sof_sdw: add quirk for new ADL-P Rvp ASoC: Intel: boards: add support for adl boards in sof-rt5682 Yong Zhi (1): ASoC: Intel: Boards: tgl_max98373: Add BT offload support sound/soc/intel/boards/Kconfig | 18 ++ sound/soc/intel/boards/Makefile | 28 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 1 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 1 + sound/soc/intel/boards/ehl_rt5660.c | 1 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 1 + sound/soc/intel/boards/hda_dsp_common.c | 5 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_da7219_max98373.c | 1 + sound/soc/intel/boards/sof_maxim_common.c | 24 +- sound/soc/intel/boards/sof_maxim_common.h | 6 +- sound/soc/intel/boards/sof_pcm512x.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 67 ++++- sound/soc/intel/boards/sof_sdw.c | 32 ++- sound/soc/intel/boards/sof_sdw_common.h | 1 + sound/soc/intel/boards/sof_sdw_max98373.c | 4 +- .../intel/common/soc-acpi-intel-adl-match.c | 249 ++++++++++++++++++ 18 files changed, 415 insertions(+), 27 deletions(-) -- 2.25.1

2 13

[PATCH alsa-lib] mixer: simple - Fix "Capture Volume" and "Capture Switch" being seen as global controls
by Hans de Goede 04 May '21

04 May '21

Fix the "Capture Volume" and "Capture Switch" exceptions no longer working after commit 86b9c67774bc ("mixer: simple - Unify simple_none: base_len() exception handling") because they were moved to after the suffix checking, so they would be treated as CTL_GLOBAL_VOLUME resp. CTL_GLOBAL_SWITCH based on their suffix before the exception check has a chance to check for a match. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/mixer/simple_none.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/src/mixer/simple_none.c b/src/mixer/simple_none.c index 71d88486..9b9f0000 100644 --- a/src/mixer/simple_none.c +++ b/src/mixer/simple_none.c @@ -913,6 +913,16 @@ static int base_len(const char *name, selem_ctl_type_t *type) const struct suf *p; size_t nlen = strlen(name); + /* exception: "Capture Volume" and "Capture Switch" */ + if (!strcmp(name, "Capture Volume")) { + *type = CTL_CAPTURE_VOLUME; + return strlen("Capture"); + } + if (!strcmp(name, "Capture Switch")) { + *type = CTL_CAPTURE_SWITCH; + return strlen("Capture"); + } + for (p = suffixes; p->suffix; p++) { size_t slen = strlen(p->suffix); size_t l; @@ -926,16 +936,6 @@ static int base_len(const char *name, selem_ctl_type_t *type) } } - /* exception: "Capture Volume" and "Capture Switch" */ - if (!strcmp(name, "Capture Volume")) { - *type = CTL_CAPTURE_VOLUME; - return strlen("Capture"); - } - if (!strcmp(name, "Capture Switch")) { - *type = CTL_CAPTURE_SWITCH; - return strlen("Capture"); - } - /* Special case - handle "Input Source" as a capture route. * Note that it's *NO* capture source. A capture source is split over * sub-elements, and multiple capture-sources will result in an error. -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: Add fixup for HP OMEN laptop
by Takashi Iwai 04 May '21

04 May '21

HP OMEN dc0019-ur with codec SSID 103c:84da requires the pin config overrides and the existing mic/mute LED setup. This patch implements those in the fixup table. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212733 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ee2841dd0c6d..6d58f24c9702 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6506,6 +6506,7 @@ enum { ALC285_FIXUP_IDEAPAD_S740_COEF, ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST, ALC295_FIXUP_ASUS_DACS, + ALC295_FIXUP_HP_OMEN, }; static const struct hda_fixup alc269_fixups[] = { @@ -8014,6 +8015,26 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc295_fixup_asus_dacs, }, + [ALC295_FIXUP_HP_OMEN] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0xb7a60130 }, + { 0x13, 0x40000000 }, + { 0x14, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x90170110 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x02a11030 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x04a19030 }, + { 0x1d, 0x40600001 }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x03211020 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8172,6 +8193,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), + SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), @@ -8577,6 +8599,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, + {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.26.2

1 0

[PATCH v3 0/1] Enable using multiple different type kcontrols for widgets
by Jaska Uimonen 04 May '21

04 May '21

Hi, This is a patch v3 for enabling multiple different types of kcontrols for a dapm widget. Currently asoc allows to define and parse multiple same type kcontrols for single widget. So you can have for example two volume controls in a widget but not one byte and one enum control. Lately we've seen couple of cases where different types of controls would be useful and alsa topology actually allows you to create these. This patch refactors the dapm kcontrol parsing by lifting the creation loop and memory allocation up one level and making the type variable to hold multiple type values. Sof driver is modified to use this type information in the same patch as it is quite cumbersome to split the changes nicely. changes since v2: - only 1 patch for not breaking bisect - sof internal reviews -> minor changes + reviewed by tags - no upstream comments after v2 -> assuming I'm on the right track here Jaska Uimonen (1): ASoC: dapm: Add support for multiple kcontrol types to a widget include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 466 +++++++++++++++++------------------ sound/soc/sof/topology.c | 15 +- 3 files changed, 233 insertions(+), 250 deletions(-) -- 2.24.1

1 1

[PATCH] ALSA: hda/realtek: Fix speaker amp on HP Envy AiO 32
by Takashi Iwai 04 May '21

04 May '21

HP Envy AiO 32-a12xxx has an external amp that is controlled via GPIO bit 0x04. However, unlike other devices, this amp seems to shut down itself after the certain period, hence the OS needs to up/down the bit dynamically only during the actual playback. This patch adds the control of the GPIO bit via the existing pcm_hook mechanism. Ideally it should be triggered at the actual stream start, but we have only the state change at prepare/cleanup, so use those for switching the GPIO bit on/off. This should be good enough for the purpose, and was actually confirmed to work fine. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212873 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bbbe37f3e724..ee2841dd0c6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4338,6 +4338,35 @@ static void alc245_fixup_hp_x360_amp(struct hda_codec *codec, } } +/* toggle GPIO2 at each time stream is started; we use PREPARE state instead */ +static void alc274_hp_envy_pcm_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) +{ + switch (action) { + case HDA_GEN_PCM_ACT_PREPARE: + alc_update_gpio_data(codec, 0x04, true); + break; + case HDA_GEN_PCM_ACT_CLEANUP: + alc_update_gpio_data(codec, 0x04, false); + break; + } +} + +static void alc274_fixup_hp_envy_gpio(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PROBE) { + spec->gpio_mask |= 0x04; + spec->gpio_dir |= 0x04; + spec->gen.pcm_playback_hook = alc274_hp_envy_pcm_hook; + } +} + static void alc_update_coef_led(struct hda_codec *codec, struct alc_coef_led *led, bool polarity, bool on) @@ -6465,6 +6494,7 @@ enum { ALC255_FIXUP_XIAOMI_HEADSET_MIC, ALC274_FIXUP_HP_MIC, ALC274_FIXUP_HP_HEADSET_MIC, + ALC274_FIXUP_HP_ENVY_GPIO, ALC256_FIXUP_ASUS_HPE, ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, ALC287_FIXUP_HP_GPIO_LED, @@ -7907,6 +7937,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC274_FIXUP_HP_MIC }, + [ALC274_FIXUP_HP_ENVY_GPIO] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc274_fixup_hp_envy_gpio, + }, [ALC256_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -8140,6 +8174,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), -- 2.26.2

1 0

[PATCH alsa-lib] ucm: Fix sysw sequence command not working when requesting to ignore errors
by Hans de Goede 04 May '21

04 May '21

When the user requests to ignore sysfs write errors by prefixing the path with a '-' then we need to skip the '-' when building the actual path otherwise the write will never work. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/ucm/main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 2b07cc95..3df9b62a 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -517,7 +517,10 @@ static int execute_sysw(const char *sysw) if (sysw == NULL || *sysw == '\0') return 0; - ignore_error = sysw[0] == '-'; + if (sysw[0] == '-') { + ignore_error = true; + sysw++; + } if (sysw[0] == ':') return -EINVAL; -- 2.31.1

2 1

[PATCH] ALSA: hda/realtek: Fix silent headphone output on ASUS UX430UA
by Takashi Iwai 04 May '21

04 May '21

It was reported that the headphone output on ASUS UX430UA (SSID 1043:1740) with ALC295 codec is silent while the speaker works. After the investigation, it turned out that the DAC assignment has to be fixed on this machine; unlike others, it expects DAC 0x02 to be assigned to the speaker pin 0x07 while DAC 0x03 to headphone pin 0x21. This patch provides a fixup for the fixed DAC/pin mapping for this device. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212933 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0fa99a31c23..bbbe37f3e724 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5695,6 +5695,18 @@ static void alc_fixup_tpt470_dacs(struct hda_codec *codec, spec->gen.preferred_dacs = preferred_pairs; } +static void alc295_fixup_asus_dacs(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t preferred_pairs[] = { + 0x17, 0x02, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.preferred_dacs = preferred_pairs; +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -6463,6 +6475,7 @@ enum { ALC256_FIXUP_ACER_HEADSET_MIC, ALC285_FIXUP_IDEAPAD_S740_COEF, ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC295_FIXUP_ASUS_DACS, }; static const struct hda_fixup alc269_fixups[] = { @@ -7963,6 +7976,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC295_FIXUP_ASUS_DACS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_asus_dacs, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8161,6 +8178,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), 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, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.26.2

1 0

[PATCH] ALSA: usb-audio: Add dB range mapping for Sennheiser Communications Headset PC 8
by Takashi Iwai 03 May '21

03 May '21

From: Timo Gurr <timo.gurr(a)gmail.com> The decibel volume range contains a negative maximum value resulting in pipewire complaining about the device and effectivly having no sound output. The wrong values also resulted in the headset sounding muted already at a mixer level of about ~25%. PipeWire BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1049 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212897 Signed-off-by: Timo Gurr <timo.gurr(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 646deb6244b1..c5794e83fd80 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -337,6 +337,13 @@ static const struct usbmix_name_map bose_companion5_map[] = { { 0 } /* terminator */ }; +/* Sennheiser Communications Headset [PC 8], the dB value is reported as -6 negative maximum */ +static const struct usbmix_dB_map sennheiser_pc8_dB = {-9500, 0}; +static const struct usbmix_name_map sennheiser_pc8_map[] = { + { 9, NULL, .dB = &sennheiser_pc8_dB }, + { 0 } /* terminator */ +}; + /* * Dell usb dock with ALC4020 codec had a firmware problem where it got * screwed up when zero volume is passed; just skip it as a workaround @@ -593,6 +600,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x17aa, 0x1046), .map = lenovo_p620_rear_map, }, + { + /* Sennheiser Communications Headset [PC 8] */ + .id = USB_ID(0x1395, 0x0025), + .map = sennheiser_pc8_map, + }, { 0 } /* terminator */ }; -- 2.26.2

1 0

[PATCH] ALSA: hda/realtek: ALC285 Thinkpad jack pin quirk is unreachable
by Sami Loone 03 May '21

03 May '21

In 07d9e69cf34a ("ALSA: hda/realtek: fix static noise on ALC285 Lenovo laptops") an existing Lenovo quirk was made more generic by removing a 0x12 pin requirement from the entry. This made the second chance table Thinkpad jack entry unreachable as the pin configurations became identical. Revert the 0x12 pin requirement removal and move Thinkpad jack pin quirk back to the primary pin table as they can co-exist when more specific configurations come first. Add a more targeted pin quirk for Lenovo devices that have 0x12 as 0x40000000. Tested on Yoga 6 (AMD) laptop. Fixes: 07d9e69cf34a ("ALSA: hda/realtek: fix static noise on ALC285 Lenovo laptops") Signed-off-by: Sami Loone <sami(a)loone.fi> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bd7bfd7c9ee7..a0fa99a31c23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8801,6 +8801,16 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x19, 0x03a11020}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + {0x12, 0x90a60130}, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x1d, 0x40600001}, + {0x21, 0x04211020}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), @@ -8971,10 +8981,6 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x19, 0x40000000}, {0x1a, 0x40000000}), - SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, - {0x14, 0x90170110}, - {0x19, 0x04a11040}, - {0x21, 0x04211020}), {} }; -- 2.31.1

2 1

[PATCH] sound: virtio: correct the function name in kernel-doc comment
by Randy Dunlap 03 May '21

03 May '21

Fix kernel-doc warning that the wrong function name is used in a kernel-doc comment: ../sound/virtio/virtio_ctl_msg.c:70: warning: expecting prototype for virtsnd_ctl_msg_request(). Prototype was for virtsnd_ctl_msg_response() instead Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Cc: "Michael S. Tsirkin" <mst(a)redhat.com> Cc: virtualization(a)lists.linux-foundation.org Cc: alsa-devel(a)alsa-project.org --- sound/virtio/virtio_ctl_msg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20210414.orig/sound/virtio/virtio_ctl_msg.c +++ linux-next-20210414/sound/virtio/virtio_ctl_msg.c @@ -61,7 +61,7 @@ void *virtsnd_ctl_msg_request(struct vir } /** - * virtsnd_ctl_msg_request() - Get a pointer to the response header. + * virtsnd_ctl_msg_response() - Get a pointer to the response header. * @msg: Control message. * * Context: Any context.

3 2

Surround sound
by GitHub issues - opened 02 May '21

02 May '21

alsa-project/alsa-ucm-conf issue #91 was opened from mchemweno: Is there a way to enable surround sound?? My laptop is a HP Envy13 so it has 4 speakers. Tried using the pulse/daemon.conf to no avail. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/91 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: codecs: lpass-wsa-macro: make sure array index is set
by trix＠redhat.com 30 Apr '21

30 Apr '21

From: Tom Rix <trix(a)redhat.com> Static analysis reports this problem lpass-wsa-macro.c:1732:6: warning: Array subscript is undefined if (wsa->ec_hq[ec_tx]) { ^~~~~~~~~~~~~~~~~ The happens because 'ec_tx' is never initialized and there is no default in the switch statement that sets ec_tx. Because there are only two cases for the switch, convert it to an if-else. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/soc/codecs/lpass-wsa-macro.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 1a7fa5492f28..c11ae72f2148 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1718,15 +1718,13 @@ static int wsa_macro_enable_echo(struct snd_soc_dapm_widget *w, val = snd_soc_component_read(component, CDC_WSA_RX_INP_MUX_RX_MIX_CFG0); - switch (w->shift) { - case WSA_MACRO_EC0_MUX: + if (w->shift == WSA_MACRO_EC0_MUX) { val = val & CDC_WSA_RX_MIX_TX0_SEL_MASK; ec_tx = val - 1; - break; - case WSA_MACRO_EC1_MUX: + } else { + /* WSA_MACRO_EC1_MUX */ val = val & CDC_WSA_RX_MIX_TX1_SEL_MASK; ec_tx = (val >> CDC_WSA_RX_MIX_TX1_SEL_SHFT) - 1; - break; } if (wsa->ec_hq[ec_tx]) { -- 2.26.3

2 1

[PATCH v4 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 30 Apr '21

30 Apr '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Am planning to send support for MBHC once this driver gets accepted! Thanks, srini Many thanks for reviewing v3. Changes since v3: - Moved to using driver component framework - Added device links to allow correct runtime PM - added runtime pm support - split codec bindings into two parts one for codec and other for soundwire device. - Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 61 + .../bindings/sound/qcom,wcd938x.yaml | 176 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 230 + sound/soc/codecs/wcd938x.c | 3786 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 670 +++ 9 files changed, 5294 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

3 17

[v8 1/4] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 29 Apr '21

29 Apr '21

From: Chris Morgan <macromorgan(a)hotmail.com> Add rk817 codec support cell to rk808 mfd driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. drivers/mfd/rk808.c | 85 +++++++++++++++++++++++++++++++++++++++ include/linux/mfd/rk808.h | 81 +++++++++++++++++++++++++++++++++++++ 2 files changed, 166 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..9231209184e0 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -65,6 +65,7 @@ static bool rk817_is_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case RK817_SECONDS_REG ... RK817_WEEKS_REG: case RK817_RTC_STATUS_REG: + case RK817_CODEC_DTOP_LPT_SRST: case RK817_INT_STS_REG0: case RK817_INT_STS_REG1: case RK817_INT_STS_REG2: @@ -163,6 +164,11 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { @@ -201,6 +207,85 @@ static const struct rk808_reg_data rk808_pre_init_reg[] = { static const struct rk808_reg_data rk817_pre_init_reg[] = { {RK817_RTC_CTRL_REG, RTC_STOP, RTC_STOP}, + /* Codec specific registers */ + { RK817_CODEC_DTOP_VUCTL, MASK_ALL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, MASK_ALL, 0x06 }, + { RK817_CODEC_AADC_CFG0, MASK_ALL, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC2, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_NG, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_HPF, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AHP_ANTI0, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_ANTI1, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_CFG0, MASK_ALL, 0xe0 }, + { RK817_CODEC_AHP_CFG1, MASK_ALL, 0x1f }, + { RK817_CODEC_AHP_CP, MASK_ALL, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, MASK_ALL, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, MASK_ALL, 0x44 }, + { RK817_CODEC_APLL_CFG0, MASK_ALL, 0x04 }, + { RK817_CODEC_APLL_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_APLL_CFG2, MASK_ALL, 0x30 }, + { RK817_CODEC_APLL_CFG3, MASK_ALL, 0x19 }, + { RK817_CODEC_APLL_CFG4, MASK_ALL, 0x65 }, + { RK817_CODEC_APLL_CFG5, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_CKM, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_RSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, MASK_ALL, 0x00 }, {RK817_GPIO_INT_CFG, RK817_INT_POL_MSK, RK817_INT_POL_L}, {RK817_SYS_CFG(1), RK817_HOTDIE_TEMP_MSK | RK817_TSD_TEMP_MSK, RK817_HOTDIE_105 | RK817_TSD_140}, diff --git a/include/linux/mfd/rk808.h b/include/linux/mfd/rk808.h index e07f6e61cd38..a96e6d43ca06 100644 --- a/include/linux/mfd/rk808.h +++ b/include/linux/mfd/rk808.h @@ -437,6 +437,87 @@ enum rk809_reg_id { #define RK817_RTC_COMP_LSB_REG 0x10 #define RK817_RTC_COMP_MSB_REG 0x11 +/* RK817 Codec Registers */ +#define RK817_CODEC_DTOP_VUCTL 0x12 +#define RK817_CODEC_DTOP_VUCTIME 0x13 +#define RK817_CODEC_DTOP_LPT_SRST 0x14 +#define RK817_CODEC_DTOP_DIGEN_CLKE 0x15 +#define RK817_CODEC_AREF_RTCFG0 0x16 +#define RK817_CODEC_AREF_RTCFG1 0x17 +#define RK817_CODEC_AADC_CFG0 0x18 +#define RK817_CODEC_AADC_CFG1 0x19 +#define RK817_CODEC_DADC_VOLL 0x1a +#define RK817_CODEC_DADC_VOLR 0x1b +#define RK817_CODEC_DADC_SR_ACL0 0x1e +#define RK817_CODEC_DADC_ALC1 0x1f +#define RK817_CODEC_DADC_ALC2 0x20 +#define RK817_CODEC_DADC_NG 0x21 +#define RK817_CODEC_DADC_HPF 0x22 +#define RK817_CODEC_DADC_RVOLL 0x23 +#define RK817_CODEC_DADC_RVOLR 0x24 +#define RK817_CODEC_AMIC_CFG0 0x27 +#define RK817_CODEC_AMIC_CFG1 0x28 +#define RK817_CODEC_DMIC_PGA_GAIN 0x29 +#define RK817_CODEC_DMIC_LMT1 0x2a +#define RK817_CODEC_DMIC_LMT2 0x2b +#define RK817_CODEC_DMIC_NG1 0x2c +#define RK817_CODEC_DMIC_NG2 0x2d +#define RK817_CODEC_ADAC_CFG0 0x2e +#define RK817_CODEC_ADAC_CFG1 0x2f +#define RK817_CODEC_DDAC_POPD_DACST 0x30 +#define RK817_CODEC_DDAC_VOLL 0x31 +#define RK817_CODEC_DDAC_VOLR 0x32 +#define RK817_CODEC_DDAC_SR_LMT0 0x35 +#define RK817_CODEC_DDAC_LMT1 0x36 +#define RK817_CODEC_DDAC_LMT2 0x37 +#define RK817_CODEC_DDAC_MUTE_MIXCTL 0x38 +#define RK817_CODEC_DDAC_RVOLL 0x39 +#define RK817_CODEC_DDAC_RVOLR 0x3a +#define RK817_CODEC_AHP_ANTI0 0x3b +#define RK817_CODEC_AHP_ANTI1 0x3c +#define RK817_CODEC_AHP_CFG0 0x3d +#define RK817_CODEC_AHP_CFG1 0x3e +#define RK817_CODEC_AHP_CP 0x3f +#define RK817_CODEC_ACLASSD_CFG1 0x40 +#define RK817_CODEC_ACLASSD_CFG2 0x41 +#define RK817_CODEC_APLL_CFG0 0x42 +#define RK817_CODEC_APLL_CFG1 0x43 +#define RK817_CODEC_APLL_CFG2 0x44 +#define RK817_CODEC_APLL_CFG3 0x45 +#define RK817_CODEC_APLL_CFG4 0x46 +#define RK817_CODEC_APLL_CFG5 0x47 +#define RK817_CODEC_DI2S_CKM 0x48 +#define RK817_CODEC_DI2S_RSD 0x49 +#define RK817_CODEC_DI2S_RXCR1 0x4a +#define RK817_CODEC_DI2S_RXCR2 0x4b +#define RK817_CODEC_DI2S_RXCMD_TSD 0x4c +#define RK817_CODEC_DI2S_TXCR1 0x4d +#define RK817_CODEC_DI2S_TXCR2 0x4e +#define RK817_CODEC_DI2S_TXCR3_TXCMD 0x4f + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + #define RK817_POWER_EN_REG(i) (0xb1 + (i)) #define RK817_POWER_SLP_EN_REG(i) (0xb5 + (i)) -- 2.25.1

3 5

[PATCH 1/2] Revert "ALSA: usb-audio: Add support for many Roland devices..."
by Takashi Iwai 29 Apr '21

29 Apr '21

This reverts commit d86f43b17ed4 ("ALSA: usb-audio: Add support for many Roland devices' feedback quirks"). It turned out that many quirk entries there don't contain the proper EP values and/or the quirk types, which lead to the broken operations. As we're going to cover all Roland/BOSS devices in a more generic way rather the explicit lists, let's revert the previous additions at first. Fixes: d86f43b17ed4 ("ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 59 -------------------------------------------- 1 file changed, 59 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 77ffcea294dd..94acfaa7f2ef 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -79,72 +79,13 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a6, 0x0d, 0x01), /* Roland JUNO-G */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a9, 0x0d, 0x01), /* Roland MC-808 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ad, 0x0d, 0x01), /* Roland SH-201 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b2, 0x0d, 0x01), /* Roland VG-99 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b3, 0x0d, 0x01), /* Roland VG-99 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c2, 0x0d, 0x01), /* Roland SonicCell */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c4, 0x0d, 0x01), /* Edirol M-16DX */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c5, 0x0d, 0x01), /* Roland SP-555 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c7, 0x0d, 0x01), /* Roland V-Synth GT */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00d1, 0x0d, 0x01), /* Roland Music Atelier */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00da, 0x0d, 0x01), /* BOSS GT-10 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00db, 0x0d, 0x01), /* BOSS GT-10 Guitar Effects Processor */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00dc, 0x0d, 0x01), /* BOSS GT-10B */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00de, 0x0d, 0x01), /* Roland Fantom-G */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00eb, 0x0d, 0x01), /* Roland VS-100 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00f8, 0x0d, 0x01), /* Roland JUNO Series */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fc, 0x0d, 0x01), /* Roland VS-700C */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fd, 0x0d, 0x01), /* Roland VS-700 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fe, 0x0d, 0x01), /* Roland VS-700 M1 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ff, 0x0d, 0x01), /* Roland VS-700 M2 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0100, 0x0d, 0x01), /* Roland VS-700 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0101, 0x0d, 0x01), /* Roland VS-700 M2 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0102, 0x0d, 0x01), /* Roland VB-99 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x0109, 0x0d, 0x01), /* BOSS eBand JS-8 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x0111, 0x0d, 0x01), /* Roland GAIA SH-01 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0113, 0x0d, 0x01), /* BOSS ME-25 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0114, 0x0d, 0x01), /* Roland SD-50 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0117, 0x0d, 0x01), /* Roland VS-20 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0119, 0x0d, 0x01), /* Roland OCTAPAD SPD-30 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x011c, 0x0d, 0x01), /* Roland Lucina AX-09 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x011e, 0x0d, 0x01), /* BOSS BR-800 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0120, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0121, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0123, 0x0d, 0x01), /* Roland JUNO-Gi */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0124, 0x0d, 0x01), /* Roland M-300 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x0127, 0x0d, 0x01), /* Roland GR-55 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x012b, 0x0d, 0x01), /* Roland DUO-CAPTURE */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x012f, 0x0d, 0x01), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0132, 0x0d, 0x01), /* Roland TRI-CAPTURE */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0134, 0x0d, 0x01), /* Roland V-Mixer */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0137, 0x0d, 0x01), /* Roland DUO-CAPTURE Advanced Mode */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x0138, 0x0d, 0x01), /* BOSS RC-300 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x013a, 0x0d, 0x01), /* Roland JUPITER-80 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x013e, 0x0d, 0x01), /* Roland R-26 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0145, 0x0d, 0x01), /* Roland SPD-SX */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x014b, 0x0d, 0x01), /* BOSS eBand JS-10 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x014d, 0x0d, 0x01), /* BOSS GT-100 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0150, 0x0d, 0x01), /* Roland TD-15 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0151, 0x0d, 0x01), /* Roland TD-11 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0154, 0x0d, 0x01), /* Roland JUPITER-50 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0158, 0x0d, 0x01), /* Roland TD-30 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0159, 0x0d, 0x01), /* Roland DUO-CAPTURE EX */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x015b, 0x0d, 0x01), /* Roland INTEGRA-7 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x015d, 0x0d, 0x01), /* Roland R-88 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x017a, 0x0d, 0x01), /* Roland VT-3 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x017c, 0x0d, 0x01), /* Roland TR-8 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x01b5, 0x0d, 0x01), /* Roland Boutique JP-08 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x01fd, 0x0d, 0x01), /* Roland Boutique SH-01A */ - IMPLICIT_FB_BOTH_DEV(0x0582, 0x01ff, 0x0d, 0x01), /* Roland Boutique D-05 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */ -- 2.26.2

4 14

[PATCH] drivers: slimbus: qcom-ngd-ctrl.c: Added space after comma
by Shubhankar Kuranagatti 29 Apr '21

29 Apr '21

A space has been give after the ','. This is done to maintain code uniformity. Signed-off-by: Shubhankar Kuranagatti <shubhankarvk(a)gmail.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..cb9612a5c30c 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1348,7 +1348,7 @@ static const struct of_device_id qcom_slim_ngd_dt_match[] = { { .compatible = "qcom,slim-ngd-v1.5.0", .data = &ngd_v1_5_offset_info, - },{ + }, { .compatible = "qcom,slim-ngd-v2.1.0", .data = &ngd_v1_5_offset_info, }, -- 2.17.1

1 0

[PATCH] ASoC: max98088: fix ni clock divider calculation
by Marco Felsch 28 Apr '21

28 Apr '21

The ni1/ni2 ratio formula [1] uses the pclk which is the prescaled mclk. The max98088 datasheet [2] has no such formula but table-12 equals so we can assume that it is the same for both devices. While on it make use of DIV_ROUND_CLOSEST_ULL(). [1] https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf; page 86 [2] https://datasheets.maximintegrated.com/en/ds/MAX98088.pdf; page 82 Signed-off-by: Marco Felsch <m.felsch(a)pengutronix.de> --- Hi, please be noted that I used the old formatting style which uses spaces instead of tabs. Therefore checkpatch.pl complains. The whole driver needs a style adaption but I wanted to keep it for now. Regards, Marco sound/soc/codecs/max98088.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 4be24e7f51c8..f8e49e45ce33 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -41,6 +41,7 @@ struct max98088_priv { enum max98088_type devtype; struct max98088_pdata *pdata; struct clk *mclk; + unsigned char mclk_prescaler; unsigned int sysclk; struct max98088_cdata dai[2]; int eq_textcnt; @@ -998,13 +999,16 @@ static int max98088_dai1_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_14_DAI1_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, @@ -1065,13 +1069,16 @@ static int max98088_dai2_hw_params(struct snd_pcm_substream *substream, /* Configure NI when operating as master */ if (snd_soc_component_read(component, M98088_REG_1C_DAI2_FORMAT) & M98088_DAI_MAS) { + unsigned long pclk; + if (max98088->sysclk == 0) { dev_err(component->dev, "Invalid system clock frequency\n"); return -EINVAL; } ni = 65536ULL * (rate < 50000 ? 96ULL : 48ULL) * (unsigned long long int)rate; - do_div(ni, (unsigned long long int)max98088->sysclk); + pclk = DIV_ROUND_CLOSEST(max98088->sysclk, max98088->mclk_prescaler); + ni = DIV_ROUND_CLOSEST_ULL(ni, pclk); snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, (ni >> 8) & 0x7F); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, @@ -1113,8 +1120,10 @@ static int max98088_dai_set_sysclk(struct snd_soc_dai *dai, */ if ((freq >= 10000000) && (freq < 20000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x10); + max98088->mclk_prescaler = 1; } else if ((freq >= 20000000) && (freq < 30000000)) { snd_soc_component_write(component, M98088_REG_10_SYS_CLK, 0x20); + max98088->mclk_prescaler = 2; } else { dev_err(component->dev, "Invalid master clock frequency\n"); return -EINVAL; -- 2.29.2

2 1

[PATCH 0/2] ASoC: Revert clk_hw_get_clk() cleanup
by Jerome Brunet 28 Apr '21

28 Apr '21

There is problem with clk_hw_get_hw(). Using it pins the clock provider to itself, making it impossible to remove the related module. Revert the two commits using this function until this gets sorted out. Jerome Brunet (2): ASoC: stm32: do not request a new clock consummer reference ASoC: da7219: do not request a new clock consummer reference sound/soc/codecs/da7219.c | 5 +---- sound/soc/stm/stm32_sai_sub.c | 5 +---- 2 files changed, 2 insertions(+), 8 deletions(-) -- 2.31.1

3 4

[PATCH] ASoC: amd: fix for pcm_read() error
by Vijendar Mukunda 28 Apr '21

28 Apr '21

Below phython script throwing pcm_read() error. import subprocess p = subprocess.Popen(["aplay -t raw -D plughw:1,0 /dev/zero"], shell=True) subprocess.call(["arecord -Dhw:1,0 --dump-hw-params"], shell=True) subprocess.call(["arecord -Dhw:1,0 -fdat -d1 /dev/null"], shell=True) p.kill() Handling ACP global external interrupt enable register causing this issue. This register got updated wrongly when there is active stream causing interrupts disabled for active stream. Refactored code to handle enabling and disabling external interrupts. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 ---------- sound/soc/amd/raven/acp3x.h | 1 + sound/soc/amd/raven/pci-acp3x.c | 15 +++++++++++++++ 3 files changed, 16 insertions(+), 10 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index f22bb2b..8148b0d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -235,10 +235,6 @@ static int acp3x_dma_open(struct snd_soc_component *component, return ret; } - if (!adata->play_stream && !adata->capture_stream && - !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) - rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - i2s_data->acp3x_base = adata->acp3x_base; runtime->private_data = i2s_data; return ret; @@ -365,12 +361,6 @@ static int acp3x_dma_close(struct snd_soc_component *component, } } - /* Disable ACP irq, when the current stream is being closed and - * another stream is also not active. - */ - if (!adata->play_stream && !adata->capture_stream && - !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) - rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 03fe9391..c3f0c8b 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -77,6 +77,7 @@ #define ACP_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_ITER_IRER_SAMP_LEN_MASK 0x38 +#define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF struct acp3x_platform_info { u16 play_i2s_instance; diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index d3536fd..a013a60 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -76,6 +76,19 @@ static int acp3x_reset(void __iomem *acp3x_base) return -ETIMEDOUT; } +static void acp3x_enable_interrupts(void __iomem *acp_base) +{ + rv_writel(0x01, acp_base + mmACP_EXTERNAL_INTR_ENB); +} + +static void acp3x_disable_interrupts(void __iomem *acp_base) +{ + rv_writel(ACP_EXT_INTR_STAT_CLEAR_MASK, acp_base + + mmACP_EXTERNAL_INTR_STAT); + rv_writel(0x00, acp_base + mmACP_EXTERNAL_INTR_CNTL); + rv_writel(0x00, acp_base + mmACP_EXTERNAL_INTR_ENB); +} + static int acp3x_init(struct acp3x_dev_data *adata) { void __iomem *acp3x_base = adata->acp3x_base; @@ -93,6 +106,7 @@ static int acp3x_init(struct acp3x_dev_data *adata) pr_err("ACP3x reset failed\n"); return ret; } + acp3x_enable_interrupts(acp3x_base); return 0; } @@ -100,6 +114,7 @@ static int acp3x_deinit(void __iomem *acp3x_base) { int ret; + acp3x_disable_interrupts(acp3x_base); /* Reset */ ret = acp3x_reset(acp3x_base); if (ret) { -- 2.7.4

2 1

[PATCH 00/13] ALSA: hda: Re-order quirk table entries
by Takashi Iwai 28 Apr '21

28 Apr '21

Hi, this is a series of trivial changes just to re-order the quirk table entries for HD-audio codec drivers. We've tried to keep the entries in the PCI SSID order, but occasionally somd disorder came in. It's not only bad for the readability but also leading to the oversight of duplicated entries or the unreachable entries unexpectedly. Basically this could be done in a shot by a script, but I split to parts, so that each piece can be better applied (or fixed) for stable kernels. There is one fix patch in the series that addresses the duplicated entries. Other than that, all about shuffling and no functional changes must be introduced by this series. Takashi === Takashi Iwai (13): ALSA: hda/realtek: Re-order ALC882 Acer quirk table entries ALSA: hda/realtek: Re-order ALC882 Sony quirk table entries ALSA: hda/realtek: Re-order ALC882 Clevo quirk table entries ALSA: hda/realtek: Re-order ALC269 HP quirk table entries ALSA: hda/realtek: Re-order ALC269 Acer quirk table entries ALSA: hda/realtek: Re-order ALC269 Dell quirk table entries ALSA: hda/realtek: Re-order ALC269 ASUS quirk table entries ALSA: hda/realtek: Re-order ALC269 Sony quirk table entries ALSA: hda/realtek: Re-order ALC269 Lenovo quirk table entries ALSA: hda/realtek: Re-order remaining ALC269 quirk table entries ALSA: hda/realtek: Re-order ALC662 quirk table entries ALSA: hda/realtek: Remove redundant entry for ALC861 Haier/Uniwill devices ALSA: hda/conexant: Re-order CX5066 quirk table entries sound/pci/hda/patch_conexant.c | 14 ++-- sound/pci/hda/patch_realtek.c | 113 ++++++++++++++++----------------- 2 files changed, 62 insertions(+), 65 deletions(-) -- 2.26.2

1 13

[PATCH] ALSA: usb-audio: Remove redundant assignment to len
by Jiapeng Chong 28 Apr '21

28 Apr '21

Variable len is set to zero but this value is never read as it is overwritten with a new value later on, hence it is a redundant assignment and can be removed. Cleans up the following clang-analyzer warning: sound/usb/mixer.c:2713:3: warning: Value stored to 'len' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/usb/mixer.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index b004b2e..937bc17 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2710,7 +2710,6 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, #define MAX_ITEM_NAME_LEN 64 for (i = 0; i < desc->bNrInPins; i++) { struct usb_audio_term iterm; - len = 0; namelist[i] = kmalloc(MAX_ITEM_NAME_LEN, GFP_KERNEL); if (!namelist[i]) { err = -ENOMEM; -- 1.8.3.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Intel Clevo PCx0Dx
by Werner Sembach 28 Apr '21

28 Apr '21

From: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> This applies a SND_PCI_QUIRK(...) to the Clevo PCx0Dx barebones. This fix enables audio output over the headset jack and ensures that a microphone connected via the headset combo jack is correctly recognized when pluged in. Signed-off-by: Eckhart Mohr <e.mohr(a)tuxedocomputers.com> Co-developed-by: Werner Sembach <wse(a)tuxedocomputers.com> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- >From 7dee2aee3ab045c4dafa9793a4016929a559d9cd Mon Sep 17 00:00:00 2001 From: Werner Sembach <wse(a)tuxedocomputers.com> Date: Tue, 13 Apr 2021 14:02:08 +0200 Subject: [PATCH] Add SND_PCI_QUIRK for Clevo PCx0Dx --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..dba9fd27adec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2555,6 +2555,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.25.1

2 1

ucm2: soundwire: use the codec kcontrols for rt711-sdca
by GitHub pull_request - opened 28 Apr '21

28 Apr '21

alsa-project/alsa-ucm-conf pull request #90 was opened from libinyang: Use "rt711 FU0F" as the HW control, instead of the "PGA2.0 2". Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/90 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/90.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Intro to Topology2.0
by GitHub pull_request - opened 28 Apr '21

28 Apr '21

alsa-project/alsa-utils pull request #86 was opened from ranj063: Add support for pre-processing Topology2.0 conf file and building the tplg binary. An example implementation of topology with Topology2.0 can be found here: https://github.com/thesofproject/sof/pull/3983 Introduction to Topology 2.0 ----- Topology2.0 is a high level keyword extension on top of the existing ALSA conf topology format designed to: 1) Simplify the ALSA conf topology definitions by providing high level "classes" so topology designers need to write less config for common object definitions. 2) Allow simple reuse of objects. Define once and reuse (like M4) with the ability to alter objects configuration attributes from defaults. 3) Allow data type and value verification. This is not done today and frequently crops up in FW bug reports. Common Topology Classes ----------------------- Topology today has some common classes that are often reused throughout with slightly altered configurations. i.e. widgets (components), pipelines, dais and controls. Topology2.0 introduces the high level concept of reusable "class" like definition for a AIF_IN/AIF_OUT type object that can be used to create topology objects. Common Topology Attributes -------------------------- Topology defines a lot of attributes per object with different types and constraints. Today there is no easy way to validate type or constraints and this can lead to many hard to find problems in FW at runtime. A new keyword "DefineAttribute" has been added to define attribute type, size, min value, max value, enum_values. This then allows alsatplg to validate each topology object attribute. Topology Classes define the list of attributes that they use and whether the attribute is mandatory, can be overridden by parent users or is immutable. This also helps alsatplg emit the appropriate errors for attribute misuse. Attribute validation ------------------------- One of the main features of Topology2.0 is the ability to add constraints to attributes to define a set of valid values, specify min/max values etc in the class definition. Attribute Inheritance ---------------------- One of the key features of Topology2.0 is howthe attribute values are propagated from a parent object to a child object. This is accomplished by adding attributes/arguments with the same name for a parent and an object. By doing so, when creating a child object, the value for the common attribute is populated from the parent. If the value is provided in the child object instance, then it overrides the value coming from the parent. **ALSA Conf Parser** ---------------- All the changes being proposed and discussed here must be 100% compliant with the ALSA conf parser. i.e. no syntax changes or changes to semantics for any existing keyword. It's intended that there will be NO changes to the ALSA conf parser Request URL : https://github.com/alsa-project/alsa-utils/pull/86 Patch URL : https://github.com/alsa-project/alsa-utils/pull/86.patch Repository URL: https://github.com/alsa-project/alsa-utils

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Re: ASoc / PCM recording-related regression between v5.4 and v5.5
by Hin-Tak Leung 27 Apr '21

27 Apr '21

Hi all, I have come up with a second way of fixing the regression (the first one was a few commits before that): https://github.com/HinTak/seeed-voicecard/commit/19067f3333d7759bdf626e62bc… As far as I understand it, the breakage came from the somewhat unsual design of the SoC device: it has both recording / ADC and a playback / DAC codecs. (a pair of ADC ones, actually) to do real-time echo cancellation / noise filtering applications. The recording codecs use the playback codec's clock; and the playback codec's trigger needs to be called, before the clock can start. I am not entirely happy with the rest of the vendor's code in terms of what code is where of event sequences, but I'll leave that for now. Would appreciate comments, etc.

1 0

[PATCH v2 0/5] ASoC: clock provider clean-up
by Jerome Brunet 27 Apr '21

27 Apr '21

The purpose of this patchset it remove the use the clk member of 'struct clk_hw' in ASoC. 'struct clk' is a per-user reference to an actual clock. In the future, the clk member in 'struct clk_hw' may go away. The usage of this member by a clock provider usually falls into either of following categories: * Mis-usage of the clock consumer API by a clock provider. * Clock provider also being a user of its own clocks. In this case the provider should request a 'struct clk' through the appropriate API instead of poking in 'struct clk_hw' internals. v1 [0] -> v2: * finalize lpass provider move to devm [0]: https://lore.kernel.org/r/20210410111356.467340-1-jbrunet@baylibre.com Jerome Brunet (5): ASoC: stm32: properly get clk from the provider ASoC: wcd934x: use the clock provider API ASoC: rt5682: clock driver must use the clock provider API ASoC: lpass: use the clock provider API ASoC: da7219: properly get clk from the provider sound/soc/codecs/da7219.c | 5 ++++- sound/soc/codecs/lpass-va-macro.c | 7 ++----- sound/soc/codecs/lpass-wsa-macro.c | 11 +++-------- sound/soc/codecs/rt5682.c | 6 +++--- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/stm/stm32_sai_sub.c | 5 ++++- 6 files changed, 20 insertions(+), 20 deletions(-) -- 2.31.1

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[PATCH 1/2] ALSA: pcm: Add support for 705.6KHz and 768KHz sample rate
by Shengjiu Wang 27 Apr '21

27 Apr '21

Some high resolution codecs support 705.6KHz and 768KHz rates. So extend supported sample rate to 768kHz in pcm.h. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- include/sound/pcm.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2e1200d17d0c..88056824ffec 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -121,6 +121,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ #define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ #define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ +#define SNDRV_PCM_RATE_705600 (1<<15) /* 705600Hz */ +#define SNDRV_PCM_RATE_768000 (1<<16) /* 768000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -136,6 +138,9 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000|\ SNDRV_PCM_RATE_352800|\ SNDRV_PCM_RATE_384000) +#define SNDRV_PCM_RATE_8000_768000 (SNDRV_PCM_RATE_8000_768000|\ + SNDRV_PCM_RATE_705600|\ + SNDRV_PCM_RATE_768000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) -- 2.27.0

3 3

ASoC: Serving gpio_desc in codec driver with I2C GPIO expander?
by Pavel Hofman 27 Apr '21

27 Apr '21

Hi, Please is there any reasonable way to use an existing codec driver controlled by gpio_desc (e.g. https://elixir.bootlin.com/linux/v5.12/source/sound/soc/codecs/ak4458.c#L47 ) where the actual pins are provided by outputs of an I2C GPIO expander? I am trying to decide whether to base my setup (ADC + DAC + clock generator + I2S DAI, all controlled via I2C + pins via I2C GPIO expander) purely on ASoC, or whether to keep the ASoC driver minimal (like asoc-simple-card) and do all the I2C settings in userspace as part of the software which writes/reads audio data. It seems the ASoC path is way more complicated as it involves multiple layers of abstractions prone to errors from my side. Thank you very much for your opinion and suggestions. Best regards, Pavel.

1 0

[PATCH] ALSA: virtio: fix kernel-doc
by Pierre-Louis Bossart 27 Apr '21

27 Apr '21

make W=1 warning: sound/virtio/virtio_ctl_msg.c:70: warning: expecting prototype for virtsnd_ctl_msg_request(). Prototype was for virtsnd_ctl_msg_response() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/virtio/virtio_ctl_msg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/virtio/virtio_ctl_msg.c b/sound/virtio/virtio_ctl_msg.c index 26ff7e7cc041..18dc5aca2e0c 100644 --- a/sound/virtio/virtio_ctl_msg.c +++ b/sound/virtio/virtio_ctl_msg.c @@ -61,7 +61,7 @@ void *virtsnd_ctl_msg_request(struct virtio_snd_msg *msg) } /** - * virtsnd_ctl_msg_request() - Get a pointer to the response header. + * virtsnd_ctl_msg_response() - Get a pointer to the response header. * @msg: Control message. * * Context: Any context. -- 2.25.1

2 1

[PATCH 0/7] ASoC: adds new .get_fmt support
by Kuninori Morimoto 27 Apr '21

27 Apr '21

Hi Mark These are a little bit challenging patch-set. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. If we can use it, Sound Card Driver user no longer need to select it, and I want to use this style on new audio-graph-card2. This patch-set adds new .get_fmt callback which indicate available dai_fmt to ALSA SoC Framework. Of course Sound Card Driver can still select dai_link->dai_fmt, same as before. If Sound Card Driver didn't set it, and if both CPU / Codec had .get_fmt callback, dai_link->dai_fmt will be automatically selected. Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .get_fmt support ASoC: pcm3168a: add .get_fmt support ASoC: rsnd: add .get_fmt support ASoC: fsi: add .get_fmt support ASoC: hdmi-codec: add .get_fmt support include/sound/soc-dai.h | 35 ++++++ sound/soc/codecs/ak4613.c | 10 ++ sound/soc/codecs/hdmi-codec.c | 19 ++++ sound/soc/codecs/pcm3168a.c | 13 +++ sound/soc/sh/fsi.c | 13 +++ sound/soc/sh/rcar/core.c | 16 +++ sound/soc/soc-core.c | 195 +++++++++++++++++++++++----------- sound/soc/soc-dai.c | 19 ++++ sound/soc/soc-utils.c | 26 +++++ 9 files changed, 284 insertions(+), 62 deletions(-) -- 2.25.1

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[PATCH v2 0/2] Fix couple of bugs reported by Canonical
by Vitaly Rodionov 26 Apr '21

26 Apr '21

CirrusLogic: Cracking noises appears in built-in speaker when output volume is set >80% BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1923557i CirrusLogic: The default input volume is "0%" BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1924997 In v2: - Add additional information in commit messages. Stefan Binding (2): ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to -26 dB ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye sound/pci/hda/patch_cirrus.c | 38 ++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) -- 2.25.1

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[PATCH] sound/isa/ib: Fix two use after free in snd_sb_qsound_build
by Lv Yunlong 26 Apr '21

26 Apr '21

In snd_sb_qsound_build, snd_ctl_add(..,p->qsound_switch...) and snd_ctl_add(..,p->qsound_space..) are called. But the second arguments of snd_ctl_add() could be freed via snd_ctl_add_replace() ->snd_ctl_free_one(). After the error code is returned, snd_sb_qsound_destroy(p) is called in __error branch. But in snd_sb_qsound_destroy(), the freed p->qsound_switch and p->qsound_space are still used by snd_ctl_remove(). My patch set p->qsound_switch and p->qsound_space to NULL if snd_ctl_add() failed to avoid the uaf bugs. But these codes need to further be improved with the code style. Signed-off-by: Lv Yunlong <lyl2019(a)mail.ustc.edu.cn> --- sound/isa/sb/sb16_csp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 8635a2b6b36b..4789345a8fdd 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1045,10 +1045,14 @@ static int snd_sb_qsound_build(struct snd_sb_csp * p) spin_lock_init(&p->q_lock); - if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) + if ((err = snd_ctl_add(card, p->qsound_switch = snd_ctl_new1(&snd_sb_qsound_switch, p))) < 0) { + p->qsound_switch = NULL; goto __error; - if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) + } + if ((err = snd_ctl_add(card, p->qsound_space = snd_ctl_new1(&snd_sb_qsound_space, p))) < 0) { + p->qsound_space = NULL; goto __error; + } return 0; -- 2.25.1

2 1

[GIT PULL] ASoC updates for v5.13
by Mark Brown 26 Apr '21

26 Apr '21

The following changes since commit bf05bf16c76bb44ab5156223e1e58e26dfe30a88: Linux 5.12-rc8 (2021-04-18 14:45:32 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.13 for you to fetch changes up to ffc9841d5200a484ea0ecc645157b4d7b873f3a6: Merge remote-tracking branch 'asoc/for-5.13' into asoc-next (2021-04-23 19:01:02 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.13 A lot of changes here for quite a quiet release in subsystem terms - there's been a lot of fixes and cleanups all over the subsystem both from generic work and from people working on specific drivers. - More cleanup and consolidation work in the core and the generic card drivers from Morimoto-san. - Lots of cppcheck fixes for Pierre-Louis Brossart. - New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358 accessory detection, and Realtek RT1019, RT1316, RT711 and RT715. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl_ssi: Fix TDM slot setup for I2S mode Annaliese McDermond (2): ASoC: tlv320aic32x4: Increase maximum register in regmap ASoC: tlv320aic32x4: Register clocks before registering component Argus Lin (2): ASoC: mt6359: add ASoC mt6359 ASoC accdet jack document ASoC: mediatek: mt6359: add MT6359 accdet jack driver Arnd Bergmann (1): ASoC: tegra: mark runtime-pm functions as __maybe_unused Benjamin Rood (1): ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe Bhaskar Chowdhury (5): ASoC: ti: Fix a typo in the file ams-delta.c ASoC: Fix a typo in the file rt5682.txt ASoC: Intel: Fix a typo ASoC: Intel: Fix a typo ASoC: Intel: Fix a typo Brent Lu (1): ASoC: Intel: sof_rt5682: Add ALC1015Q-VB speaker amp support Cezary Rojewski (1): ASoC: Intel: Skylake: Compile when any configuration is selected Chen Lifu (1): ASoC: sti: sti_uniperif: add missing MODULE_DEVICE_TABLE Christophe JAILLET (2): ASoC: cs35l35: Fix an error handling path in 'cs35l35_i2c_probe()' ASoC: cs35l36: Fix an error handling path in 'cs35l36_i2c_probe()' Codrin Ciubotariu (7): ASoC: mchp-i2s-mcc: Add compatible for SAMA7G5 ASoC: mchp-i2s-mcc: Add multi-channel support for I2S and LEFT_J formats ASoC: mchp-i2s-mcc: Add support to select TDM pins ASoC: mchp-i2s-mcc: Add FIFOs support ASoC: mchp,i2s-mcc: Add SAMA7G5 to binding ASoC: mchp, i2s-mcc: Add property to specify pin pair for TDM ASoC: convert Microchip I2SMCC binding to yaml Colin Ian King (5): ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx ASoC: codecs: lpass-rx-macro: remove redundant initialization of variable hph_pwr_mode ASoC: Intel: boards: sof-wm8804: add check for PLL setting ASoC: fsl: fsl_easrc: Fix uninitialized variable st2_mem_alloc ASoC: mediatek: mt6359: Fix spelling mistake "reate" -> "create" Daniel Baluta (1): ASoC: core: Don't set platform name when of_node is set David Ward (5): ASoC: rt286: Fix upper byte in DMIC2 configuration ASoC: rt286: Make RT286_SET_GPIO_* readable and writable ASoC: rt286: Configure combo jack for headphones ASoC: rt298: Configure combo jack for headphones ASoC: rt286: Generalize support for ALC3263 codec Dinghao Liu (1): ASoC: codecs: Fix runtime PM imbalance in tas2552_probe Dmitry Baryshkov (1): ASoC: q6afe-clocks: fix reprobing of the driver Dmitry Osipenko (16): ASoC: tegra20: ac97: Add reset control ASoC: tegra20: i2s: Add reset control ASoC: tegra30: i2s: Restore hardware state on runtime PM resume ASoC: tegra30: ahub: Switch to use reset-bulk API ASoC: tegra20: spdif: Correct driver removal order ASoC: tegra20: spdif: Remove handing of disabled runtime PM ASoC: tegra20: i2s: Add system level suspend-resume callbacks ASoC: tegra20: i2s: Correct driver removal order ASoC: tegra20: i2s: Use devm_clk_get() ASoC: tegra20: i2s: Remove handing of disabled runtime PM ASoC: tegra30: i2s: Correct driver removal order ASoC: tegra30: i2s: Use devm_clk_get() ASoC: tegra30: i2s: Remove handing of disabled runtime PM ASoC: tegra30: ahub: Reset global variable ASoC: tegra30: ahub: Correct suspend-resume callbacks ASoC: tegra30: ahub: Remove handing of disabled runtime PM Fabio Estevam (3): ASoC: wm8524: Do not print probe defer error ASoC: sgtl5000: Fix identation of .driver elements ASoC: mx27vis: Remove unused file Guangqing Zhu (1): ASoC: codecs: wcd934x: Fix missing IRQF_ONESHOT as only threaded handler Guennadi Liakhovetski (3): ASoC: SOF: fix debugfs initialisation error handling ASoC: SOF: only allocate debugfs cache buffers for IPC flood entries ASoC: SOF: remove superfluous NULL check in debugfs read Gyeongtaek Lee (1): ASoC: soc-compress: lock pcm_mutex to resolve lockdep error Hans de Goede (33): ASoC: rt5670: Remove 'OUT Channel Switch' control ASoC: rt5670: Remove 'HP Playback Switch' control ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer settings ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: es8316: Simplify adc_pga_gain_tlv table ASoC: rt5645: Move rt5645_platform_data to sound/soc/codecs/rt5645.c ASoC: rt5645: The ALC3270 variant does not have a headset-mic pin ASoC: Intel: bytcr_rt5640: Enable jack-detect support on Asus T100TAF ASoC: rt5640: Rename 'Mono DAC Playback Volume' to 'DAC2 Playback Volume' ASoC: Intel: bytcr_rt5640: Add used AIF to the components string mfd: arizona: Drop arizona-extcon cells extcon: arizona: Fix some issues when HPDET IRQ fires after the jack has been unplugged extcon: arizona: Fix various races on driver unbind extcon: arizona: Fix flags parameter to the gpiod_get("wlf,micd-pol") call extcon: arizona: Always use pm_runtime_get_sync() when we need the device to be awake ASoC/extcon: arizona: Move arizona jack code to sound/soc/codecs/arizona-jack.c ASoC: arizona-jack: Move jack-detect variables to struct arizona_priv ASoC: arizona-jack: Use arizona->dev for runtime-pm ASoC: arizona-jack: convert into a helper library for codec drivers ASoC: arizona-jack: Use snd_soc_jack to report jack events ASoC: arizona-jack: Cleanup logging ASoC: arizona: Make the wm5102, wm5110, wm8997 and wm8998 drivers use the new jack library ASoC: Intel: bytcr_wm5102: Add jack detect support ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Hi8 tablet ASoC: Intel: Unify the thinkpad10 and aegex10 byt-match quirks ASoC: Intel: Baytrail: Add quirk for the Dell Venue 10 Pro 5055 tablet ASoC: Intel: cht_bsw_rt5672: Add support for Bay Trail CR / SSP0 ASoC: rt5670: Add a quirk for the Dell Venue 10 Pro 5055 ASoC: rt5670: Add a rt5670_components() helper ASoC: Intel: cht_bsw_rt5672: Set card.components string Jack Yu (13): ASoC: rt1015: fix i2c communication error ASoC: rt1015: enable BCLK detection after calibration ASoC: rt715-sdca: Add RT715 sdca vendor-specific driver ASoC: rt1015p: add acpi device id for rt1015p ASoC: rt1019: add rt1019 amplifier driver ASoC: rt1015: Add bclk detection and dc detection ASoC: rt1019: add address-cells and size-cells information ASoC: rt1011: remove pack_id check in rt1011 ASoC: rt1019: remove registers to sync with rt1019 datasheet ASoC: rt715: modification for code simplicity ASoC: rt715: add main capture switch and main capture volume ASoC: rt715: remove kcontrols which no longer be used ASoC: rt1015p: add support on format S32_LE Jaroslav Kysela (6): ALSA: control - introduce snd_ctl_notify_one() helper ALSA: control - add layer registration routines ALSA: control - add generic LED trigger module as the new control layer ALSA: HDA - remove the custom implementation for the audio LED trigger ALSA: control - add sysfs support to the LED trigger module ALSA: led control - add sysfs kcontrol LED marking layer Jaska Uimonen (3): ASoC: SOF: parse multiple SSP DAI and hw configs ASoC: SOF: Intel: HDA: add hw params callback for SSP DAIs ASoC: SOF: match SSP config with pcm hw params Jerome Brunet (8): ASoC: meson: axg-frddr: set fifo depth according to the period ASoC: meson: axg-fifo: add NO_PERIOD_WAKEUP support ASoC: meson: axg-frddr: fix fifo depth on g12 and sm1 ASoC: stm32: properly get clk from the provider ASoC: wcd934x: use the clock provider API ASoC: rt5682: clock driver must use the clock provider API ASoC: lpass: use the clock provider API ASoC: da7219: properly get clk from the provider Jiri Prchal (6): ASoC: codecs: tlv320aic3x: move model definitions ASoC: codecs: tlv320aic3x: rename probe function ASoC: codecs: tlv320aic3x: move I2C to separated file ASoC: codecs: tlv320aic3x: add SPI support ASoC: codecs: tlv320aic3x: add AIC3106 ASoC: codecs: tlv320aic3x: add SPI to the DT binding Jon Hunter (1): ASoC: soc-core: Prevent warning if no DMI table is present Julia Lawall (4): ASoC: mmp-sspa: drop unneeded snd_soc_dai_set_drvdata ASoC: mxs-saif: drop unneeded snd_soc_dai_set_drvdata ASoC: sun4i-i2s: drop unneeded snd_soc_dai_set_drvdata ASoC: fsl: drop unneeded snd_soc_dai_set_drvdata Kai Vehmanen (2): ASoC: SOF: Intel: hda: turn off display power in resume ASoC: SOF: Intel: hda: drop display power on/off in D0i3 flows Keyon Jie (2): ASoC: SOF: add a helper to get topology configured mclk ASoC: intel: sof_rt5682: use the topology mclk Krzysztof Kozlowski (2): ASoC: constify of_phandle_args in snd_soc_get_dai_name() ASoC: simple-card: fix possible uninitialized single_cpu local variable Kuninori Morimoto (82): ASoC: soc-pcm: remove strange format storing ASoC: soc-pcm: unpack dpcm_init_runtime_hw() ASoC: soc-pcm: add dpcm_runtime_setup_fe() ASoC: soc-pcm: add dpcm_runtime_setup() ASoC: soc-pcm: unpack dpcm_set_fe_runtime() ASoC: rsnd: cleanup ppcheck warning for ssiu.c ASoC: rsnd: cleanup ppcheck warning for ssi.c ASoC: rsnd: cleanup ppcheck warning for core.c ASoC: rsnd: cleanup ppcheck warning for cmd.c ASoC: rsnd: cleanup ppcheck warning for adg.c ASoC: soc-pcm: check DAI activity under soc_pcm_apply_symmetry() ASoC: soc-pcm: add soc_cpu/codec_dai_name() macro ASoC: soc-pcm: direct copy at snd_soc_set_runtime_hwparams() ASoC: soc-pcm: add soc_pcm_update_symmetry() ASoC: soc-pcm: add soc_hw_sanity_check() ASoC: soc-pcm: fixup dpcm_be_dai_startup() user count ASoC: soc-pcm: remove unneeded !rtd->dai_link check ASoC: soc-pcm: share DPCM BE DAI stop operation ASoC: soc-pcm: indicate error message at soc_pcm_open() ASoC: soc-pcm: indicate error message at soc_pcm_hw_params() ASoC: soc-pcm: indicate error message at soc_pcm_prepare() ASoC: soc-pcm: indicate error message at dpcm_path_get() ASoC: soc-pcm: indicate error message at dpcm_be_dai_trigger() ASoC: soc-pcm: indicate error message at dpcm_apply_symmetry() ASoC: soc-pcm: indicate error message at dpcm_run_update_startup/shutdown() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_startup() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_hw_params() ASoC: soc-pcm: indicate error message at dpcm_fe/be_dai_prepare() ASoC: soc-pcm: don't indicate error message for soc_pcm_hw_free() ASoC: soc-pcm: don't indicate error message for dpcm_be_dai_hw_free() ASoC: don't indicate error message for snd_soc_[pcm_]dai_xxx() ASoC: don't indicate error message for snd_soc_[pcm_]component_xxx() ASoC: soc.h: add asoc_link_to_cpu/codec/platform() macro ASoC: soc.h: fixup return timing for snd_soc_fixup_dai_links_platform_name() ASoC: soc.h: return error if multi platform at snd_soc_fixup_dai_links_platform_name() ASoC: simple-card-utils.c: remove old comment ASoC: simple-card-utils: alloc dai_link information for CPU/Codec/Platform ASoC: audio-graph: count DAI / link numbers as in order ASoC: audio-graph: cleanup graph_for_each_link() ASoC: simple-card: count DAI / link numbers as in order ASoC: simple-card: cleanup graph_for_each_link() ASoC: soc-core: add comment for rtd freeing ASoC: soc-core: use device_unregister() if rtd allocation failed ASoC: simple-card-utils: enable flexible CPU/Codec/Platform ASoC: simple-card-utils: share dummy DAI and reduce memory ASoC: simple-card-utils: setup dai_props cpu_dai/codec_dai at initial timing ASoC: simple-card-utils: remove li->dais/li->conf ASoC: simple-card-utils: use for_each_prop_xxx() ASoC: simple-card-utils: remove asoc_simple_parse_xxx() ASoC: simple-card-utils: care multi DAI at asoc_simple_clean_reference() ASoC: simple-card-utils: indicate dai_fmt if exist ASoC: simple-card-utils: indicate missing CPU/Codec numbers for debug ASoC: simple-card-utils: add simple_props_to_xxx() macro ASoC: simple-card-utils: multi support at asoc_simple_canonicalize_cpu/platform() ASoC: simple-card-utils: tidyup debug info for clock ASoC: simple-card-utils: tidyup dev_dbg() to use 1 line ASoC: simple-card-utils: tidyup asoc_simple_parse_convert() ASoC: rsnd: call rsnd_ssi_master_clk_start() from rsnd_ssi_init() ASoC: rsnd: check all BUSIF status when error ASoC: rsnd: add rsnd_ssi_busif_err_status_clear() ASoC: rsnd: add rsnd_ssi_busif_err_irq_enable/disable() ASoC: rsnd: add usage for SRC ASoC: simple-card: remove unused variable from simple_parse_of() ASoC: simple-card: use asoc_link_to_xxx() macro ASoC: simple-card: use simple_props_to_xxx() macro ASoC: audio-graph: use asoc_link_to_xxx() macro ASoC: audio-graph: use simple_props_to_xxx() macro ASoC: soc-pcm: don't use "name" on __soc_pcm_params_symmetry() macro ASoC: soc-pcm: indicate DAI name if soc_pcm_params_symmetry() failed ASoC: soc-utils: add snd_soc_component_is_dummy() ASoC: soc-pcm: ignore dummy-DAI at soc_pcm_params_symmetry() ASoC: audio-graph: move audio_graph_card_probe() to simple-card-utils.c ASoC: audio-graph: move audio_graph_remove() to simple-card-utils.c ASoC: audio-graph: check ports if exists ASoC: audio-graph: remove "audio-graph-card, " preix support ASoC: audio-graph: remove unused "node" from graph_parse_mclk_fs() ASoC: audio-graph: remove Platform support ASoC: audio-graph: add graph_parse_node() ASoC: audio-graph: add graph_link_init() ASoC: simple-card: add simple_parse_node() ASoC: simple-card: add simple_link_init() ASoC: audio-graph: tidyup graph_dai_link_of_dpcm() Libin Yang (2): ASoC: SOF: Kconfig: fix typo of SND_SOC_SOF_PCI ASoC: SOF: Intel: add missing use_acpi_target_states for TGL platforms Lucas Tanure (12): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset ASoC: cs42l42: Remove power if the driver is being removed ASoC: cs42l42: Disable regulators if probe fails ASoC: cs42l42: Provide finer control on playback path ASoC: cs42l42: Set clock source for both ways of stream ASoC: cs42l42: Add Capture Support ASoC: cs42l42: Report jack and button detection ASoC: cs42l42: Use bclk from hw_params if set_sysclk was not called Lukasz Majczak (1): ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup function Mac Chiang (3): ASoC: Intel: add max98390 echo reference support ASoC: Intel: Boards: cml_da7219_max98390: add capture stream for echo reference ASoc: Intel: board: add BE DAI link for WoV Mark Brown (57): Merge series "ASoC: rt5670: Various kcontrol fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "AsoC: rt5640/rt5651: Volume control fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: qcom: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt*: Constify static structs" from Rikard Falkeborn <rikard.falkeborn(a)gmail.com>: Merge series "ASoC: fsl: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "soc-pcm: tidyup snd_pcm_hardware setup for FE/BE" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: samsung: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: core: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: sh: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "drop unneeded snd_soc_dai_set_drvdata" from Julia Lawall <Julia.Lawall(a)inria.fr>: Merge series "ASoC: rsnd: cleanup ppcheck warning for Renesas sound driver" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Add I2S-MCC support for Microchip's SAMA7G5" from Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com>: Merge series "ASoC: Intel: soc-acpi: remove unused TigerLake configurations" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: realtek: fix cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: remove more make W=1 warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal(a)opensource.cirrus.com>: Merge series "Add I2S-MCC support for Microchip's SAMA7G5" from Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com>: Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal(a)opensource.cirrus.com>: Merge series "ASoC: codecs: wolfson: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: mediatek: mt8183-mt6358: support machine driver for rt1015p" from Tzung-Bi Shih <tzungbi(a)google.com>: Merge series "ASoC: soc-pcm: cleanup each functions" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: samsung: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "AsoC: rt5640/rt5651: Volume control fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: SOF: debug: cleanups" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Fix reset controls and RPM of NVIDIA Tegra ASoC drivers" from Dmitry Osipenko <digetx(a)gmail.com>: Merge series "ASoC: codecs: remove cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: tidyup error message timing" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge tag 'ib-mfd-extcon-v5.13' of git://git.kernel.org/pub/scm/linux/kernel/git/lee/mfd into asoc-5.13 Merge series "MFD/extcon/ASoC: Rework arizona codec jack-detect support" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "ASoC: soc.h: small cleanups" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "Add audio driver base on rpmsg on i.MX platform" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Merge series "Add mediatek MT6359 ASoC accdet jack driver" from Argus Lin <argus.lin(a)mediatek.com>: Merge series "ASoC: simple-card: cleanup and prepare for Multi CPU/Codec support" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge tag 'mute-led-rework' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-5.13 Merge series "ASoC: remove cppcheck warnings for multiple SOCs" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: soc-core: tidyup error handling for rtd" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: dt-bindings: Rework audio-graph-port schema" from Rob Herring <robh(a)kernel.org>: Merge series "ASoC: remove cppchecks warnings on lm49453 and da732x" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "kbl_da7219_max9357a machine changes for wov and MST" from vamshi.krishna.gopal(a)intel.com Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.corp-partner.google.com>: Merge series "Adds SPI support" from Jiri Prchal <jiri.prchal(a)aksignal.cz>: Merge series "ASoC: simple-card-utils: prepare for multi support" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: SOF: simplify nocodec mode" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: SOF: simplify probe and report errors" from Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: descriptor corrections for TGL and ADL" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: simple-card-utils: prepare for multi support" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: rsnd: tidyup Renesas sound" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "kunit: Fix formatting of KUNIT tests to meet the standard" from Nico Pache <npache(a)redhat.com>: Merge series "ASoC: simple-card/audio-graph: adjust to multi CPU/Codec" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: soc-pcm: ignore dummy-DAI at soc_pcm_params_symmetry()" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge series "ASoC: Intel: add new TGL/ADL configurations" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: remove more cppcheck warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward(a)gatech.edu>: Merge series "ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack" from David Ward <david.ward(a)gatech.edu>: Merge series "ASoC: audio-graph: cleanups" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2 Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus Merge remote-tracking branch 'asoc/for-5.13' into asoc-next Mikhail Durnev (1): ASoC: rsnd: core: Check convert rate in rsnd_hw_params Muhammad Usama Anjum (1): ASoC: fsl: sunxi: remove redundant dev_err call Nick Desaulniers (1): ASoC: Intel: Skylake: skl-topology: fix -frame-larger-than Nico Pache (1): ASoC: topology: adhere to KUNIT formatting standard Niklas Carlsson (2): ASoC: sigmadsp: Disable cache mechanism for readbacks ASoC: adau17x1: Avoid overwriting CHPF Peter Ujfalusi (2): ASoC: SOF: core: Add missing error prints to device probe operation ASoC: SOF: Simplify sof_probe_complete handling for acpi/pci/of Philipp Zabel (1): reset: Add reset_control_bulk API Pierre-Louis Bossart (147): ASoC: SOF: Intel: unregister DMIC device on probe error ASoC: qcom: lpass-hdmi: remove useless return ASoC: qcom: lpass-platform: remove useless assignment ASoC: qcom: q6dsp-dai: clarify expression ASoC: qcom: q6afe: remove useless assignments ASoC: qcom: q6afe: align function prototype ASoC: qcom: q6asm: align function prototypes ASoC: wcd-clsh-v2: align function prototypes ASoC: wcd9335: clarify return value ASoC: wcd934x: remove useless return ASoC: lpass-wsa-macro: add missing test ASoC: fsl: fsl_asrc: remove useless assignment ASoC: fsl: fsl_dma: remove unused variable ASoC: fsl: fsl_easrc: remove useless assignments ASoC: fsl: fsl_esai: clarify expression ASoC: fsl: fsl_ssi: remove unnecessary tests ASoC: fsl: imx-hdmi: remove unused structure members ASoC: fsl: mpc5200: signed parameter in snprintf format ASoC: fsl: mpc8610: remove useless assignment ASoC: fsl: p1022_ds: remove useless assignment ASoC: samsung: i2s: remove unassigned variable ASoC: samsung: s3c24xx_simtec: add missing error check ASoC: samsung: smdk_wm8994: add missing return ASoC: samsung: snow: remove useless test ASoC: soc-ops: remove useless assignment ASoC: soc-pcm: remove redundant assignment ASoC: soc-pcm: remove shadowing variable ASoC: soc-pcm: add error log ASoC: soc-topology: clarify expression ASoC: generic: simple-card-utils: remove useless assignment ASoC: sh: dma-sh7760: remove unused variable ASoC: sh: rcar: align function prototypes ASoC: sh: rcar: simplify return ASoC: sh: rcar: ctu: add missing error check ASoC: sh: rcar: ssi: remove redundant assignment ASoC: sh: siu_pcm: remove useless assignment ASoC: sh: siu_pcm: remove unused variable ASoC: Intel: soc-acpi: remove unused TGL table with rt5682 only ASoC: Intel: soc-acpi: remove TGL RVP mixed SoundWire/TDM config ASoC: codecs: nau8825: fix kernel-doc ASoC: soc-dai: fix kernel-doc ASoC: Intel: atom: fix kernel-doc ASoC: rt1011: use logical OR ASoC: rt1011: remove redundant test ASoC: rt1011: clarify expression ASoC: rt1015: clarify expression ASoC: rt1016: clarify expression ASoC: rt1305: clarify expression ASoC: rt1308: clarify expression ASoC: rt5640: clarify expression ASoC: rt5645: use logical OR ASoC: rt5645: clarify expression ASoC: rt5651: clarify expression ASoC: rt5651: remove useless assignment ASoC: rt5659: clarify expression ASoC: rt5660: clarify expression ASoC: rt5663: clarify expression ASoC: rt5665: clarify expression ASoC: rt5668: clarify expression ASoC: rt5668: remove useless assignments ASoC: rt5670: clarify expression ASoC: rt5677: clarify expression ASoC: rt5677: remove useless assignment ASoC: rt5682: clarify expression ASoC: rt5682: remove useless assignments ASoC: rt5682: remove useless initialization ASoC: cs4270: fix kernel-doc ASoC: jz4760: fix set but not used warning ASoC: rt5631: fix kernel-doc warning ASoC: sigmadsp-regmap: fix kernel-doc warning ASoC: amd: renoir: remove invalid kernel-doc comment ASoC: fsl: fsl_ssi: fix kernel-doc warning ASoC: fsl: fsl_easrc: fix kernel-doc warning ASoC: Intel: bytcr_wm5102: remove unused static variable ASoC: qcom: q6dsp: fix kernel-doc warning ASoC: arizona: fix function argument ASoC: madera: align function prototype ASoC: wm2200: remove unused structure ASoC: wm8903: remove useless assignments ASoC: wm8958-dsp2: rename local 'control' arrays ASoC: wm8978: clarify expression ASoC: wm8994: align function prototype ASoC: wm8996: clarify expression ASoC: wm_adsp: simplify return value ASoC: wm_hubs: align function prototype ASoC: soc-core: fix DMI handling ASoC: mediatek: mtk-btcvsd: remove useless assignment ASoC: mediatek: mt2701: align function prototype ASoC: mediatek: mt2701: rename shadowed array ASoC: mediatek: mt8173: rename local irq variable ASoC: samsung: tm2_wm5110: check of of_parse return value ASoC: samsung: tm2_wm5110: remove shadowed variable ASoC: ab8500-codec: remove useless structure ASoC: ad1836: remove useless return ASoC: adau1977: remove useless return ASoC: cros_ec_codec: remove null pointer dereference warning ASoC: cx2070x: remove useless assignment ASoC: cx2070x: remove duplicate else branch ASoC: da7219-aad: remove useless initialization ASoC: hdac_hdmi: remove useless initializations ASoC: hdac_hdmi: align function arguments ASoC: hdmi-codec: remove useless initialization ASoC: hdmi-codec: remove unused spk_mask member ASoC: max98090: remove useless assignment ASoC: mt6358: remove useless initializations ASoC: mt6359: remove useless assignment ASoC: nau8825: remove useless assignment ASoC: pcm1681: remove useless assignment ASoC: sigmadsp: align function prototype ASoC: sti-sas: remove unused struct members ASoC: tas2562: remove useless assignment ASoC: tas2562: remove warning on return value ASoC: tas2770: remove useless initialization ASoC: tlv320dac33: clarify expression ASoC: tscs454: remove useless test on PLL disable ASoC: lm49453: fix useless assignment before return ASoC: da732x: simplify code ASoC: amd: renoir: acp3x-pdm-dma: remove unnecessary assignments ASoC: atmel: fix shadowed variable ASoC: atmel: atmel-i2s: remove useless initialization ASoC: bcm: cygnus_ssp: remove useless initialization ASoC: meson: axg-tdmin: remove useless assignment ASoC: meson: axg-tdmout: remove useless assignment ASoC: pxa: remove useless assignment ASoC: sti: sti_uniperif: add missing error check ASoC: sti: uniperif: align function prototypes ASoC: stm: stm32_adfsdm: fix snprintf format string ASoC: sunxi: sun8i-codec: clarify expression ASoC: tegra: tegra20_das: clarify expression ASoC: tegra: tegra20_das: align function prototypes ASoC: ti: omap-abe-twl6040: remove useless assignment ASoC: ti: omap-mcsp: remove duplicate test ASoC: ux500: mop500: rename shadowing variable ASoC: ux500: mop500: align function prototype ASoC: SOF: add Kconfig option for probe workqueues ASoC: soc-acpi: add new fields for mach_params ASoC: SOF: change signature of set_mach_params() callback ASoC: SOF: Intel: update set_mach_params() ASoC: SOF: pcm: export snd_pcm_dai_link_fixup ASOC: SOF: simplify nocodec mode ASoC: Intel: soc-acpi: add ADL SoundWire base configurations ASoC: Intel: soc-acpi: add ADL jack-less SoundWire configurations ASoC: soc-acpi: remove useless initialization ASoC: soc-core: fix signed/unsigned issue ASoC: soc-core: fix always-false condition ASoC: codecs: lpass-rx-macro: remove useless return ASoC: codecs: rt5682: clarify expression Ranjani Sridharan (3): ASoC: SOF: Intel: hda: enable async suspend ASoC: SOF: topology: remove useless code ASoC: SOF: use current DAI config during resume Richard Fitzgerald (3): ASoC: cs42l42: Wait at least 150us after writing SCLK_PRESENT ASoC: cs42l42: Only start PLL if it is needed ASoC: cs42l42: Wait for PLL to lock before switching to it Rikard Falkeborn (4): ASoC: rt*: Constify static struct sdw_slave_ops ASoC: rt*: Constify static struct snd_soc_dai_ops ASoC: rt*: Constify static struct acpi_device_id ASoc: rt5631: Constify static struct coeff_clk_div Rob Herring (5): ASoC: dt-bindings: nvidia, tegra210-ahub: Add missing child nodes ASoC: dt-bindings: Move port/ports properties out of audio-graph-port.yaml ASoC: dt-bindings: Use OF graph schema ASoC: dt-bindings: socionext: Use audio-graph-port schema ASoC: mt6359: Drop ASoC mt6359 ASoC accdet jack document Robin Gong (1): ASoC: fsl_asrc_dma: request dma channel from specific controller Ryan Lee (1): ASoC: max98373: Added controls for autorestart config Sathya Prakash M R (1): ASoC: SOF: Intel: Update ADL P to use its own descriptor Shengjiu Wang (25): ASoC: ak4458: Add MODULE_DEVICE_TABLE ASoC: ak5558: Add MODULE_DEVICE_TABLE ASoC: fsl_sai: Add pm qos cpu latency support ASoC: wm8962: Relax bit clock divider searching ASoC: fsl_sai: Don't use devm_regmap_init_mmio_clk ASoC: fsl_esai: Don't use devm_regmap_init_mmio_clk ASoC: fsl_spdif: Don't use devm_regmap_init_mmio_clk ASoC: fsl_asrc: Don't use devm_regmap_init_mmio_clk ASoC: fsl_easrc: Don't use devm_regmap_init_mmio_clk ASoC: fsl_audmix: Don't use devm_regmap_init_mmio_clk ASoC: fsl_micfil: Don't use devm_regmap_init_mmio_clk ASoC: soc-component: Add snd_soc_pcm_component_ack ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg audio device ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg ASoC: fsl-asoc-card: Add support for WM8958 codec ASoC: bindings: fsl-asoc-card: add compatible string for WM8958 codec ASoC: wm8960: Remove bitclk relax condition in wm8960_configure_sysclk ASoC: fsl_rpmsg: initialise pointers to NULL ASoC: dt-bindings: ak5558: Add compatible string for ak5552 ASoC: ak5558: Fix s/show/slow/ typo ASoC: ak5558: correct reset polarity ASoC: ak5558: change function name to ak5558_reset Shuming Fan (5): ASoC: rt1316: Add RT1316 SDCA vendor-specific driver ASoC: rt711-sdca: Add RT711 SDCA vendor-specific driver ASoC: rt5682: add delay time of workqueue to control next IRQ event ASoC: rt711-sdca: change capture switch controls ASoC: rt711-sdca: add the notification when volume changed Sia Jee Heng (1): ASoC: Intel: KMB: Fix random noise at the HDMI output Steve Lee (1): ASoC: max98390: Add support for tx slot configuration. Tang Bin (3): ASoC: codec: Omit superfluous error message in jz4760_codec_probe() ASoC: fsl_xcvr: Use devm_platform_ioremap_resource_byname() to simplify code ASoC: codecs/jz4770: Remove superfluous error message Thierry Reding (4): ASoC: tegra: Set driver name explicitly ASoC: simple-card-utils: Propagate errors on too many links ASoC: simple-card-utils: Increase maximum number of links to 128 ASoC: simple-card-utils: Allocate link info structure on heap Tian Tao (1): spi: imx: Use of_device_get_match_data() helper Tzung-Bi Shih (2): ASoC: dt-bindings: mt8183: add compatible string for using rt1015p ASoC: mediatek: mt8183: support machine driver with rt1015p Vamshi Krishna Gopal (2): ASoC: Intel: kbl: Add MST route change to kbl machine drivers ASoC: Intel: sof_sdw: add quirk for new ADL-P Rvp Vijendar Mukunda (5): ASoC: amd: Add support for RT5682 codec in machine driver ASoC: amd: update spdx license for acp machine driver ASoC: amd: fix acpi dependency kernel warning ASoC: amd: Add support for ALC1015P codec in acp3x machine driver ASoC: amd: drop S24_LE format support Viorel Suman (5): ASoC: fsl_xcvr: move reset assert into runtime_resume ASoC: fsl_spdif: use snd_ctl_boolean_mono_info ASoC: ak5558: Add support for ak5552 ASoC: ak4458: check reset control status ASoC: ak4458: enable daisy chain Wei Yongjun (4): ASoC: rt1316: Fix return value check in rt1316_sdw_probe() ASoC: rt715-sdca: Fix return value check in rt715_sdca_sdw_probe() ASoC: rt715-sdca: Remove unused including <linux/version.h> ASoC: rt1019: make symbol 'rt1019_i2c_driver' static Yang Li (3): ASoC: codecs: fix platform_no_drv_owner.cocci warnings ASoC: imx-hdmi: fix platform_no_drv_owner.cocci warnings ASoC: cs35l35: remove unused including <linux/version.h> Ye Bin (16): ASoC: codecs: lpass-va-macro: constify static struct snd_soc_dai_ops ASoC: codecs: wsa881x: constify static struct snd_soc_dai_ops ASoC: tas2764: constify static struct snd_soc_dai_ops ASoC: rt1019: constify static struct snd_soc_dai_ops ASoC: cx2072x: constify static struct snd_soc_dai_ops ASoC: tas2770: Constify static struct snd_soc_dai_ops ASoC: amd: raven: acp3x-i2s: Constify static struct snd_soc_dai_ops ASoC: codecs: lpass-rx-macro: constify static struct snd_soc_dai_ops ASoC: Intel: KMB: Constify static struct snd_soc_dai_ops ASoC: rt715-sdca: Constify static struct snd_soc_dai_ops ASoC: wcd934x: constify static struct snd_soc_dai_ops ASoC: rt711-sdca: Constify static struct snd_soc_dai_ops ASoC: codecs: lpass-wsa-macro: constify static struct snd_soc_dai_ops ASoC: amd: renoir: acp3x-pdm-dma: constify static struct snd_soc_dai_ops ASoC: codecs: lpass-tx-macro: constify static struct snd_soc_dai_ops ASoC: wcd9335: constify static struct snd_soc_dai_ops YueHaibing (1): ASoC: dwc: Fix -Wmissing-prototypes warnings Zhen Lei (1): ASoC: intel, keembay-i2s: Fix a dt_binding_check warning dingsenjie (1): ASoC: uniphier: Simplify the return expression of uniphier_aio_startup Documentation/devicetree/bindings/sound/ak5558.txt | 2 +- .../bindings/sound/audio-graph-port.yaml | 104 +- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 108 ++ .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 + .../bindings/sound/intel,keembay-i2s.yaml | 2 +- .../bindings/sound/marvell,mmp-sspa.yaml | 6 +- .../devicetree/bindings/sound/mchp,i2s-mcc.yaml | 108 ++ .../devicetree/bindings/sound/mchp-i2s-mcc.txt | 43 - .../sound/mt8183-mt6358-ts3a227-max98357.txt | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 11 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 11 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 27 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 11 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 11 +- .../devicetree/bindings/sound/renesas,rsnd.yaml | 5 +- .../devicetree/bindings/sound/rt1019.yaml | 35 + Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- .../bindings/sound/socionext,uniphier-aio.yaml | 8 +- .../bindings/sound/socionext,uniphier-evea.yaml | 8 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 21 +- MAINTAINERS | 3 +- drivers/extcon/Kconfig | 8 - drivers/extcon/Makefile | 1 - drivers/firmware/dmi_scan.c | 1 + drivers/mfd/arizona-core.c | 20 - drivers/reset/core.c | 215 +++ drivers/spi/spi-imx.c | 5 +- include/linux/platform_data/asoc-mx27vis.h | 12 - include/linux/reset.h | 315 ++++ include/sound/control.h | 35 +- include/sound/graph_card.h | 4 - include/sound/rt5645.h | 32 - include/sound/simple_card_utils.h | 128 +- include/sound/soc-acpi.h | 4 + include/sound/soc-component.h | 8 +- include/sound/soc-dpcm.h | 10 +- include/sound/soc.h | 36 +- include/sound/sof.h | 4 +- sound/core/Kconfig | 6 + sound/core/Makefile | 2 + sound/core/control.c | 182 +- sound/core/control_led.c | 770 ++++++++ sound/pci/hda/Kconfig | 4 +- sound/pci/hda/hda_codec.c | 69 +- sound/pci/hda/hda_generic.c | 162 +- sound/pci/hda/hda_generic.h | 15 +- sound/pci/hda/hda_local.h | 16 +- sound/pci/hda/patch_ca0132.c | 4 +- sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 6 +- sound/pci/hda/thinkpad_helper.c | 2 +- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 4 +- sound/soc/amd/Kconfig | 8 +- sound/soc/amd/acp-da7219-max98357a.c | 386 +++- sound/soc/amd/acp3x-rt5682-max9836.c | 46 +- sound/soc/amd/raven/acp3x-i2s.c | 8 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 4 +- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 +- sound/soc/atmel/Kconfig | 3 + sound/soc/atmel/atmel-classd.c | 6 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 161 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/Kconfig | 51 +- sound/soc/codecs/Makefile | 16 +- sound/soc/codecs/ab8500-codec.c | 7 - sound/soc/codecs/ad1836.c | 2 - sound/soc/codecs/adau17x1.c | 13 +- sound/soc/codecs/adau1977.c | 2 - sound/soc/codecs/ak4458.c | 66 +- sound/soc/codecs/ak4458.h | 1 + sound/soc/codecs/ak5558.c | 122 +- .../soc/codecs/arizona-jack.c | 577 +++--- sound/soc/codecs/arizona.h | 46 +- sound/soc/codecs/cros_ec_codec.c | 2 +- sound/soc/codecs/cs35l35.c | 3 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4270.c | 1 + sound/soc/codecs/cs42l42.c | 323 ++-- sound/soc/codecs/cs42l42.h | 28 +- sound/soc/codecs/cx2072x.c | 13 +- sound/soc/codecs/da7219-aad.c | 2 +- sound/soc/codecs/da7219.c | 5 +- sound/soc/codecs/da732x.c | 17 +- sound/soc/codecs/da732x.h | 12 +- sound/soc/codecs/hdac_hdmi.c | 14 +- sound/soc/codecs/hdac_hdmi.h | 2 +- sound/soc/codecs/hdmi-codec.c | 3 +- sound/soc/codecs/jz4760.c | 11 +- sound/soc/codecs/jz4770.c | 7 +- sound/soc/codecs/lm49453.c | 2 - sound/soc/codecs/lpass-rx-macro.c | 7 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 9 +- sound/soc/codecs/lpass-wsa-macro.c | 15 +- sound/soc/codecs/madera.h | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max98373.c | 14 + sound/soc/codecs/max98373.h | 3 + sound/soc/codecs/max98390.c | 62 + sound/soc/codecs/max98390.h | 2 + sound/soc/codecs/mt6358.c | 4 +- sound/soc/codecs/mt6359-accdet.c | 1080 ++++++++++++ sound/soc/codecs/mt6359-accdet.h | 128 ++ sound/soc/codecs/mt6359.c | 4 +- sound/soc/codecs/mt6359.h | 1864 ++++++++++++++++++-- sound/soc/codecs/nau8825.c | 11 +- sound/soc/codecs/pcm1681.c | 2 +- sound/soc/codecs/rt1011.c | 65 +- sound/soc/codecs/rt1011.h | 1 - sound/soc/codecs/rt1015.c | 27 +- sound/soc/codecs/rt1015.h | 10 + sound/soc/codecs/rt1015p.c | 13 +- sound/soc/codecs/rt1016.c | 11 +- sound/soc/codecs/rt1019.c | 608 +++++++ sound/soc/codecs/rt1019.h | 158 ++ sound/soc/codecs/rt1305.c | 6 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt1308.c | 10 +- sound/soc/codecs/rt1316-sdw.c | 744 ++++++++ sound/soc/codecs/rt1316-sdw.h | 55 + sound/soc/codecs/rt286.c | 34 +- sound/soc/codecs/rt298.c | 9 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/rt5640.c | 12 +- sound/soc/codecs/rt5645.c | 47 +- sound/soc/codecs/rt5645.h | 2 - sound/soc/codecs/rt5651.c | 6 +- sound/soc/codecs/rt5659.c | 4 +- sound/soc/codecs/rt5660.c | 4 +- sound/soc/codecs/rt5663.c | 4 +- sound/soc/codecs/rt5665.c | 4 +- sound/soc/codecs/rt5668.c | 8 +- sound/soc/codecs/rt5670.c | 55 +- sound/soc/codecs/rt5670.h | 2 + sound/soc/codecs/rt5677.c | 6 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 6 +- sound/soc/codecs/rt5682.c | 20 +- sound/soc/codecs/rt5682.h | 1 + sound/soc/codecs/rt700-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca-sdw.c | 425 +++++ sound/soc/codecs/rt711-sdca-sdw.h | 99 ++ sound/soc/codecs/rt711-sdca.c | 1583 +++++++++++++++++ sound/soc/codecs/rt711-sdca.h | 240 +++ sound/soc/codecs/rt711-sdw.c | 2 +- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715-sdca-sdw.c | 278 +++ sound/soc/codecs/rt715-sdca-sdw.h | 170 ++ sound/soc/codecs/rt715-sdca.c | 1075 +++++++++++ sound/soc/codecs/rt715-sdca.h | 136 ++ sound/soc/codecs/rt715-sdw.c | 2 +- sound/soc/codecs/rt715.c | 369 +++- sound/soc/codecs/rt715.h | 3 + sound/soc/codecs/sgtl5000.c | 6 +- sound/soc/codecs/sigmadsp-regmap.c | 2 +- sound/soc/codecs/sigmadsp.c | 18 +- sound/soc/codecs/sigmadsp.h | 2 +- sound/soc/codecs/sti-sas.c | 3 - sound/soc/codecs/tas2552.c | 4 +- sound/soc/codecs/tas2562.c | 3 +- sound/soc/codecs/tas2764.c | 2 +- sound/soc/codecs/tas2770.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 12 +- sound/soc/codecs/tlv320aic3x-i2c.c | 72 + sound/soc/codecs/tlv320aic3x-spi.c | 78 + sound/soc/codecs/tlv320aic3x.c | 111 +- sound/soc/codecs/tlv320aic3x.h | 13 + sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tscs454.c | 7 +- sound/soc/codecs/wcd-clsh-v2.h | 6 +- sound/soc/codecs/wcd9335.c | 4 +- sound/soc/codecs/wcd934x.c | 12 +- sound/soc/codecs/wm2200.c | 7 - sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 12 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8903.c | 2 - sound/soc/codecs/wm8958-dsp2.c | 16 +- sound/soc/codecs/wm8960.c | 12 +- sound/soc/codecs/wm8962.c | 23 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8994.h | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 14 +- sound/soc/codecs/wm8998.c | 9 + sound/soc/codecs/wm_adsp.c | 2 +- sound/soc/codecs/wm_hubs.h | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/dwc/local.h | 6 +- sound/soc/fsl/Kconfig | 32 + sound/soc/fsl/Makefile | 6 + sound/soc/fsl/fsl-asoc-card.c | 17 +- sound/soc/fsl/fsl_asrc.c | 59 +- sound/soc/fsl/fsl_asrc_dma.c | 5 +- sound/soc/fsl/fsl_aud2htx.c | 4 +- sound/soc/fsl/fsl_audmix.c | 3 +- sound/soc/fsl/fsl_dma.c | 3 - sound/soc/fsl/fsl_easrc.c | 13 +- sound/soc/fsl/fsl_esai.c | 50 +- sound/soc/fsl/fsl_micfil.c | 27 +- sound/soc/fsl/fsl_rpmsg.c | 279 +++ sound/soc/fsl/fsl_rpmsg.h | 35 + sound/soc/fsl/fsl_sai.c | 62 +- sound/soc/fsl/fsl_sai.h | 4 + sound/soc/fsl/fsl_spdif.c | 33 +- sound/soc/fsl/fsl_ssi.c | 7 +- sound/soc/fsl/fsl_xcvr.c | 19 +- sound/soc/fsl/imx-audio-rpmsg.c | 140 ++ sound/soc/fsl/imx-hdmi.c | 5 - sound/soc/fsl/imx-pcm-rpmsg.c | 918 ++++++++++ sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++ sound/soc/fsl/imx-rpmsg.c | 150 ++ sound/soc/fsl/mpc5200_dma.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/generic/audio-graph-card.c | 422 ++--- sound/soc/generic/simple-card-utils.c | 299 +++- sound/soc/generic/simple-card.c | 272 +-- sound/soc/intel/Makefile | 2 +- sound/soc/intel/atom/sst-mfld-dsp.h | 4 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 31 +- sound/soc/intel/boards/bytcr_wm5102.c | 36 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 77 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 58 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 48 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 14 +- sound/soc/intel/boards/sof_realtek_common.c | 105 ++ sound/soc/intel/boards/sof_realtek_common.h | 7 + sound/soc/intel/boards/sof_rt5682.c | 35 +- sound/soc/intel/boards/sof_sdw.c | 11 + sound/soc/intel/boards/sof_wm8804.c | 6 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 223 +++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 40 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 13 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 23 - sound/soc/intel/keembay/kmb_platform.c | 7 +- sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-topology.c | 17 +- sound/soc/intel/skylake/skl.h | 2 +- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 4 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 4 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 + sound/soc/meson/aiu-acodec-ctrl.c | 2 +- sound/soc/meson/aiu-codec-ctrl.c | 2 +- sound/soc/meson/aiu.c | 4 +- sound/soc/meson/aiu.h | 2 +- sound/soc/meson/axg-fifo.c | 11 +- sound/soc/meson/axg-frddr.c | 27 +- sound/soc/meson/axg-tdmin.c | 2 +- sound/soc/meson/axg-tdmout.c | 2 +- sound/soc/mxs/mxs-saif.c | 10 - sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 1 - sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass-hdmi.c | 4 - sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6afe-clocks.c | 209 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 4 +- sound/soc/qcom/qdsp6/q6afe.c | 9 +- sound/soc/qcom/qdsp6/q6afe.h | 4 +- sound/soc/qcom/qdsp6/q6asm.h | 6 +- sound/soc/samsung/i2s.c | 3 +- sound/soc/samsung/s3c24xx_simtec.c | 5 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/snow.c | 5 +- sound/soc/samsung/tm2_wm5110.c | 5 +- sound/soc/sh/dma-sh7760.c | 1 - sound/soc/sh/rcar/adg.c | 15 +- sound/soc/sh/rcar/cmd.c | 15 +- sound/soc/sh/rcar/core.c | 103 +- sound/soc/sh/rcar/ctu.c | 2 + sound/soc/sh/rcar/rsnd.h | 6 +- sound/soc/sh/rcar/src.c | 9 + sound/soc/sh/rcar/ssi.c | 267 ++- sound/soc/sh/rcar/ssiu.c | 22 +- sound/soc/sh/siu_pcm.c | 6 +- sound/soc/soc-acpi.c | 2 +- sound/soc/soc-component.c | 16 +- sound/soc/soc-compress.c | 15 +- sound/soc/soc-core.c | 55 +- sound/soc/soc-dai.c | 2 +- sound/soc/soc-dapm.c | 24 +- sound/soc/soc-ops.c | 2 +- sound/soc/soc-pcm.c | 527 +++--- sound/soc/soc-topology.c | 16 +- sound/soc/soc-utils.c | 6 + sound/soc/sof/Kconfig | 11 + sound/soc/sof/core.c | 14 +- sound/soc/sof/debug.c | 21 +- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/intel/bdw.c | 8 +- sound/soc/sof/intel/byt.c | 22 +- sound/soc/sof/intel/hda-dai.c | 44 + sound/soc/sof/intel/hda-dsp.c | 11 +- sound/soc/sof/intel/hda.c | 10 +- sound/soc/sof/intel/hda.h | 2 +- sound/soc/sof/intel/pci-tgl.c | 20 +- sound/soc/sof/intel/tgl.c | 2 +- sound/soc/sof/nocodec.c | 39 +- sound/soc/sof/ops.h | 6 +- sound/soc/sof/pcm.c | 38 +- sound/soc/sof/sof-acpi-dev.c | 18 +- sound/soc/sof/sof-audio.c | 63 +- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/sof-of-dev.c | 18 +- sound/soc/sof/sof-pci-dev.c | 20 +- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sof/topology.c | 250 ++- sound/soc/sti/sti_uniperif.c | 3 + sound/soc/sti/uniperif.h | 4 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 5 +- sound/soc/sunxi/sun4i-codec.c | 4 +- sound/soc/sunxi/sun4i-i2s.c | 2 - sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 21 + sound/soc/tegra/tegra20_ac97.h | 1 + sound/soc/tegra/tegra20_das.c | 8 +- sound/soc/tegra/tegra20_das.h | 6 +- sound/soc/tegra/tegra20_i2s.c | 64 +- sound/soc/tegra/tegra20_i2s.h | 1 + sound/soc/tegra/tegra20_spdif.c | 20 +- sound/soc/tegra/tegra30_ahub.c | 172 +- sound/soc/tegra/tegra30_ahub.h | 5 +- sound/soc/tegra/tegra30_i2s.c | 69 +- sound/soc/tegra/tegra_audio_graph_card.c | 5 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 2 +- sound/soc/ti/omap-mcbsp.c | 3 +- sound/soc/uniphier/aio-cpu.c | 7 +- sound/soc/ux500/mop500.c | 6 +- sound/soc/ux500/mop500_ab8500.h | 2 +- 344 files changed, 17633 insertions(+), 3662 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml create mode 100644 Documentation/devicetree/bindings/sound/mchp,i2s-mcc.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mchp-i2s-mcc.txt create mode 100644 Documentation/devicetree/bindings/sound/rt1019.yaml delete mode 100644 include/linux/platform_data/asoc-mx27vis.h delete mode 100644 include/sound/rt5645.h create mode 100644 sound/core/control_led.c rename drivers/extcon/extcon-arizona.c => sound/soc/codecs/arizona-jack.c (76%) create mode 100644 sound/soc/codecs/mt6359-accdet.c create mode 100644 sound/soc/codecs/mt6359-accdet.h create mode 100644 sound/soc/codecs/rt1019.c create mode 100644 sound/soc/codecs/rt1019.h create mode 100644 sound/soc/codecs/rt1316-sdw.c create mode 100644 sound/soc/codecs/rt1316-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca-sdw.c create mode 100644 sound/soc/codecs/rt711-sdca-sdw.h create mode 100644 sound/soc/codecs/rt711-sdca.c create mode 100644 sound/soc/codecs/rt711-sdca.h create mode 100644 sound/soc/codecs/rt715-sdca-sdw.c create mode 100644 sound/soc/codecs/rt715-sdca-sdw.h create mode 100644 sound/soc/codecs/rt715-sdca.c create mode 100644 sound/soc/codecs/rt715-sdca.h create mode 100644 sound/soc/codecs/tlv320aic3x-i2c.c create mode 100644 sound/soc/codecs/tlv320aic3x-spi.c create mode 100644 sound/soc/fsl/fsl_rpmsg.c create mode 100644 sound/soc/fsl/fsl_rpmsg.h create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h create mode 100644 sound/soc/fsl/imx-rpmsg.c

2 1

[PATCH 00/14] ASoC: Intel/SOF: extend run-time driver selection to ACPI devices
by Pierre-Louis Bossart 26 Apr '21

26 Apr '21

The module snd-intel-dspcfg, suggested by Jaroslav last year, currently provide the means to select a PCI driver at run-time, based on quirks, recommendations or user selection via a kernel parameter. This capability removed a lot of confusions in distributions and removed the need for recompilations to select legacy HDaudio, SST or SOF drivers. This patchset extends the concept to ACPI devices. This was driven by the desire to at some point deprecate the Atom/SST driver for Baytrail and Cherrytrail, which is no longer maintained by Intel. By having the SOF driver enabled by distributions for Baytrail/Cherrytrail, we can enable more end-user tests and make the transition easier for distributions (likely in 2021 at this point). This patchset provides the same solution for Broadwell, mainly to have a single build for all Intel platforms. SOF on Broadwell remains an option not recommended for distributions, as long as the 'catpt' driver is maintained there is no burning desire to make SOF the default on the three Broadwell-based platforms with the DSP enabled. Pierre-Louis Bossart (14): ASoC: Intel: broadwell: add missing pm_ops ASoC: Intel: bdw-rt5677: add missing pm_ops ALSA: hda: intel-dsp-config: add helper for ACPI DSP driver selection ASoC: soc-acpi: add helper to identify parent driver. ASoC: Intel: boards: byt/cht: set card and driver name at run time ASoC: Intel: byt/cht: set pm ops dynamically ASoC: SOF: acpi: add dynamic selection of DSP driver ASoC: Intel: Atom: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and Atom/SST drivers ALSA: hda: intel-dsp-config: add Broadwell ACPI DSP driver selection ASoC: Intel: broadwell: set card and driver name dynamically ASoC: Intel: catpt: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and catpt drivers ALSA: hda: intel-dsp-config: ignore dsp_driver parameter for PCI legacy devices include/sound/intel-dsp-config.h | 7 ++ include/sound/soc-acpi.h | 6 + sound/hda/intel-dsp-config.c | 111 +++++++++++++++++++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/atom/sst/sst_acpi.c | 8 ++ sound/soc/intel/boards/bdw-rt5650.c | 17 ++- sound/soc/intel/boards/bdw-rt5677.c | 18 ++- sound/soc/intel/boards/broadwell.c | 20 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 27 +++-- sound/soc/intel/boards/bytcht_da7213.c | 27 +++-- sound/soc/intel/boards/bytcht_es8316.c | 29 +++-- sound/soc/intel/boards/bytcr_rt5640.c | 30 +++-- sound/soc/intel/boards/bytcr_rt5651.c | 27 +++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 29 +++-- sound/soc/intel/boards/cht_bsw_nau8824.c | 29 +++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 38 ++++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 29 +++-- sound/soc/intel/catpt/device.c | 12 ++ sound/soc/sof/intel/Kconfig | 33 +++--- sound/soc/sof/sof-acpi-dev.c | 14 ++- 20 files changed, 392 insertions(+), 121 deletions(-) -- 2.25.1

6 54

[PATCH] sound/isa/sb/emu8000: Fix a use after free in snd_emu8000_create_mixer
by Lv Yunlong 26 Apr '21

26 Apr '21

Our code analyzer reported a uaf. In snd_emu8000_create_mixer, the callee snd_ctl_add(..,emu->controls[i]) calls snd_ctl_add_replace(.., kcontrol,..). Inside snd_ctl_add_replace(), if error happens, kcontrol will be freed by snd_ctl_free_one(kcontrol). Then emu->controls[i] points to a freed memory, and the execution comes to __error branch of snd_emu8000_create_mixer. The freed emu->controls[i] is used in snd_ctl_remove(card, emu->controls[i]). My patch set emu->controls[i] to NULL if snd_ctl_add() failed to avoid the uaf. Signed-off-by: Lv Yunlong <lyl2019(a)mail.ustc.edu.cn> --- sound/isa/sb/emu8000.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 0aa545ac6e60..1c90421a88dc 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -1029,8 +1029,10 @@ snd_emu8000_create_mixer(struct snd_card *card, struct snd_emu8000 *emu) memset(emu->controls, 0, sizeof(emu->controls)); for (i = 0; i < EMU8000_NUM_CONTROLS; i++) { - if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) + if ((err = snd_ctl_add(card, emu->controls[i] = snd_ctl_new1(mixer_defs[i], emu))) < 0) { + emu->controls[i] = NULL; goto __error; + } } return 0; -- 2.25.1

3 2

[PATCH] ALSA: hda/realtek: fix static noise on ALC285 Lenovo laptops
by Sami Loone 26 Apr '21

26 Apr '21

Remove a duplicate vendor+subvendor pin fixup entry as one is masking the other and making it unreachable. Consider the more specific newcomer as a second chance instead. The generic entry is made less strict to also match for laptops with slightly different 0x12 pin configuration. Tested on Lenovo Yoga 6 (AMD) where 0x12 is 0x40000000. Fixes: 607184cb1635 ("ALSA: hda/realtek - Add supported for more Lenovo ALC285 Headset Button") Signed-off-by: Sami Loone <sami(a)loone.fi> --- sound/pci/hda/patch_realtek.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..98a53c94ca48 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8733,12 +8733,7 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x12, 0x90a60130}, {0x19, 0x03a11020}, {0x21, 0x0321101f}), - SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, - {0x14, 0x90170110}, - {0x19, 0x04a11040}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE, - {0x12, 0x90a60130}, {0x14, 0x90170110}, {0x19, 0x04a11040}, {0x21, 0x04211020}), @@ -8909,6 +8904,10 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0285, 0x17aa, "Lenovo", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK, + {0x14, 0x90170110}, + {0x19, 0x04a11040}, + {0x21, 0x04211020}), {} }; -- 2.31.1

2 1

[PATCH] firmware: replace HOTPLUG with UEVENT in FW_ACTION defines
by Shawn Guo 26 Apr '21

26 Apr '21

With commit 312c004d36ce ("[PATCH] driver core: replace "hotplug" by "uevent"") already in the tree over a decade, update the name of FW_ACTION defines to follow semantics, and reflect what the defines are really meant for, i.e. whether or not generate user space event. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- drivers/dma/imx-sdma.c | 2 +- drivers/media/platform/exynos4-is/fimc-is.c | 2 +- drivers/mfd/iqs62x.c | 2 +- drivers/misc/lattice-ecp3-config.c | 2 +- drivers/net/wireless/ti/wlcore/main.c | 2 +- drivers/platform/x86/dell/dell_rbu.c | 2 +- drivers/remoteproc/remoteproc_core.c | 2 +- drivers/scsi/lpfc/lpfc_init.c | 2 +- drivers/tty/serial/ucc_uart.c | 2 +- include/linux/firmware.h | 4 ++-- lib/test_firmware.c | 10 +++++----- sound/soc/codecs/wm8958-dsp2.c | 6 +++--- 12 files changed, 19 insertions(+), 19 deletions(-) diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index d5590c08db51..e2b559945c11 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -1829,7 +1829,7 @@ static int sdma_get_firmware(struct sdma_engine *sdma, int ret; ret = request_firmware_nowait(THIS_MODULE, - FW_ACTION_HOTPLUG, fw_name, sdma->dev, + FW_ACTION_UEVENT, fw_name, sdma->dev, GFP_KERNEL, sdma, sdma_load_firmware); return ret; diff --git a/drivers/media/platform/exynos4-is/fimc-is.c b/drivers/media/platform/exynos4-is/fimc-is.c index 972d9601d236..8f3ba74f19f2 100644 --- a/drivers/media/platform/exynos4-is/fimc-is.c +++ b/drivers/media/platform/exynos4-is/fimc-is.c @@ -436,7 +436,7 @@ static void fimc_is_load_firmware(const struct firmware *fw, void *context) static int fimc_is_request_firmware(struct fimc_is *is, const char *fw_name) { return request_firmware_nowait(THIS_MODULE, - FW_ACTION_HOTPLUG, fw_name, &is->pdev->dev, + FW_ACTION_UEVENT, fw_name, &is->pdev->dev, GFP_KERNEL, is, fimc_is_load_firmware); } diff --git a/drivers/mfd/iqs62x.c b/drivers/mfd/iqs62x.c index d1fc38a78acb..9805cf191245 100644 --- a/drivers/mfd/iqs62x.c +++ b/drivers/mfd/iqs62x.c @@ -998,7 +998,7 @@ static int iqs62x_probe(struct i2c_client *client) device_property_read_string(&client->dev, "firmware-name", &fw_name); - ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, fw_name ? : iqs62x->dev_desc->fw_name, &client->dev, GFP_KERNEL, iqs62x, iqs62x_firmware_load); diff --git a/drivers/misc/lattice-ecp3-config.c b/drivers/misc/lattice-ecp3-config.c index 5eaf74447ca1..0f54730c7ed5 100644 --- a/drivers/misc/lattice-ecp3-config.c +++ b/drivers/misc/lattice-ecp3-config.c @@ -198,7 +198,7 @@ static int lattice_ecp3_probe(struct spi_device *spi) spi_set_drvdata(spi, data); init_completion(&data->fw_loaded); - err = request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + err = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, FIRMWARE_NAME, &spi->dev, GFP_KERNEL, spi, firmware_load); if (err) { diff --git a/drivers/net/wireless/ti/wlcore/main.c b/drivers/net/wireless/ti/wlcore/main.c index 8509b989940c..5bf15355c2b3 100644 --- a/drivers/net/wireless/ti/wlcore/main.c +++ b/drivers/net/wireless/ti/wlcore/main.c @@ -6784,7 +6784,7 @@ int wlcore_probe(struct wl1271 *wl, struct platform_device *pdev) if (pdev_data->family && pdev_data->family->nvs_name) { nvs_name = pdev_data->family->nvs_name; - ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, nvs_name, &pdev->dev, GFP_KERNEL, wl, wlcore_nvs_cb); if (ret < 0) { diff --git a/drivers/platform/x86/dell/dell_rbu.c b/drivers/platform/x86/dell/dell_rbu.c index 03c3ff34bcf5..8758d717ae5c 100644 --- a/drivers/platform/x86/dell/dell_rbu.c +++ b/drivers/platform/x86/dell/dell_rbu.c @@ -573,7 +573,7 @@ static ssize_t image_type_write(struct file *filp, struct kobject *kobj, if (!rbu_data.entry_created) { spin_unlock(&rbu_data.lock); req_firm_rc = request_firmware_nowait(THIS_MODULE, - FW_ACTION_NOHOTPLUG, "dell_rbu", + FW_ACTION_NOUEVENT, "dell_rbu", &rbu_device->dev, GFP_KERNEL, &context, callbackfn_rbu); if (req_firm_rc) { diff --git a/drivers/remoteproc/remoteproc_core.c b/drivers/remoteproc/remoteproc_core.c index ab150765d124..c3e01e95c0d2 100644 --- a/drivers/remoteproc/remoteproc_core.c +++ b/drivers/remoteproc/remoteproc_core.c @@ -1628,7 +1628,7 @@ static int rproc_trigger_auto_boot(struct rproc *rproc) * We're initiating an asynchronous firmware loading, so we can * be built-in kernel code, without hanging the boot process. */ - ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, rproc->firmware, &rproc->dev, GFP_KERNEL, rproc, rproc_auto_boot_callback); if (ret < 0) diff --git a/drivers/scsi/lpfc/lpfc_init.c b/drivers/scsi/lpfc/lpfc_init.c index 71f340dd4fbd..53546b079553 100644 --- a/drivers/scsi/lpfc/lpfc_init.c +++ b/drivers/scsi/lpfc/lpfc_init.c @@ -13159,7 +13159,7 @@ lpfc_sli4_request_firmware_update(struct lpfc_hba *phba, uint8_t fw_upgrade) snprintf(file_name, ELX_MODEL_NAME_SIZE, "%s.grp", phba->ModelName); if (fw_upgrade == INT_FW_UPGRADE) { - ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, file_name, &phba->pcidev->dev, GFP_KERNEL, (void *)phba, lpfc_write_firmware); diff --git a/drivers/tty/serial/ucc_uart.c b/drivers/tty/serial/ucc_uart.c index d6a8604157ab..e651c9aafdd0 100644 --- a/drivers/tty/serial/ucc_uart.c +++ b/drivers/tty/serial/ucc_uart.c @@ -1227,7 +1227,7 @@ static int soft_uart_init(struct platform_device *ofdev) * kernel, then we use it. */ ret = request_firmware_nowait(THIS_MODULE, - FW_ACTION_HOTPLUG, filename, &ofdev->dev, + FW_ACTION_UEVENT, filename, &ofdev->dev, GFP_KERNEL, &ofdev->dev, uart_firmware_cont); if (ret) { dev_err(&ofdev->dev, diff --git a/include/linux/firmware.h b/include/linux/firmware.h index 84e346ae766e..25109192cebe 100644 --- a/include/linux/firmware.h +++ b/include/linux/firmware.h @@ -6,8 +6,8 @@ #include <linux/compiler.h> #include <linux/gfp.h> -#define FW_ACTION_NOHOTPLUG 0 -#define FW_ACTION_HOTPLUG 1 +#define FW_ACTION_NOUEVENT 0 +#define FW_ACTION_UEVENT 1 struct firmware { size_t size; diff --git a/lib/test_firmware.c b/lib/test_firmware.c index b6fe89add9fe..1bccd6cd5f48 100644 --- a/lib/test_firmware.c +++ b/lib/test_firmware.c @@ -260,8 +260,8 @@ static ssize_t config_show(struct device *dev, len += scnprintf(buf + len, PAGE_SIZE - len, "send_uevent:\t\t%s\n", test_fw_config->send_uevent ? - "FW_ACTION_HOTPLUG" : - "FW_ACTION_NOHOTPLUG"); + "FW_ACTION_UEVENT" : + "FW_ACTION_NOUEVENT"); len += scnprintf(buf + len, PAGE_SIZE - len, "into_buf:\t\t%s\n", test_fw_config->into_buf ? "true" : "false"); @@ -729,7 +729,7 @@ static ssize_t trigger_custom_fallback_store(struct device *dev, mutex_lock(&test_fw_mutex); release_firmware(test_firmware); test_firmware = NULL; - rc = request_firmware_nowait(THIS_MODULE, FW_ACTION_NOHOTPLUG, name, + rc = request_firmware_nowait(THIS_MODULE, FW_ACTION_NOUEVENT, name, dev, GFP_KERNEL, NULL, trigger_async_request_cb); if (rc) { @@ -938,8 +938,8 @@ ssize_t trigger_batched_requests_async_store(struct device *dev, pr_info("batched loading '%s' custom fallback mechanism %u times\n", test_fw_config->name, test_fw_config->num_requests); - send_uevent = test_fw_config->send_uevent ? FW_ACTION_HOTPLUG : - FW_ACTION_NOHOTPLUG; + send_uevent = test_fw_config->send_uevent ? FW_ACTION_UEVENT : + FW_ACTION_NOUEVENT; for (i = 0; i < test_fw_config->num_requests; i++) { req = &test_fw_config->reqs[i]; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 3bce9a14f0f3..a5ea1645993b 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -912,13 +912,13 @@ void wm8958_dsp2_init(struct snd_soc_component *component) /* We don't *require* firmware and don't want to delay boot */ - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "wm8958_mbc.wfw", component->dev, GFP_KERNEL, component, wm8958_mbc_loaded); - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "wm8958_mbc_vss.wfw", component->dev, GFP_KERNEL, component, wm8958_mbc_vss_loaded); - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "wm8958_enh_eq.wfw", component->dev, GFP_KERNEL, component, wm8958_enh_eq_loaded); -- 2.17.1

3 3

[PATCH] ALSA: usb-audio: Fix implicit sync clearance at stopping stream
by Takashi Iwai 26 Apr '21

26 Apr '21

The recent endpoint management change for implicit feedback mode added a clearance of ep->sync_sink (formerly ep->sync_slave) pointer at snd_usb_endpoint_stop() to assure no leftover for the feedback from the already stopped capture stream. This turned out to cause a regression, however, when full-duplex streams were running and only a capture was stopped. Because of the above clearance of ep->sync_sink pointer, no more feedback is done, hence the playback will stall. This patch fixes the ep->sync_sink clearance to be done only after all endpoints are released, for addressing the regression. Reported-and-tested-by: Lucas Endres <jaffa225man(a)gmail.com> Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index f4c3d2b38abb..014c43862826 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1443,11 +1443,11 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) if (snd_BUG_ON(!atomic_read(&ep->running))) return; - if (ep->sync_source) - WRITE_ONCE(ep->sync_source->sync_sink, NULL); - - if (!atomic_dec_return(&ep->running)) + if (!atomic_dec_return(&ep->running)) { + if (ep->sync_source) + WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false); + } } /** -- 2.26.2

1 0

[PATCH v1 0/2] Fix couple of bugs reported by Canonical
by Vitaly Rodionov 25 Apr '21

25 Apr '21

CirrusLogic: Cracking noises appears in built-in speaker when output volume is set >80% BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1923557i CirrusLogic: The default input volume is "0%" BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1924997 Stefan Binding (2): ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to 16% ALSA: hda/cirrus: Use CS8409 Equalizer to fix abnormal sounds on Bullseye sound/pci/hda/patch_cirrus.c | 37 ++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) -- 2.25.1

2 4

ASoc / PCM recording-related regression between v5.4 and v5.5
by Hin-Tak Leung 24 Apr '21

24 Apr '21

Hi, I am trying to find the cause of a rather unusual regression between kernel v5.4 and v5.5, for an out-of-tree driver on the raspberrypi. Everything looks identical between v5.4 and 5.5, within accountable differences, even when I enable dynamic debugging and echo "file sound/* +p" > /sys/kernel/debug/dynamic_debug/control for the whole of sound sub-system. It records fines against v5.4, but blocks on v5.5, and on ctrl-c, does ^CAborted by signal Interrupt... arecord: pcm_read:2145: read error: Interrupted system call I have gradually replaced the include/sound and sound/ directories between the two (the pi v5.4 and v5.5 kernels are separate branches off linux-stable, so one cannot do ordinary bisects), and narrowed it down to the following 13 commits from 5 people: 4104faaeeda0e23a169c50e43f309ff7435087b1 ASoC: soc-core: disable route checks for legacy devices 4bf2e385aa59c2fae5f880aa25cfd2b470109093 ASoC: core: Init pcm runtime work early to avoid warnings fb5126778333d289b2623a7e6260beeb1ac1b819 ASoC: core: add SND_SOC_BYTES_E dc73d73aa7145f55412611f3eead1e85ae026785 ASoC: add control components management 509ba54fcfd1e45bceebe8ccea59dc496312f1a0 ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY e443c20593de9f8efd9b2935ed40eb0bbacce30b ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() 5d07519703bc2f0bf19d33652401552a480d68b8 ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() e11381f38f34789b374880c4a149e25e8d7f0cfd ASoC: soc-core: add snd_soc_unregister_dai() 4baabbf932ed4f97df8e18cf546d39b7c2138020 ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro 2b544dd7b43b19fb55ea4fbb3e30b60eb20b7828 ASoC: soc-core: add for_each_rtd_components() and replace 33536a14879515949b065721cdb7fedb276d8e8a ASoC: soc-core: remove for_each_rtdcom_safe() 8ec241c495dde3d19a0459304298c2468c60182b ASoC: soc-core: add snd_soc_pcm_lib_ioctl() b7c5bc45ee94a03a0dc45a862180e17db8ea8e9d ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() I think it is overlay / devicetree related, since another audio device of the same family and similar hardware but different overlay works fine. As I already did 'echo "file sound/* +p" > /sys/kernel/debug/dynamic_debug/control' , I likely need some additional debug statement or enable debugging at another part to go further, despite it being a regression in 'include/sound/soc*.h' and 'sound/soc/soc-*'. Building the pi kernel cleanly - from 'make mrproper', copy .config then 'make oldconnfig' - takes about an hour, so switching between the two and compare takes about 3 hours, and quite slow. I can possibly narrow the changes further from the 13 commits (9 of them from Kuninori Morimoto), but I think some more dev_dbg() is needed as the current ones in "sound/*" don't help; and it is worth bring it to alsa-devel since it is so unusual - all the dev_dbg() in kernel/sound/ and the driver side emits to the same messages, v5.4 can records while v5.5 blocks on stuck system call. More details at https://github.com/raspberrypi/linux/issues/4279 and it was initially at https://github.com/respeaker/seeed-voicecard/issues/290 . The "seeed-8mic-voicecard-overlay.dts" is the dts corresponds to the problematic device ; The one with similar hardware but okay is "seeed-4mic-voicecard-overlay.dts" . I am subscribed to alsa-devel actually, but receive it in digest form, and haven't opened the digests for a while... Thanks a lot. Hin-Tak

3 10

Warnings on startup
by GitHub issues - opened 23 Apr '21

23 Apr '21

alsa-project/alsa-plugins issue #20 was opened from bramtayl: I've been trying to get Julia's PortAudio to load alsa-plugins (it's been a process). I thought I might as well see if I can cut down on the warnings on start up (`using PortAudio` from Julia) and some of them I've been able to knock out but some of them not so much: These ones ``` ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ``` I got rid of by commenting out lines in my alsa.conf file: ``` # pcm.rear cards.pcm.rear # pcm.center_lfe cards.pcm.center_lfe # pcm.side cards.pcm.side ``` This one ``` ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ``` I got rid of by commening out: ``` # pcm.surround21 cards.pcm.surround21 # pcm.surround40 cards.pcm.surround40 # pcm.surround41 cards.pcm.surround41 # pcm.surround50 cards.pcm.surround50 # pcm.surround51 cards.pcm.surround51 # pcm.surround71 cards.pcm.surround71 ``` I'm not sure if these work arounds are the best solutions...shouldn't alsa be able to tell that my laptop doesn't have surround sound? This one is expected, I haven't yet figured out how to build jack on Julia's BinaryBuilder: ``` ALSA lib dlmisc.c:339:(snd_dlobj_cache_get0) Cannot open shared library libasound_module_pcm_jack.so (/home/brandon/.julia/artifacts/d587f48a6815c72dec4b083bf297caec7a8e3f4b/lib/alsa-lib/libasound_module_pcm_jack.so: cannot open shared object file: No such file or directory) ``` These ones are a bit confusing; ``` ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ``` Is it possible that `oss/50-oss.conf` and `usb_stream/98-usb-stream.conf` need to get updated? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/20 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2] ASoC: rt711-sdca: add the notification when volume changed
by shumingf＠realtek.com 23 Apr '21

23 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> This patch adds the return value when the volume settings were changed. The userspace application might monitor the kcontrols to check which control changed. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 39848d2641c3..cc36739f7fcf 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -509,12 +509,16 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); unsigned int read_l, read_r, gain_l_val, gain_r_val; - unsigned int i, adc_vol_flag = 0; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; if (strstr(ucontrol->id.name, "FU1E Capture Volume") || strstr(ucontrol->id.name, "FU0F Capture Volume")) adc_vol_flag = 1; + regmap_read(rt711->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt711->mbq_regmap, mc->rreg, &rvalue); + /* control value to 2's complement value */ /* L Channel */ gain_l_val = ucontrol->value.integer.value[0]; @@ -560,6 +564,11 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, gain_r_val &= 0xffff; } + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + else + return 0; + for (i = 0; i < 3; i++) { /* retry 3 times at most */ /* Lch*/ regmap_write(rt711->mbq_regmap, mc->reg, gain_l_val); @@ -573,7 +582,7 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, break; } - return i == 3 ? -EIO : 0; + return i == 3 ? -EIO : changed; } static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, -- 2.29.0

2 1

[PATCH v4] ASoC: rt711-sdca: change capture switch controls
by shumingf＠realtek.com 23 Apr '21

23 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The DAPM event and mixer control could mute/unmute the capture directly. That will be confused that capture still works if the user settings is unmute before the capture. Therefore, this patch uses the variables to record the capture switch status of DAPM and mixer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 168 +++++++++++++++++++++++++--------- sound/soc/codecs/rt711-sdca.h | 2 + 2 files changed, 127 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..39848d2641c3 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -642,6 +642,114 @@ static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt711_sdca_set_fu0f_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt711->fu0f_dapm_mute || rt711->fu0f_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt711->fu0f_dapm_mute || rt711->fu0f_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt711_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu1e_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu1e_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu1e_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu1e_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu1e_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu1e_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu1e_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + +static int rt711_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu0f_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu0f_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -652,14 +760,10 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), + SOC_DOUBLE_EXT("FU1E Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu1e_capture_get, rt711_sdca_fu1e_capture_put), + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu0f_capture_get, rt711_sdca_fu0f_capture_put), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), @@ -809,28 +913,15 @@ static int rt711_sdca_fu0f_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu0f_dapm_mute = false; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); + rt711->fu0f_dapm_mute = true; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; } return 0; @@ -842,29 +933,16 @@ static int rt711_sdca_fu1e_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu1e_dapm_mute = false; + rt711_sdca_set_fu1e_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); - break; + rt711->fu1e_dapm_mute = true; + rt711_sdca_set_fu1e_capture_ctl(rt711); + break; } return 0; } @@ -1330,6 +1408,10 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, */ rt711->hw_init = false; rt711->first_hw_init = false; + rt711->fu0f_dapm_mute = true; + rt711->fu1e_dapm_mute = true; + rt711->fu0f_mixer_l_mute = rt711->fu0f_mixer_r_mute = true; + rt711->fu1e_mixer_l_mute = rt711->fu1e_mixer_r_mute = true; /* JD source uses JD2 in default */ rt711->jd_src = RT711_JD2; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 98a022cec0bd..43ae82b7fdb3 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -30,6 +30,8 @@ struct rt711_sdca_priv { int jack_type, jd_src; unsigned int scp_sdca_stat1, scp_sdca_stat2; int hw_ver; + bool fu0f_dapm_mute, fu0f_mixer_l_mute, fu0f_mixer_r_mute; + bool fu1e_dapm_mute, fu1e_mixer_l_mute, fu1e_mixer_r_mute; }; struct sdw_stream_data { -- 2.29.0

2 1

[PATCH] ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2
by Mark Brown 23 Apr '21

23 Apr '21

A KernelCI bisection identified 59c35c44a9cf89 "ASoC: simple-card: add simple_parse_node()" as causing simple-card to fail to instantiate on kontron-sl28-var3-ads2 systems. Since the merge window is expected to open over the weekend drop that commit and subsequent ones which depend on it for now in case other systems are affected too. The boot log showed the error as: <4>[ 9.948821] sysfs: cannot create duplicate filename '/devices/platform/sound/(null)-wm8904-hifi' (backtrace) <3>[ 10.191982] kobject_add_internal failed for (null)-wm8904-hifi with -EEXIST, don't try to register things with the same name in the same directory. The dropped commits are: 73371bacf0475a20ab6 "ASoC: audio-graph: tidyup graph_dai_link_of_dpcm()" 434392271afcff350fe "ASoC: simple-card: add simple_link_init()" 59c35c44a9cf89a83a9 "ASoC: simple-card: add simple_parse_node()" Reported-by: Guillaume Tucker <guillaume.tucker(a)collabora.com> Reported-by: "kernelci.org bot" <bot(a)kernelci.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 30 ++--- sound/soc/generic/simple-card.c | 178 ++++++++++++++------------- 2 files changed, 106 insertions(+), 102 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0159a4576e9c..2c8a2fcb7922 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -276,19 +276,24 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct device_node *ep = li->cpu ? cpu_ep : codec_ep; + struct device_node *port; + struct device_node *ports; + struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); char dai_name[64]; int ret; + port = of_get_parent(ep); + ports = of_get_parent(port); + dev_dbg(dev, "link_of DPCM (%pOF)\n", ep); if (li->cpu) { - struct snd_soc_card *card = simple_priv_to_card(priv); - struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); - /* Codec is dummy */ /* FE settings */ @@ -297,7 +302,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, ret = graph_parse_node(priv, cpu_ep, li, 1); if (ret) - return ret; + goto out_put_node; snprintf(dai_name, sizeof(dai_name), "fe.%pOFP.%s", cpus->of_node, cpus->dai_name); @@ -314,10 +319,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (card->component_chaining && !soc_component_is_pcm(cpus)) dai_link->no_pcm = 1; } else { - struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); - struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); - struct device_node *port; - struct device_node *ports; + struct snd_soc_codec_conf *cconf; /* CPU is dummy */ @@ -325,25 +327,22 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; + cconf = simple_props_to_codec_conf(dai_props, 0); + ret = graph_parse_node(priv, codec_ep, li, 0); if (ret < 0) - return ret; + goto out_put_node; snprintf(dai_name, sizeof(dai_name), "be.%pOFP.%s", codecs->of_node, codecs->dai_name); /* check "prefix" from top node */ - port = of_get_parent(ep); - ports = of_get_parent(port); snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, "prefix"); if (of_node_name_eq(ports, "ports")) snd_soc_of_parse_node_prefix(ports, cconf, codecs->of_node, "prefix"); snd_soc_of_parse_node_prefix(port, cconf, codecs->of_node, "prefix"); - - of_node_put(ports); - of_node_put(port); } graph_parse_convert(dev, ep, &dai_props->adata); @@ -352,8 +351,11 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); +out_put_node: li->link++; + of_node_put(ports); + of_node_put(port); return ret; } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6a04632944b8..a1373be4558f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -93,11 +93,12 @@ static void simple_parse_convert(struct device *dev, } static void simple_parse_mclk_fs(struct device_node *top, - struct device_node *np, + struct device_node *cpu, + struct device_node *codec, struct simple_dai_props *props, char *prefix) { - struct device_node *node = of_get_parent(np); + struct device_node *node = of_get_parent(cpu); char prop[128]; snprintf(prop, sizeof(prop), "%smclk-fs", PREFIX); @@ -105,79 +106,12 @@ static void simple_parse_mclk_fs(struct device_node *top, snprintf(prop, sizeof(prop), "%smclk-fs", prefix); of_property_read_u32(node, prop, &props->mclk_fs); - of_property_read_u32(np, prop, &props->mclk_fs); + of_property_read_u32(cpu, prop, &props->mclk_fs); + of_property_read_u32(codec, prop, &props->mclk_fs); of_node_put(node); } -static int simple_parse_node(struct asoc_simple_priv *priv, - struct device_node *np, - struct link_info *li, - char *prefix, - int is_cpu) -{ - struct device *dev = simple_priv_to_dev(priv); - struct device_node *top = dev->of_node; - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); - struct snd_soc_dai_link_component *dlc; - struct asoc_simple_dai *dai; - int ret, single = 0; - - if (is_cpu) { - dlc = asoc_link_to_cpu(dai_link, 0); - dai = simple_props_to_dai_cpu(dai_props, 0); - } else { - dlc = asoc_link_to_codec(dai_link, 0); - dai = simple_props_to_dai_codec(dai_props, 0); - } - - simple_parse_mclk_fs(top, np, dai_props, prefix); - - ret = asoc_simple_parse_dai(np, dlc, &single); - if (ret) - return ret; - - ret = asoc_simple_parse_clk(dev, np, dai, dlc); - if (ret) - return ret; - - ret = asoc_simple_parse_tdm(np, dai); - if (ret) - return ret; - - if (is_cpu) { - struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); - - asoc_simple_canonicalize_cpu(dlc, single); - asoc_simple_canonicalize_platform(platforms, dlc); - } - - return 0; -} - -static int simple_link_init(struct asoc_simple_priv *priv, - struct device_node *node, - struct device_node *codec, - struct link_info *li, - char *prefix, - char *name) -{ - struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - int ret; - - ret = asoc_simple_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - return 0; - - dai_link->init = asoc_simple_dai_init; - dai_link->ops = &simple_ops; - - return asoc_simple_set_dailink_name(dev, dai_link, name); -} - static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, @@ -187,32 +121,50 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); + struct asoc_simple_dai *dai; struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); char *prefix = ""; - char dai_name[64]; int ret; dev_dbg(dev, "link_of DPCM (%pOF)\n", np); + li->link++; + /* For single DAI link & old style of DT node */ if (is_top) prefix = PREFIX; if (li->cpu) { + int is_single_links = 0; + /* Codec is dummy */ /* FE settings */ dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = simple_parse_node(priv, np, li, prefix, 1); + dai = simple_props_to_dai_cpu(dai_props, 0); + + ret = asoc_simple_parse_dai(np, cpus, &is_single_links); + if (ret) + goto out_put_node; + + ret = asoc_simple_parse_clk(dev, np, dai, cpus); if (ret < 0) goto out_put_node; - snprintf(dai_name, sizeof(dai_name), "fe.%s", cpus->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + cpus->dai_name); + if (ret < 0) + goto out_put_node; + + asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf; @@ -222,13 +174,22 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; + dai = simple_props_to_dai_codec(dai_props, 0); cconf = simple_props_to_codec_conf(dai_props, 0); - ret = simple_parse_node(priv, np, li, prefix, 0); + ret = asoc_simple_parse_dai(np, codecs, NULL); + if (ret < 0) + goto out_put_node; + + ret = asoc_simple_parse_clk(dev, np, dai, codecs); if (ret < 0) goto out_put_node; - snprintf(dai_name, sizeof(dai_name), "be.%s", codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); + if (ret < 0) + goto out_put_node; /* check "prefix" from top node */ snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, @@ -240,14 +201,23 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, } simple_parse_convert(dev, np, &dai_props->adata); + simple_parse_mclk_fs(top, np, codec, dai_props, prefix); + + ret = asoc_simple_parse_tdm(np, dai); + if (ret) + goto out_put_node; + + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto out_put_node; snd_soc_dai_link_set_capabilities(dai_link); - ret = simple_link_init(priv, node, codec, li, prefix, dai_name); + dai_link->ops = &simple_ops; + dai_link->init = asoc_simple_dai_init; out_put_node: - li->link++; - of_node_put(node); return ret; } @@ -260,19 +230,23 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); + struct asoc_simple_dai *cpu_dai = simple_props_to_dai_cpu(dai_props, 0); + struct asoc_simple_dai *codec_dai = simple_props_to_dai_codec(dai_props, 0); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); + struct device_node *top = dev->of_node; struct device_node *cpu = NULL; struct device_node *node = NULL; struct device_node *plat = NULL; - char dai_name[64]; char prop[128]; char *prefix = ""; - int ret; + int ret, single_cpu = 0; cpu = np; node = of_get_parent(np); + li->link++; dev_dbg(dev, "link_of (%pOF)\n", node); @@ -283,11 +257,18 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); - ret = simple_parse_node(priv, cpu, li, prefix, 1); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) goto dai_link_of_err; - ret = simple_parse_node(priv, codec, li, prefix, 0); + simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); + + ret = asoc_simple_parse_dai(cpu, cpus, &single_cpu); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_dai(codec, codecs, NULL); if (ret < 0) goto dai_link_of_err; @@ -295,18 +276,39 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; - snprintf(dai_name, sizeof(dai_name), - "%s-%s", cpus->dai_name, codecs->dai_name); - ret = simple_link_init(priv, node, codec, li, prefix, dai_name); + ret = asoc_simple_parse_tdm(cpu, cpu_dai); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_tdm(codec, codec_dai); if (ret < 0) goto dai_link_of_err; + ret = asoc_simple_parse_clk(dev, cpu, cpu_dai, cpus); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_clk(dev, codec, codec_dai, codecs); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + cpus->dai_name, + codecs->dai_name); + if (ret < 0) + goto dai_link_of_err; + + dai_link->ops = &simple_ops; + dai_link->init = asoc_simple_dai_init; + + asoc_simple_canonicalize_cpu(cpus, single_cpu); + asoc_simple_canonicalize_platform(platforms, cpus); + dai_link_of_err: of_node_put(plat); of_node_put(node); - li->link++; - return ret; } -- 2.20.1

1 0

[PATCH v1 01/14] spi: pxa2xx: Use one point of return when ->probe() fails
by Andy Shevchenko 23 Apr '21

23 Apr '21

When we can't allocate SPI controller, jump to the error path rather than return locally. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- drivers/spi/spi-pxa2xx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index 0cc767283674..0ef22045b006 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1701,8 +1701,8 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) if (!controller) { dev_err(&pdev->dev, "cannot alloc spi_controller\n"); - pxa_ssp_free(ssp); - return -ENOMEM; + status = -ENOMEM; + goto out_error_controller_alloc; } drv_data = spi_controller_get_devdata(controller); drv_data->controller = controller; -- 2.30.2

2 16

New udev rules for firewire character device are added to systemd
by Takashi Sakamoto 23 Apr '21

23 Apr '21

Hi, Summary of this message: * systemd got udev rules with new database for firewire node/unit * then fw character device for audio is owned by 'audio' group with ACL * the entries are added by my investigation, thus doesn't cover all * if you have firewire audio devices not listed in README of below repository, please contact to me with image of configuration ROM: * https://github.com/takaswie/am-config-roms/ The way to create image file of configuration ROM is typically: ``` $ cat /sys/bus/firewire/devices/fw1/config_rom > filename.img ``` Here, I presuppose that Linux FireWire subsystem detects your device as 'fw1'. Well, in the past, access permission of Linux firewire character device is decided by udev rules just for video devices[1]. This was inconvenient some project such as ALSA and FFADO to produce audio application. The source code of libffado includes own file for udev rules[2] to take firewire character device owned by 'audio' group. Additionally the rules gives 'ID_FFADO' tag, and systemd includes another udev rule[3] to ACL at logging-in time according to it. As a whole, the above is not comprehensive and self-contained. I proposed patchset to systemd for better solution and today it was merged. * https://github.com/systemd/systemd/pull/19124 In the patchset, I add some udev rules, based on hwdb for new entries of node and units in IEEE 1394 bus. You can see the database[4]. The entries of database have below variables when matching to either node or unit devices: * IEEE1394_UNIT_FUNCTION_MIDI * IEEE1394_UNIT_FUNCTION_AUDIO * IEEE1394_UNIT_FUNCTION_VIDEO The added udev rules interpret the content of variables and decide group owner of fw character device(see [1]). Furthermore, the variables are used again to decide ACL in logging-in time(see [3], too). The entries of database also include below variables: * ID_VENDOR_FROM_DATABASE * ID_MODEL_FROM_DATABASE They are expected to use applications such as PipeWire and PulseAudio for better names of sound device, which binds to unit instead of node. I expect the variables can obsolete my former patch for pulseaudio[5]. I handy write the entries of database from my investigation, thus it could includes the lack of your device, or mistakes. I wish you to contact to me with image file of configuration ROM when you can not find your device in README of my collection repository[6], or when you find any mistakes in database file. Thanks for your cooperation in advance. [1] 4 rules in 'rules.d/50-udev-default.rules' https://github.com/systemd/systemd/blob/main/rules.d/50-udev-default.rules.… [2] many rules in 'libffado/60-ffado.rules' http://subversion.ffado.org/browser/trunk/libffado/libffado/60-ffado.rules?… [3] 'src/login/70-uaccess.rules.m4' https://github.com/systemd/systemd/blob/main/src/login/70-uaccess.rules.m4 [4] 'hwdb.d/80-ieee1394-unit-function.hwdb' https://github.com/systemd/systemd/blob/main/hwdb.d/80-ieee1394-unit-functi… [5] udev: use ID_MODEL/ID_VENDOR to give friendly name for FireWire devices https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/3ac73598c67c [6] https://github.com/takaswie/am-config-roms/ Takashi Sakamoto

1 0

[PATCH] ASoC: adau17x1: Avoid overwriting CHPF
by Niklas Carlsson 22 Apr '21

22 Apr '21

From: Niklas Carlsson <niklasc(a)axis.com> Configuring number of channels per LRCLK frame by using e.g. snd_soc_dai_set_tdm_slot before configuring DAI format was being overwritten by the latter due to a regmap_write which would write over the whole register. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau17x1.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 546ee8178038..8aae7ab74091 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -553,6 +553,7 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, { struct adau *adau = snd_soc_component_get_drvdata(dai->component); unsigned int ctrl0, ctrl1; + unsigned int ctrl0_mask; int lrclk_pol; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -612,8 +613,16 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, if (lrclk_pol) ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + /* Set the mask to update all relevant bits in ADAU17X1_SERIAL_PORT0 */ + ctrl0_mask = ADAU17X1_SERIAL_PORT0_MASTER | + ADAU17X1_SERIAL_PORT0_LRCLK_POL | + ADAU17X1_SERIAL_PORT0_BCLK_POL | + ADAU17X1_SERIAL_PORT0_PULSE_MODE; + + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0_mask, + ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, ctrl1); adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; -- 2.20.1

2 1

[PATCH] ASoC: tegra: mark runtime-pm functions as __maybe_unused
by Arnd Bergmann 22 Apr '21

22 Apr '21

From: Arnd Bergmann <arnd(a)arndb.de> A reorganization of the driver source led to two of them causing a compile time warning in some configurations: tegra/tegra20_spdif.c:36:12: error: 'tegra20_spdif_runtime_resume' defined but not used [-Werror=unused-function] 36 | static int tegra20_spdif_runtime_resume(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra20_spdif.c:27:12: error: 'tegra20_spdif_runtime_suspend' defined but not used [-Werror=unused-function] 27 | static int tegra20_spdif_runtime_suspend(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra30_ahub.c:64:12: error: 'tegra30_ahub_runtime_resume' defined but not used [-Werror=unused-function] 64 | static int tegra30_ahub_runtime_resume(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra30_ahub.c:43:12: error: 'tegra30_ahub_runtime_suspend' defined but not used [-Werror=unused-function] 43 | static int tegra30_ahub_runtime_suspend(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ Mark these functions as __maybe_unused to avoid this kind of warning. Fixes: b5571449e618 ("ASoC: tegra30: ahub: Remove handing of disabled runtime PM") Fixes: c53b396f0dd4 ("ASoC: tegra20: spdif: Remove handing of disabled runtime PM") Fixes: 80ec4a4cb36d ("ASoC: tegra20: i2s: Remove handing of disabled runtime PM") Fixes: b5f6f781fcb2 ("ASoC: tegra30: i2s: Remove handing of disabled runtime PM") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/tegra/tegra20_i2s.c | 4 ++-- sound/soc/tegra/tegra20_spdif.c | 4 ++-- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 1b27f81c10fe..b280ebd72591 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -34,7 +34,7 @@ #define DRV_NAME "tegra20-i2s" -static int tegra20_i2s_runtime_suspend(struct device *dev) +static __maybe_unused int tegra20_i2s_runtime_suspend(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); @@ -45,7 +45,7 @@ static int tegra20_i2s_runtime_suspend(struct device *dev) return 0; } -static int tegra20_i2s_runtime_resume(struct device *dev) +static __maybe_unused int tegra20_i2s_runtime_resume(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); int ret; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 7b597ee63cb5..de698ff2a69c 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -24,7 +24,7 @@ #define DRV_NAME "tegra20-spdif" -static int tegra20_spdif_runtime_suspend(struct device *dev) +static __maybe_unused int tegra20_spdif_runtime_suspend(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); @@ -33,7 +33,7 @@ static int tegra20_spdif_runtime_suspend(struct device *dev) return 0; } -static int tegra20_spdif_runtime_resume(struct device *dev) +static __maybe_unused int tegra20_spdif_runtime_resume(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); int ret; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d1718f3af3cd..4692c70ed933 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -40,7 +40,7 @@ static inline void tegra30_audio_write(u32 reg, u32 val) regmap_write(ahub->regmap_ahub, reg, val); } -static int tegra30_ahub_runtime_suspend(struct device *dev) +static __maybe_unused int tegra30_ahub_runtime_suspend(struct device *dev) { regcache_cache_only(ahub->regmap_apbif, true); regcache_cache_only(ahub->regmap_ahub, true); @@ -61,7 +61,7 @@ static int tegra30_ahub_runtime_suspend(struct device *dev) * stopping streams should dynamically adjust the clock as required. However, * this is not yet implemented. */ -static int tegra30_ahub_runtime_resume(struct device *dev) +static __maybe_unused int tegra30_ahub_runtime_resume(struct device *dev) { int ret; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 8730ffa0f691..36344f0a64c1 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -36,7 +36,7 @@ #define DRV_NAME "tegra30-i2s" -static int tegra30_i2s_runtime_suspend(struct device *dev) +static __maybe_unused int tegra30_i2s_runtime_suspend(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); @@ -47,7 +47,7 @@ static int tegra30_i2s_runtime_suspend(struct device *dev) return 0; } -static int tegra30_i2s_runtime_resume(struct device *dev) +static __maybe_unused int tegra30_i2s_runtime_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); int ret; -- 2.29.2

2 1

[PATCH] ALSA: usb-audio: Apply implicit feedback mode for BOSS devices
by Takashi Iwai 22 Apr '21

22 Apr '21

During the recent rewrite of the implicit feedback support, we've tested to apply the implicit fb on BOSS devices, but it failed, as the capture stream didn't start without the playback. As the end result, it got another type of quirk for tying both streams but starts playback always (commit 6234fdc1cede "ALSA: usb-audio: Quirk for BOSS GT-001"). Meanwhile, Mike Oliphant has tested the real implicit feedback mode for the playback again with the latest code, and found out that it actually works if the initial feedback sync is skipped; that is, on those BOSS devices, the playback stream has to be started at first without waiting for the capture URB completions. Otherwise it gets stuck. In the rest operations after the capture stream processed, we can take them as the implicit feedback source. This patch is an attempt to improve the support for BOSS devices with the implicit feedback mode in the way described above. It adds a new flag to snd_usb_audio, playback_first, indicating that the playback stream starts without sync with the initial capture completion. This flag is set in the quirk table with the new IMPLICIT_FB_BOTH type. Reported-and-tested-by: Mike Oliphant <oliphant(a)nostatic.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 3 ++- sound/usb/implicit.c | 35 +++++++++++++++++++++++------------ sound/usb/usbaudio.h | 1 + 3 files changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 102d53515a76..f4c3d2b38abb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1375,7 +1375,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (!ep_state_update(ep, EP_STATE_STOPPED, EP_STATE_RUNNING)) goto __error; - if (snd_usb_endpoint_implicit_feedback_sink(ep)) { + if (snd_usb_endpoint_implicit_feedback_sink(ep) && + !ep->chip->playback_first) { for (i = 0; i < ep->nurbs; i++) { struct snd_urb_ctx *ctx = ep->urb + i; list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 19622c58b72b..4bd9c105a529 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -21,6 +21,7 @@ enum { IMPLICIT_FB_NONE, IMPLICIT_FB_GENERIC, IMPLICIT_FB_FIXED, + IMPLICIT_FB_BOTH, /* generic playback + capture (for BOSS) */ }; struct snd_usb_implicit_fb_match { @@ -36,6 +37,9 @@ struct snd_usb_implicit_fb_match { #define IMPLICIT_FB_FIXED_DEV(vend, prod, ep, ifnum) \ { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_FIXED, .ep_num = (ep),\ .iface = (ifnum) } +#define IMPLICIT_FB_BOTH_DEV(vend, prod, ep, ifnum) \ + { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_BOTH, .ep_num = (ep),\ + .iface = (ifnum) } #define IMPLICIT_FB_SKIP_DEV(vend, prod) \ { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_NONE } @@ -75,14 +79,14 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */ }; @@ -268,10 +272,17 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } } - /* Don't apply playback quirks for the devices with capture quirk */ + /* Special handling for devices with capture quirks */ p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); - if (p && p->type == IMPLICIT_FB_FIXED) - return 0; /* no quirk */ + if (p) { + switch (p->type) { + case IMPLICIT_FB_FIXED: + return 0; /* no quirk */ + case IMPLICIT_FB_BOTH: + chip->playback_first = 1; + return add_generic_implicit_fb(chip, fmt, alts); + } + } /* Generic UAC2 implicit feedback */ if (attr == USB_ENDPOINT_SYNC_ASYNC && @@ -321,7 +332,7 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, const struct snd_usb_implicit_fb_match *p; p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); - if (p && p->type == IMPLICIT_FB_FIXED) + if (p && (p->type == IMPLICIT_FB_FIXED || p->type == IMPLICIT_FB_BOTH)) return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); return 0; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a536ee33d36e..538831cbd925 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ unsigned int need_delayed_register:1; /* warn for delayed registration */ + unsigned int playback_first:1; /* for implicit fb: don't wait for the first capture URBs */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; -- 2.26.2

4 13

[PATCH] ASoC: rt711-sdca: add the notification when volume changed
by shumingf＠realtek.com 22 Apr '21

22 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> This patch adds the return value when the volume settings were changed. The userspace application might monitor the kcontrols to check which control changed. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 3ab9048b4ea3..98b26bd69e19 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -509,12 +509,16 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); unsigned int read_l, read_r, gain_l_val, gain_r_val; - unsigned int i, adc_vol_flag = 0; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; if (strstr(ucontrol->id.name, "FU1E Capture Volume") || strstr(ucontrol->id.name, "FU0F Capture Volume")) adc_vol_flag = 1; + regmap_read(rt711->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt711->mbq_regmap, mc->rreg, &rvalue); + /* control value to 2's complement value */ /* L Channel */ gain_l_val = ucontrol->value.integer.value[0]; @@ -560,6 +564,9 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, gain_r_val &= 0xffff; } + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + for (i = 0; i < 3; i++) { /* retry 3 times at most */ /* Lch*/ regmap_write(rt711->mbq_regmap, mc->reg, gain_l_val); @@ -573,7 +580,7 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, break; } - return i == 3 ? -EIO : 0; + return i == 3 ? -EIO : changed; } static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, -- 2.29.0

3 2

[PATCH v3] ASoC: rt711-sdca: change capture switch controls
by shumingf＠realtek.com 22 Apr '21

22 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The DAPM event and mixer control could mute/unmute the capture directly. That will be confused that capture still works if the user settings is unmute before the capture. Therefore, this patch uses the variables to record the capture switch status of DAPM and mixer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 208 +++++++++++++++++++++++++++------- sound/soc/codecs/rt711-sdca.h | 2 + 2 files changed, 167 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..3ab9048b4ea3 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -642,6 +642,154 @@ static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt711_sdca_set_fu0f_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + + if (rt711->fu0f_dapm_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + if (err < 0) + return err; + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + if (err < 0) + return err; + } else { + if (rt711->fu0f_mixer_l_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + } else { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x00); + } + if (err < 0) + return err; + + if (rt711->fu0f_mixer_r_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + } else { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x00); + } + if (err < 0) + return err; + } + return 0; +} + +static int rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + + if (rt711->fu1e_dapm_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + if (err < 0) + return err; + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + if (err < 0) + return err; + } else { + if (rt711->fu1e_mixer_l_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + } else { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x00); + } + if (err < 0) + return err; + + if (rt711->fu1e_mixer_r_mute) { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + } else { + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x00); + } + if (err < 0) + return err; + } + return 0; +} + +static int rt711_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu1e_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu1e_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu1e_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu1e_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu1e_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu1e_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu1e_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + +static int rt711_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu0f_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu0f_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -652,14 +800,10 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), + SOC_DOUBLE_EXT("FU1E Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu1e_capture_get, rt711_sdca_fu1e_capture_put), + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu0f_capture_get, rt711_sdca_fu0f_capture_put), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), @@ -809,28 +953,15 @@ static int rt711_sdca_fu0f_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu0f_dapm_mute = false; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); + rt711->fu0f_dapm_mute = true; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; } return 0; @@ -842,29 +973,16 @@ static int rt711_sdca_fu1e_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu1e_dapm_mute = false; + rt711_sdca_set_fu1e_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); - break; + rt711->fu1e_dapm_mute = true; + rt711_sdca_set_fu1e_capture_ctl(rt711); + break; } return 0; } @@ -1330,6 +1448,10 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, */ rt711->hw_init = false; rt711->first_hw_init = false; + rt711->fu0f_dapm_mute = true; + rt711->fu1e_dapm_mute = true; + rt711->fu0f_mixer_l_mute = rt711->fu0f_mixer_r_mute = true; + rt711->fu1e_mixer_l_mute = rt711->fu1e_mixer_r_mute = true; /* JD source uses JD2 in default */ rt711->jd_src = RT711_JD2; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 98a022cec0bd..43ae82b7fdb3 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -30,6 +30,8 @@ struct rt711_sdca_priv { int jack_type, jd_src; unsigned int scp_sdca_stat1, scp_sdca_stat2; int hw_ver; + bool fu0f_dapm_mute, fu0f_mixer_l_mute, fu0f_mixer_r_mute; + bool fu1e_dapm_mute, fu1e_mixer_l_mute, fu1e_mixer_r_mute; }; struct sdw_stream_data { -- 2.29.0

3 2

[bug_report] ALSA: snd_hda_intel probe failed with error -16
by Steven Skeard 22 Apr '21

22 Apr '21

Hi all, I tried to get some help with this before but I don't think I articulated the problem very well and it seemed to fall between the cracks, so here I am to try again (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-February/181262.…). With the default ALSA configuration in /etc/modprobe.d/alsa-base.conf, I experience a consistent race condition where if snd-usb-audio loads before snd-hda-intel, then the snd-hda-intel driver will fail to load with the -16 error as seen in the dmsg output at the end of this alsa info dump (http://alsa-project.org/db/?f=a23ccbb818fec6eaf08b032504eac5d5d8a30589). When this error occurs, I get a single dummy output and no working audio. In this state the only way to get working audio is to reboot my computer and hope things load in the correct order. After much digging I found a couple links on the arch forums where people experienced similar problems (https://bbs.archlinux.org/viewtopic.php?id=264091, https://bbs.archlinux.org/viewtopic.php?id=260373). After reading both of those and looking at my dmsg outputs, I tested the following modifications to my modprobe configuration: 1. Remove the default line for the usb driver: "options snd-usb-audio index=-2" 2. Add a line to specify that the intel driver should be loaded first: "options snd-hda-intel index=0" 3. Add a line to specify that the realtek codec driver should be loaded second: "options snd-hda-codec-realtek index=1" 4. Add a line to specify that the usb driver should be loaded third: "options snd-usb-audio index=2" And this works, and not only works but works every reboot! I have uploaded an additional alsa info with the work around applied (http://alsa-project.org/db/?f=f2d90b82434ba89b834a67eaefe10726d538e829) So while at the end of the day I have a working audio configuration, I can't help but feel like this is a bug. But the question I have is whether this is a bug with ALSA, or with the kernel driver? And how do I properly report this so it can be fixed? Thanks, Steven Skeard

2 1

[PATCH] ASoC: fsl: imx-pcm-dma: Don't request dma channel in probe
by Shengjiu Wang 22 Apr '21

22 Apr '21

There are two requirements that we need to move the request of dma channel from probe to open. - When dma device binds with power-domains, the power will be enabled when we request dma channel. If the request of dma channel happen on probe, then the power-domains will be always enabled after kernel boot up, which is not good for power saving, so we need to move the request of dma channel to .open(); - With FE-BE case, if the dma channel is requested in probe, then there will be below issue, which is caused by that the dma channel will be requested duplicately [ 638.906268] sysfs: cannot create duplicate filename '/devices/soc0/soc/2000000.bus/2000000.spba-bus/2024000.esai/dma:tx' [ 638.919061] CPU: 1 PID: 673 Comm: aplay Not tainted 5.7.0-rc1-12956-gfc64b2585593 #287 [ 638.927113] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 638.933690] [<c0110dd8>] (unwind_backtrace) from [<c010b8ec>] (show_stack+0x10/0x14) [ 638.941464] [<c010b8ec>] (show_stack) from [<c0557fc0>] (dump_stack+0xe4/0x118) [ 638.948808] [<c0557fc0>] (dump_stack) from [<c032aeb4>] (sysfs_warn_dup+0x50/0x64) [ 638.956406] [<c032aeb4>] (sysfs_warn_dup) from [<c032b1a8>] (sysfs_do_create_link_sd+0xc8/0xd4) [ 638.965134] [<c032b1a8>] (sysfs_do_create_link_sd) from [<c05dc668>] (dma_request_chan+0xb0/0x210) [ 638.974120] [<c05dc668>] (dma_request_chan) from [<c05dc7d0>] (dma_request_slave_channel+0x8/0x14) [ 638.983111] [<c05dc7d0>] (dma_request_slave_channel) from [<c09d5548>] (fsl_asrc_dma_hw_params+0x1e0/0x438) [ 638.992881] [<c09d5548>] (fsl_asrc_dma_hw_params) from [<c09c1654>] (soc_pcm_hw_params+0x4a0/0x6a8) [ 639.001952] [<c09c1654>] (soc_pcm_hw_params) from [<c09c39d4>] (dpcm_fe_dai_hw_params+0x70/0xe4) [ 639.010765] [<c09c39d4>] (dpcm_fe_dai_hw_params) from [<c099b274>] (snd_pcm_hw_params+0x158/0x418) [ 639.019750] [<c099b274>] (snd_pcm_hw_params) from [<c099c5a0>] (snd_pcm_ioctl+0x734/0x183c) [ 639.028129] [<c099c5a0>] (snd_pcm_ioctl) from [<c029ff94>] (ksys_ioctl+0x2ac/0xb98) [ 639.035812] [<c029ff94>] (ksys_ioctl) from [<c0100080>] (ret_fast_syscall+0x0/0x28) [ 639.043490] Exception stack(0xec529fa8 to 0xec529ff0) [ 639.048565] 9fa0: bee84650 01321870 00000004 c25c4111 bee84650 0002000f [ 639.056766] 9fc0: bee84650 01321870 01321820 00000036 00001f40 00000000 0002c2f8 00000003 [ 639.064964] 9fe0: b6f483fc bee8451c b6ee2655 b6e1dcf8 [ 639.070339] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-pcm-dma.c | 173 +++++++++++++++++++++++++++++++++--- 1 file changed, 159 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 04a9bc749016..dae53b384df4 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -11,6 +11,7 @@ #include <linux/dmaengine.h> #include <linux/types.h> #include <linux/module.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,24 +30,168 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = filter, -}; +static int imx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_config config; + struct dma_chan *chan; + int ret = 0; -int imx_pcm_dma_init(struct platform_device *pdev, size_t size) + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + chan = snd_dmaengine_pcm_get_chan(substream); + if (!chan) + return -EINVAL; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + snd_dmaengine_pcm_set_config_from_dai_data(substream, + dma_data, + &config); + return dmaengine_slave_config(chan, &config); +} + +static int imx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) { - struct snd_dmaengine_pcm_config *config; + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static snd_pcm_uframes_t imx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_pointer(substream); +} + +static int imx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + return snd_dmaengine_pcm_trigger(substream, cmd); +} + +static int imx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct device *dev = component->dev; + struct snd_pcm_hardware hw; + struct dma_chan *chan; + int ret; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "failed to set pcm hw params periods\n"); + return ret; + } + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + chan = dma_request_slave_channel(cpu_dai->dev, tx ? "tx" : "rx"); + if (!chan) { + /* Try to request channel using compat_filter_fn */ + chan = snd_dmaengine_pcm_request_channel(filter, + dma_data->filter_data); + if (!chan) + return -ENXIO; + } - config = devm_kzalloc(&pdev->dev, - sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); - if (!config) - return -ENOMEM; - *config = imx_dmaengine_pcm_config; + ret = snd_dmaengine_pcm_open(substream, chan); + if (ret) + goto pcm_open_fail; - return devm_snd_dmaengine_pcm_register(&pdev->dev, - config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(chan->device->dev); + hw.buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE; + hw.fifo_size = dma_data->fifo_size; + + /* Refine the hw according to caps of DMA. */ + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); + if (ret < 0) + goto refine_runtime_hwparams_fail; + + snd_soc_set_runtime_hwparams(substream, &hw); + + /* Support allocate memory from IRAM */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_IRAM, + chan->device->dev, + hw.buffer_bytes_max, + &substream->dma_buffer); + if (ret < 0) + goto alloc_pagas_fail; + + return 0; + +alloc_pagas_fail: +refine_runtime_hwparams_fail: + snd_dmaengine_pcm_close(substream); +pcm_open_fail: + dma_release_channel(chan); + + return ret; +} + +static int imx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + + return snd_dmaengine_pcm_close_release_chan(substream); +} + +static int imx_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + + return dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); +} + +static const struct snd_soc_component_driver imx_pcm_component = { + .name = "imx-pcm-dma", + .pcm_construct = imx_pcm_new, + .open = imx_pcm_open, + .close = imx_pcm_close, + .hw_params = imx_pcm_hw_params, + .hw_free = imx_pcm_hw_free, + .trigger = imx_pcm_trigger, + .pointer = imx_pcm_pointer, +}; + +int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +{ + return devm_snd_soc_register_component(&pdev->dev, + &imx_pcm_component, NULL, 0); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -- 2.21.0

5 17

[PATCH v2] pinctrl: mediatek: use spin lock in mtk_rmw
by Tzung-Bi Shih 22 Apr '21

22 Apr '21

Commit 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") uses mutex lock in mtk_rmw. However the function is possible called from atomic context. For example call trace: mutex_lock+0x28/0x64 mtk_rmw+0x38/0x80 [snip] max98357a_daiops_trigger+0x8c/0x9c soc_pcm_trigger+0x5c/0x10c The max98357a_daiops_trigger() could run in either atomic or non-atomic context. As a result, dmesg shows some similar messages: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:254". Uses spin lock in mtk_rmw instead. Fixes: 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v1[1]: - Snip call trace. - Use _irqsave and _irqrestore version of spin lock. [1]: https://patchwork.ozlabs.org/project/linux-gpio/list/?series=239749 drivers/pinctrl/mediatek/pinctrl-moore.c | 2 +- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c | 5 +++-- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h | 2 +- drivers/pinctrl/mediatek/pinctrl-paris.c | 2 +- 4 files changed, 6 insertions(+), 5 deletions(-) diff --git a/drivers/pinctrl/mediatek/pinctrl-moore.c b/drivers/pinctrl/mediatek/pinctrl-moore.c index f77921957f15..3a4a23c40a71 100644 --- a/drivers/pinctrl/mediatek/pinctrl-moore.c +++ b/drivers/pinctrl/mediatek/pinctrl-moore.c @@ -619,7 +619,7 @@ int mtk_moore_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); /* Copy from internal struct mtk_pin_desc to register to the core */ pins = devm_kmalloc_array(&pdev->dev, hw->soc->npins, sizeof(*pins), diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c index fcf7c3eeee4a..fc79b3a7a188 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c @@ -57,15 +57,16 @@ static u32 mtk_r32(struct mtk_pinctrl *pctl, u8 i, u32 reg) void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set) { u32 val; + unsigned long flags; - mutex_lock(&pctl->lock); + spin_lock_irqsave(&pctl->lock, flags); val = mtk_r32(pctl, i, reg); val &= ~mask; val |= set; mtk_w32(pctl, i, reg, val); - mutex_unlock(&pctl->lock); + spin_unlock_irqrestore(&pctl->lock, flags); } static int mtk_hw_pin_field_lookup(struct mtk_pinctrl *hw, diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h index 65eac708a3b3..13836bb188b7 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h @@ -252,7 +252,7 @@ struct mtk_pinctrl { struct mtk_pinctrl_group *groups; const char **grp_names; /* lock pin's register resource to avoid multiple threads issue*/ - struct mutex lock; + spinlock_t lock; }; void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set); diff --git a/drivers/pinctrl/mediatek/pinctrl-paris.c b/drivers/pinctrl/mediatek/pinctrl-paris.c index 48e823f6d293..85db2e4377f0 100644 --- a/drivers/pinctrl/mediatek/pinctrl-paris.c +++ b/drivers/pinctrl/mediatek/pinctrl-paris.c @@ -970,7 +970,7 @@ int mtk_paris_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); err = mtk_pctrl_build_state(pdev); if (err) { -- 2.31.1.368.gbe11c130af-goog

2 1

[PATCH] pinctrl: mediatek: use spin lock in mtk_rmw
by Tzung-Bi Shih 22 Apr '21

22 Apr '21

Commit 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") uses mutex lock in mtk_rmw. However the function is possible called from atomic context. For example call trace: mutex_lock+0x28/0x64 mtk_rmw+0x38/0x80 mtk_hw_set_value+0x100/0x138 mtk_gpio_set+0x48/0x58 gpiod_set_raw_value_commit+0xf4/0x110 gpiod_set_value_nocheck+0x4c/0x80 gpiod_set_value+0x4c/0x6c max98357a_daiops_trigger+0x8c/0x9c soc_pcm_trigger+0x5c/0x10c The max98357a_daiops_trigger() could run in either atomic or non-atomic context. As a result, dmesg shows some similar messages: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:254". Uses spin lock in mtk_rmw instead. Fixes: 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/pinctrl/mediatek/pinctrl-moore.c | 2 +- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c | 4 ++-- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h | 2 +- drivers/pinctrl/mediatek/pinctrl-paris.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/pinctrl/mediatek/pinctrl-moore.c b/drivers/pinctrl/mediatek/pinctrl-moore.c index f77921957f15..3a4a23c40a71 100644 --- a/drivers/pinctrl/mediatek/pinctrl-moore.c +++ b/drivers/pinctrl/mediatek/pinctrl-moore.c @@ -619,7 +619,7 @@ int mtk_moore_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); /* Copy from internal struct mtk_pin_desc to register to the core */ pins = devm_kmalloc_array(&pdev->dev, hw->soc->npins, sizeof(*pins), diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c index fcf7c3eeee4a..9ffe01a8ceca 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c @@ -58,14 +58,14 @@ void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set) { u32 val; - mutex_lock(&pctl->lock); + spin_lock(&pctl->lock); val = mtk_r32(pctl, i, reg); val &= ~mask; val |= set; mtk_w32(pctl, i, reg, val); - mutex_unlock(&pctl->lock); + spin_unlock(&pctl->lock); } static int mtk_hw_pin_field_lookup(struct mtk_pinctrl *hw, diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h index 65eac708a3b3..13836bb188b7 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h @@ -252,7 +252,7 @@ struct mtk_pinctrl { struct mtk_pinctrl_group *groups; const char **grp_names; /* lock pin's register resource to avoid multiple threads issue*/ - struct mutex lock; + spinlock_t lock; }; void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set); diff --git a/drivers/pinctrl/mediatek/pinctrl-paris.c b/drivers/pinctrl/mediatek/pinctrl-paris.c index 48e823f6d293..85db2e4377f0 100644 --- a/drivers/pinctrl/mediatek/pinctrl-paris.c +++ b/drivers/pinctrl/mediatek/pinctrl-paris.c @@ -970,7 +970,7 @@ int mtk_paris_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); err = mtk_pctrl_build_state(pdev); if (err) { -- 2.31.0.208.g409f899ff0-goog

2 1

[PATCH 0/5] ASoC: simple-card/audio-graph: cleanups
by Kuninori Morimoto 21 Apr '21

21 Apr '21

Hi Mark These patches cleanups simple-card / audio-graph. This is part of prepare for new audio-graph-card2. Kuninori Morimoto (5): ASoC: audio-graph: add graph_parse_node() ASoC: audio-graph: add graph_link_init() ASoC: simple-card: add simple_parse_node() ASoC: simple-card: add simple_link_init() ASoC: audio-graph: tidyup graph_dai_link_of_dpcm() sound/soc/generic/audio-graph-card.c | 204 ++++++++++++--------------- sound/soc/generic/simple-card.c | 178 ++++++++++++----------- 2 files changed, 182 insertions(+), 200 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: Intel: KMB: Fix random noise at the HDMI output
by Sia Jee Heng 21 Apr '21

21 Apr '21

Random noise could be heard when playing audio to the HDMI output. This is due to the IEC conversion is invoked in the external loop. As a result, this additional loop takes up a lot of the processing cycle. hdmi_reformat_iec958() process the conversion using an internal loop, it is safe to move it out from the external loop to avoid unnecessary processing cycle been spent. Furthermore, ALSA IEC958 plugin works in 32bit format only. Signed-off-by: Sia Jee Heng <jee.heng.sia(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/keembay/kmb_platform.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 0fd1e8f62c89..96741c7c0fba 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -105,14 +105,15 @@ static unsigned int kmb_pcm_tx_fn(struct kmb_i2s_info *kmb_i2s, void *buf = runtime->dma_area; int i; + if (kmb_i2s->iec958_fmt) + hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr); + /* KMB i2s uses two separate L/R FIFO */ for (i = 0; i < kmb_i2s->fifo_th; i++) { if (kmb_i2s->config.data_width == 16) { writel(((u16(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0)); writel(((u16(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0)); } else { - if (kmb_i2s->iec958_fmt) - hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr); writel(((u32(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0)); writel(((u32(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0)); } base-commit: bf05bf16c76bb44ab5156223e1e58e26dfe30a88 -- 2.18.0

2 1

[PATCH] ASoC: amd: drop S24_LE format support
by Vijendar Mukunda 21 Apr '21

21 Apr '21

AMD I2S Controller doesn't support S24_LE format. Remove S24_LE format support from ACP DMA driver and CPU DAI Driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 6 ++---- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++---- 2 files changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index 396cb7d..de6f70d 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -264,8 +264,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = { .playback = { .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, @@ -274,8 +273,7 @@ static struct snd_soc_dai_driver acp3x_i2s_dai = { .capture = { .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rate_min = 8000, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 417cda2..f22bb2b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -24,8 +24,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_playback = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_96000, @@ -45,8 +44,7 @@ static const struct snd_pcm_hardware acp3x_pcm_hardware_capture = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, -- 2.7.4

2 1

[v7 1/4] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 21 Apr '21

21 Apr '21

From: Chris Morgan <macromorgan(a)hotmail.com> Add rk817 codec support cell to rk808 mfd driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. drivers/mfd/rk808.c | 85 +++++++++++++++++++++++++++++++++++++++ include/linux/mfd/rk808.h | 81 +++++++++++++++++++++++++++++++++++++ 2 files changed, 166 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..9231209184e0 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -65,6 +65,7 @@ static bool rk817_is_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case RK817_SECONDS_REG ... RK817_WEEKS_REG: case RK817_RTC_STATUS_REG: + case RK817_CODEC_DTOP_LPT_SRST: case RK817_INT_STS_REG0: case RK817_INT_STS_REG1: case RK817_INT_STS_REG2: @@ -163,6 +164,11 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { @@ -201,6 +207,85 @@ static const struct rk808_reg_data rk808_pre_init_reg[] = { static const struct rk808_reg_data rk817_pre_init_reg[] = { {RK817_RTC_CTRL_REG, RTC_STOP, RTC_STOP}, + /* Codec specific registers */ + { RK817_CODEC_DTOP_VUCTL, MASK_ALL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, MASK_ALL, 0x06 }, + { RK817_CODEC_AADC_CFG0, MASK_ALL, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC2, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_NG, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_HPF, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AHP_ANTI0, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_ANTI1, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_CFG0, MASK_ALL, 0xe0 }, + { RK817_CODEC_AHP_CFG1, MASK_ALL, 0x1f }, + { RK817_CODEC_AHP_CP, MASK_ALL, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, MASK_ALL, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, MASK_ALL, 0x44 }, + { RK817_CODEC_APLL_CFG0, MASK_ALL, 0x04 }, + { RK817_CODEC_APLL_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_APLL_CFG2, MASK_ALL, 0x30 }, + { RK817_CODEC_APLL_CFG3, MASK_ALL, 0x19 }, + { RK817_CODEC_APLL_CFG4, MASK_ALL, 0x65 }, + { RK817_CODEC_APLL_CFG5, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_CKM, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_RSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, MASK_ALL, 0x00 }, {RK817_GPIO_INT_CFG, RK817_INT_POL_MSK, RK817_INT_POL_L}, {RK817_SYS_CFG(1), RK817_HOTDIE_TEMP_MSK | RK817_TSD_TEMP_MSK, RK817_HOTDIE_105 | RK817_TSD_140}, diff --git a/include/linux/mfd/rk808.h b/include/linux/mfd/rk808.h index e07f6e61cd38..a96e6d43ca06 100644 --- a/include/linux/mfd/rk808.h +++ b/include/linux/mfd/rk808.h @@ -437,6 +437,87 @@ enum rk809_reg_id { #define RK817_RTC_COMP_LSB_REG 0x10 #define RK817_RTC_COMP_MSB_REG 0x11 +/* RK817 Codec Registers */ +#define RK817_CODEC_DTOP_VUCTL 0x12 +#define RK817_CODEC_DTOP_VUCTIME 0x13 +#define RK817_CODEC_DTOP_LPT_SRST 0x14 +#define RK817_CODEC_DTOP_DIGEN_CLKE 0x15 +#define RK817_CODEC_AREF_RTCFG0 0x16 +#define RK817_CODEC_AREF_RTCFG1 0x17 +#define RK817_CODEC_AADC_CFG0 0x18 +#define RK817_CODEC_AADC_CFG1 0x19 +#define RK817_CODEC_DADC_VOLL 0x1a +#define RK817_CODEC_DADC_VOLR 0x1b +#define RK817_CODEC_DADC_SR_ACL0 0x1e +#define RK817_CODEC_DADC_ALC1 0x1f +#define RK817_CODEC_DADC_ALC2 0x20 +#define RK817_CODEC_DADC_NG 0x21 +#define RK817_CODEC_DADC_HPF 0x22 +#define RK817_CODEC_DADC_RVOLL 0x23 +#define RK817_CODEC_DADC_RVOLR 0x24 +#define RK817_CODEC_AMIC_CFG0 0x27 +#define RK817_CODEC_AMIC_CFG1 0x28 +#define RK817_CODEC_DMIC_PGA_GAIN 0x29 +#define RK817_CODEC_DMIC_LMT1 0x2a +#define RK817_CODEC_DMIC_LMT2 0x2b +#define RK817_CODEC_DMIC_NG1 0x2c +#define RK817_CODEC_DMIC_NG2 0x2d +#define RK817_CODEC_ADAC_CFG0 0x2e +#define RK817_CODEC_ADAC_CFG1 0x2f +#define RK817_CODEC_DDAC_POPD_DACST 0x30 +#define RK817_CODEC_DDAC_VOLL 0x31 +#define RK817_CODEC_DDAC_VOLR 0x32 +#define RK817_CODEC_DDAC_SR_LMT0 0x35 +#define RK817_CODEC_DDAC_LMT1 0x36 +#define RK817_CODEC_DDAC_LMT2 0x37 +#define RK817_CODEC_DDAC_MUTE_MIXCTL 0x38 +#define RK817_CODEC_DDAC_RVOLL 0x39 +#define RK817_CODEC_DDAC_RVOLR 0x3a +#define RK817_CODEC_AHP_ANTI0 0x3b +#define RK817_CODEC_AHP_ANTI1 0x3c +#define RK817_CODEC_AHP_CFG0 0x3d +#define RK817_CODEC_AHP_CFG1 0x3e +#define RK817_CODEC_AHP_CP 0x3f +#define RK817_CODEC_ACLASSD_CFG1 0x40 +#define RK817_CODEC_ACLASSD_CFG2 0x41 +#define RK817_CODEC_APLL_CFG0 0x42 +#define RK817_CODEC_APLL_CFG1 0x43 +#define RK817_CODEC_APLL_CFG2 0x44 +#define RK817_CODEC_APLL_CFG3 0x45 +#define RK817_CODEC_APLL_CFG4 0x46 +#define RK817_CODEC_APLL_CFG5 0x47 +#define RK817_CODEC_DI2S_CKM 0x48 +#define RK817_CODEC_DI2S_RSD 0x49 +#define RK817_CODEC_DI2S_RXCR1 0x4a +#define RK817_CODEC_DI2S_RXCR2 0x4b +#define RK817_CODEC_DI2S_RXCMD_TSD 0x4c +#define RK817_CODEC_DI2S_TXCR1 0x4d +#define RK817_CODEC_DI2S_TXCR2 0x4e +#define RK817_CODEC_DI2S_TXCR3_TXCMD 0x4f + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + #define RK817_POWER_EN_REG(i) (0xb1 + (i)) #define RK817_POWER_SLP_EN_REG(i) (0xb5 + (i)) -- 2.25.1

5 10

[PATCH v2] ASoC: rt711-sdca: change capture switch controls
by shumingf＠realtek.com 21 Apr '21

21 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The DAPM event and mixer control could mute/unmute the capture directly. That will be confused that capture still works if the user settings is unmute before the capture. Therefore, this patch uses the variables to record the capture switch status of DAPM and mixer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 125 ++++++++++++++++++++++------------ sound/soc/codecs/rt711-sdca.h | 2 + 2 files changed, 84 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..95ca0d74bd10 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -642,6 +642,73 @@ static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static void rt711_sdca_set_fu0f_capture_ctl(struct rt711_sdca_priv *rt711) +{ + if (rt711->fu0f_dapm_mute || rt711->fu0f_mixer_mute) { + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + } else { + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x00); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x00); + } +} + +static void rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) +{ + if (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_mute) { + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x01); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x01); + } else { + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), 0x00); + regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), 0x00); + } +} + +static int rt711_sdca_capture_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + if (strstr(ucontrol->id.name, "FU1E Capture Switch")) + ucontrol->value.integer.value[0] = !rt711->fu1e_mixer_mute; + else if (strstr(ucontrol->id.name, "FU0F Capture Switch")) + ucontrol->value.integer.value[0] = !rt711->fu0f_mixer_mute; + return 0; +} + +static int rt711_sdca_capture_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + if (strstr(ucontrol->id.name, "FU1E Capture Switch")) { + rt711->fu1e_mixer_mute = !ucontrol->value.integer.value[0]; + rt711_sdca_set_fu1e_capture_ctl(rt711); + } else if (strstr(ucontrol->id.name, "FU0F Capture Switch")) { + rt711->fu0f_mixer_mute = !ucontrol->value.integer.value[0]; + rt711_sdca_set_fu0f_capture_ctl(rt711); + } + return 0; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -652,14 +719,10 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), + SOC_SINGLE_EXT("FU1E Capture Switch", SND_SOC_NOPM, 0, 1, 0, + rt711_sdca_capture_switch_get, rt711_sdca_capture_switch_put), + SOC_SINGLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 0, + rt711_sdca_capture_switch_get, rt711_sdca_capture_switch_put), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), @@ -809,28 +872,15 @@ static int rt711_sdca_fu0f_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu0f_dapm_mute = false; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); + rt711->fu0f_dapm_mute = true; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; } return 0; @@ -842,29 +892,16 @@ static int rt711_sdca_fu1e_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu1e_dapm_mute = false; + rt711_sdca_set_fu1e_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); - break; + rt711->fu1e_dapm_mute = true; + rt711_sdca_set_fu1e_capture_ctl(rt711); + break; } return 0; } @@ -1330,6 +1367,8 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, */ rt711->hw_init = false; rt711->first_hw_init = false; + rt711->fu0f_dapm_mute = rt711->fu0f_mixer_mute = true; + rt711->fu1e_dapm_mute = rt711->fu1e_mixer_mute = true; /* JD source uses JD2 in default */ rt711->jd_src = RT711_JD2; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 98a022cec0bd..a3b3a95094c0 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -30,6 +30,8 @@ struct rt711_sdca_priv { int jack_type, jd_src; unsigned int scp_sdca_stat1, scp_sdca_stat2; int hw_ver; + bool fu0f_dapm_mute, fu0f_mixer_mute; + bool fu1e_dapm_mute, fu1e_mixer_mute; }; struct sdw_stream_data { -- 2.29.0

3 2

[PATCH] ASoC: rt715-sdca: fix clock stop prepare timeout issue
by Jack Yu 21 Apr '21

21 Apr '21

The Clock_Stop_Not_Finished bit is currently not cleared during the clock stop sequence,which results in the clock stop sequence timing out after 300ms. This is root-caused to a missing register configuration impacting the transition between the bus clock and an internal clock. This problem was exposed by cleanups in the SoundWire bus code, where previously the ClockStopPrepare was mistakenly skipped when using ClockStopMode0. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715-sdca-sdw.c | 1 + sound/soc/codecs/rt715-sdca-sdw.h | 1 + sound/soc/codecs/rt715-sdca.c | 3 +++ sound/soc/codecs/rt715-sdca.h | 1 + 4 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 1350798406f0..47a8f04d7100 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -70,6 +70,7 @@ static bool rt715_sdca_mbq_readable_register(struct device *dev, unsigned int re case 0x2000036: case 0x2000037: case 0x2000039: + case 0x2000044: case 0x6100000: return true; default: diff --git a/sound/soc/codecs/rt715-sdca-sdw.h b/sound/soc/codecs/rt715-sdca-sdw.h index cd365bb60747..0cbc14844f8c 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.h +++ b/sound/soc/codecs/rt715-sdca-sdw.h @@ -113,6 +113,7 @@ static const struct reg_default rt715_mbq_reg_defaults_sdca[] = { { 0x2000036, 0x0000 }, { 0x2000037, 0x0000 }, { 0x2000039, 0xaa81 }, + { 0x2000044, 0x0202 }, { 0x6100000, 0x0100 }, { SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_FU_ADC8_9_VOL, RT715_SDCA_FU_VOL_CTRL, CH_01), 0x00 }, diff --git a/sound/soc/codecs/rt715-sdca.c b/sound/soc/codecs/rt715-sdca.c index 7db76c19e048..936e3061ca1e 100644 --- a/sound/soc/codecs/rt715-sdca.c +++ b/sound/soc/codecs/rt715-sdca.c @@ -1054,6 +1054,9 @@ int rt715_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, RT715_REV_1, 0x40, 0x40); } + /* DFLL Calibration trigger */ + rt715_sdca_index_update_bits(rt715, RT715_VENDOR_REG, + RT715_DFLL_VAD, 0x1, 0x1); /* trigger mode = VAD enable */ regmap_write(rt715->regmap, SDW_SDCA_CTL(FUN_MIC_ARRAY, RT715_SDCA_SMPU_TRIG_ST_EN, diff --git a/sound/soc/codecs/rt715-sdca.h b/sound/soc/codecs/rt715-sdca.h index 85ce4d95e5eb..1c7fbff2ce1e 100644 --- a/sound/soc/codecs/rt715-sdca.h +++ b/sound/soc/codecs/rt715-sdca.h @@ -81,6 +81,7 @@ struct rt715_sdca_kcontrol_private { #define RT715_AD_FUNC_EN 0x36 #define RT715_REV_1 0x37 #define RT715_SDW_INPUT_SEL 0x39 +#define RT715_DFLL_VAD 0x44 #define RT715_EXT_DMIC_CLK_CTRL2 0x54 /* Index (NID:61h) */ -- 2.29.0

1 0

Question about soc_pcm_pointer()
by Kuninori Morimoto 21 Apr '21

21 Apr '21

Hi ALSA ML I noticed that current soc-pcm.c :: soc_pcm_pointer() is adding both CPU-DAI's max delay (= A) and Codec-DAI's max delay (= B). static snd_pcm_uframes_t soc_pcm_pointer(...) { ... ^ for_each_rtd_cpu_dais(rtd, i, cpu_dai) (A) cpu_delay = max(cpu_delay, ...); v delay += cpu_delay; ^ for_each_rtd_codec_dais(rtd, i, codec_dai) (B) codec_delay = max(codec_delay, ...); v delay += codec_delay; runtime->delay = delay; ... } But I guess All-DAI's max delay is very enough for it. What do you think ? Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[PATCH] ASoC: ak4458: enable daisy chain
by Shengjiu Wang 20 Apr '21

20 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> Enable Daisy Chain if in TDM mode and the number of played channels is bigger than the maximum supported number of channels. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 47 ++++++++++++++++++++++++++------------- sound/soc/codecs/ak4458.h | 1 + 2 files changed, 32 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 3088128816bb..29eb78702bf3 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -306,6 +306,20 @@ static const struct snd_soc_dapm_route ak4497_intercon[] = { }; +static int ak4458_get_tdm_mode(struct ak4458_priv *ak4458) +{ + switch (ak4458->slots * ak4458->slot_width) { + case 128: + return 1; + case 256: + return 2; + case 512: + return 3; + default: + return 0; + } +} + static int ak4458_rstn_control(struct snd_soc_component *component, int bit) { int ret; @@ -333,13 +347,16 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - u8 format, dsdsel0, dsdsel1; - int nfs1, dsd_bclk, ret; + u8 format, dsdsel0, dsdsel1, dchn; + int nfs1, dsd_bclk, ret, channels, channels_max; nfs1 = params_rate(params); ak4458->fs = nfs1; /* calculate bit clock */ + channels = params_channels(params); + channels_max = dai->driver->playback.channels_max; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: @@ -419,6 +436,17 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_00_CONTROL1, AK4458_DIF_MASK, format); + /* + * Enable/disable Daisy Chain if in TDM mode and the number of played + * channels is bigger than the maximum supported number of channels + */ + dchn = ak4458_get_tdm_mode(ak4458) && + (ak4458->fmt == SND_SOC_DAIFMT_DSP_B) && + (channels > channels_max) ? AK4458_DCHAIN_MASK : 0; + + snd_soc_component_update_bits(component, AK4458_0B_CONTROL7, + AK4458_DCHAIN_MASK, dchn); + ret = ak4458_rstn_control(component, 0); if (ret) return ret; @@ -519,20 +547,7 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, ak4458->slots = slots; ak4458->slot_width = slot_width; - switch (slots * slot_width) { - case 128: - mode = AK4458_MODE_TDM128; - break; - case 256: - mode = AK4458_MODE_TDM256; - break; - case 512: - mode = AK4458_MODE_TDM512; - break; - default: - mode = AK4458_MODE_NORMAL; - break; - } + mode = ak4458_get_tdm_mode(ak4458) << AK4458_MODE_SHIFT; snd_soc_component_update_bits(component, AK4458_0A_CONTROL6, AK4458_MODE_MASK, diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index 9548c5d78621..9ad869575f8d 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -82,6 +82,7 @@ * */ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) +#define AK4458_DCHAIN_MASK (0x1 << 1) #define AK4458_DSDSEL_MASK (0x1 << 0) #define AK4458_DP_MASK (0x1 << 7) -- 2.27.0

2 1

[PATCH] ASoC: simple-card-utils: Allocate link info structure on heap
by Thierry Reding 20 Apr '21

20 Apr '21

From: Thierry Reding <treding(a)nvidia.com> struct link_info can grow fairly large and may cause the stack frame size to be exceeded when allocated on the stack. Some architectures such as 32-bit ARM, RISC-V or PowerPC have small stack frames where this causes a compiler warning, so allocate these structures on the heap instead of the stack. Fixes: 343e55e71877 ("ASoC: simple-card-utils: Increase maximum number of links to 128") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/generic/audio-graph-card.c | 18 +++++++++++------- sound/soc/generic/simple-card.c | 23 +++++++++++++---------- 2 files changed, 24 insertions(+), 17 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index c7369beee805..e45a560aa9b0 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -548,21 +548,24 @@ static int graph_get_dais_count(struct asoc_simple_priv *priv, int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) { struct snd_soc_card *card = simple_priv_to_card(priv); - struct link_info li; + struct link_info *li; int ret; + li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); + if (!li) + return -ENOMEM; + card->owner = THIS_MODULE; card->dev = dev; - memset(&li, 0, sizeof(li)); - ret = graph_get_dais_count(priv, &li); + ret = graph_get_dais_count(priv, li); if (ret < 0) return ret; - if (!li.link) + if (!li->link) return -EINVAL; - ret = asoc_simple_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, li); if (ret < 0) return ret; @@ -581,8 +584,8 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) if (ret < 0) return ret; - memset(&li, 0, sizeof(li)); - ret = graph_for_each_link(priv, &li, + memset(li, 0, sizeof(*li)); + ret = graph_for_each_link(priv, li, graph_dai_link_of, graph_dai_link_of_dpcm); if (ret < 0) @@ -600,6 +603,7 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) if (ret < 0) goto err; + devm_kfree(dev, li); return 0; err: diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8b9964d25757..ca27cb9ff9e1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -442,10 +442,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, return ret; } -static int simple_parse_of(struct asoc_simple_priv *priv) +static int simple_parse_of(struct asoc_simple_priv *priv, struct link_info *li) { struct snd_soc_card *card = simple_priv_to_card(priv); - struct link_info li; int ret; ret = asoc_simple_parse_widgets(card, PREFIX); @@ -461,8 +460,8 @@ static int simple_parse_of(struct asoc_simple_priv *priv) return ret; /* Single/Muti DAI link(s) & New style of DT node */ - memset(&li, 0, sizeof(li)); - ret = simple_for_each_link(priv, &li, + memset(li, 0, sizeof(*li)); + ret = simple_for_each_link(priv, li, simple_dai_link_of, simple_dai_link_of_dpcm); if (ret < 0) @@ -612,7 +611,7 @@ static int asoc_simple_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; struct snd_soc_card *card; - struct link_info li; + struct link_info *li; int ret; /* Allocate the private data and the DAI link array */ @@ -625,21 +624,24 @@ static int asoc_simple_probe(struct platform_device *pdev) card->dev = dev; card->probe = simple_soc_probe; - memset(&li, 0, sizeof(li)); - ret = simple_get_dais_count(priv, &li); + li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); + if (!li) + return -ENOMEM; + + ret = simple_get_dais_count(priv, li); if (ret < 0) return ret; - if (!li.link) + if (!li->link) return -EINVAL; - ret = asoc_simple_init_priv(priv, &li); + ret = asoc_simple_init_priv(priv, li); if (ret < 0) return ret; if (np && of_device_is_available(np)) { - ret = simple_parse_of(priv); + ret = simple_parse_of(priv, li); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); @@ -698,6 +700,7 @@ static int asoc_simple_probe(struct platform_device *pdev) if (ret < 0) goto err; + devm_kfree(dev, li); return 0; err: asoc_simple_clean_reference(card); -- 2.30.2

3 2

[PATCH] ASoC: rt1015p: add support on format S32_LE
by Jack Yu 20 Apr '21

20 Apr '21

Add support on format S32_LE for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 39f4c4ed2ab8..40f2063aefbe 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -96,7 +96,8 @@ static struct snd_soc_dai_driver rt1015p_dai_driver = { .name = "HiFi", .playback = { .stream_name = "HiFi Playback", - .formats = SNDRV_PCM_FMTBIT_S24, + .formats = SNDRV_PCM_FMTBIT_S24 | + SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_48000, .channels_min = 1, .channels_max = 2, -- 2.29.0

2 1

[PATCH] ALSA: usb: midi: don't return -ENOMEM when usb_urb_ep_type_check fails
by Colin King 20 Apr '21

20 Apr '21

From: Colin Ian King <colin.king(a)canonical.com> Currently when the call to usb_urb_ep_type_check fails (returning -EINVAL) the error return path returns -ENOMEM via the exit label "error". Other uses of the same error exit label set the err variable to -ENOMEM but this is not being used. I believe the original intent was for the error exit path to return the value in err rather than the hard coded -ENOMEM, so return this rather than the hard coded -ENOMEM. Addresses-Coverity: ("Unused value") Fixes: 738d9edcfd44 ("ALSA: usb-audio: Add sanity checks for invalid EPs") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 9efda4b06acb..a10ac75969a8 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1316,7 +1316,7 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi *umidi, error: snd_usbmidi_in_endpoint_delete(ep); - return -ENOMEM; + return err; } /* -- 2.30.2

2 1

[v6 3/3] dt-bindings: Add Rockchip rk817 audio CODEC support
by Chris Morgan 20 Apr '21

20 Apr '21

Create dt-binding documentation to document rk817 codec. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. .../bindings/sound/rockchip,rk817-codec.yaml | 57 +++++++++++++++++++ 1 file changed, 57 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..0059ef54d6b5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + + "#sound-dai-cells": + const: 1 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>, <&i2s_8ch_mclk>; + #sound-dai-cells = <0>; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + + ........ + + rk817_codec: codec { + mic-in-differential; + status = "okay"; + }; + + ........ + +}; -- 2.25.1

2 2

[v6 1/3] mfd: Add Rockchip rk817 audio CODEC support
by Chris Morgan 20 Apr '21

20 Apr '21

Add rk817 codec support cell to rk808 mfd driver. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. drivers/mfd/rk808.c | 89 +++++++++++++++++++++++++++++++++++++++ include/linux/mfd/rk808.h | 81 +++++++++++++++++++++++++++++++++++ 2 files changed, 170 insertions(+) diff --git a/drivers/mfd/rk808.c b/drivers/mfd/rk808.c index ad923dd4e007..b92f4c6f503e 100644 --- a/drivers/mfd/rk808.c +++ b/drivers/mfd/rk808.c @@ -65,6 +65,9 @@ static bool rk817_is_volatile_reg(struct device *dev, unsigned int reg) switch (reg) { case RK817_SECONDS_REG ... RK817_WEEKS_REG: case RK817_RTC_STATUS_REG: +#ifdef CONFIG_SND_SOC_RK817 + case RK817_CODEC_DTOP_LPT_SRST: +#endif case RK817_INT_STS_REG0: case RK817_INT_STS_REG1: case RK817_INT_STS_REG2: @@ -163,6 +166,11 @@ static const struct mfd_cell rk817s[] = { .num_resources = ARRAY_SIZE(rk817_rtc_resources), .resources = &rk817_rtc_resources[0], }, +#ifdef CONFIG_SND_SOC_RK817 + { + .name = "rk817-codec", + }, +#endif }; static const struct mfd_cell rk818s[] = { @@ -201,6 +209,87 @@ static const struct rk808_reg_data rk808_pre_init_reg[] = { static const struct rk808_reg_data rk817_pre_init_reg[] = { {RK817_RTC_CTRL_REG, RTC_STOP, RTC_STOP}, +#ifdef CONFIG_SND_SOC_RK817 + /* Codec specific registers */ + { RK817_CODEC_DTOP_VUCTL, MASK_ALL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, MASK_ALL, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, MASK_ALL, 0x06 }, + { RK817_CODEC_AADC_CFG0, MASK_ALL, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC1, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_ALC2, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_NG, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_HPF, MASK_ALL, 0x00 }, + { RK817_CODEC_DADC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DADC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AMIC_CFG0, MASK_ALL, 0x70 }, + { RK817_CODEC_AMIC_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, MASK_ALL, 0x66 }, + { RK817_CODEC_DMIC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG1, MASK_ALL, 0x00 }, + { RK817_CODEC_DMIC_NG2, MASK_ALL, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, MASK_ALL, 0x00 }, + { RK817_CODEC_ADAC_CFG1, MASK_ALL, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, MASK_ALL, 0x82 }, + { RK817_CODEC_DDAC_VOLL, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT1, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_LMT2, MASK_ALL, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, MASK_ALL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, MASK_ALL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, MASK_ALL, 0xff }, + { RK817_CODEC_AHP_ANTI0, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_ANTI1, MASK_ALL, 0x00 }, + { RK817_CODEC_AHP_CFG0, MASK_ALL, 0xe0 }, + { RK817_CODEC_AHP_CFG1, MASK_ALL, 0x1f }, + { RK817_CODEC_AHP_CP, MASK_ALL, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, MASK_ALL, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, MASK_ALL, 0x44 }, + { RK817_CODEC_APLL_CFG0, MASK_ALL, 0x04 }, + { RK817_CODEC_APLL_CFG1, MASK_ALL, 0x00 }, + { RK817_CODEC_APLL_CFG2, MASK_ALL, 0x30 }, + { RK817_CODEC_APLL_CFG3, MASK_ALL, 0x19 }, + { RK817_CODEC_APLL_CFG4, MASK_ALL, 0x65 }, + { RK817_CODEC_APLL_CFG5, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_CKM, MASK_ALL, 0x01 }, + { RK817_CODEC_DI2S_RSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, MASK_ALL, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, MASK_ALL, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, MASK_ALL, 0x00 }, +#endif {RK817_GPIO_INT_CFG, RK817_INT_POL_MSK, RK817_INT_POL_L}, {RK817_SYS_CFG(1), RK817_HOTDIE_TEMP_MSK | RK817_TSD_TEMP_MSK, RK817_HOTDIE_105 | RK817_TSD_140}, diff --git a/include/linux/mfd/rk808.h b/include/linux/mfd/rk808.h index e07f6e61cd38..a96e6d43ca06 100644 --- a/include/linux/mfd/rk808.h +++ b/include/linux/mfd/rk808.h @@ -437,6 +437,87 @@ enum rk809_reg_id { #define RK817_RTC_COMP_LSB_REG 0x10 #define RK817_RTC_COMP_MSB_REG 0x11 +/* RK817 Codec Registers */ +#define RK817_CODEC_DTOP_VUCTL 0x12 +#define RK817_CODEC_DTOP_VUCTIME 0x13 +#define RK817_CODEC_DTOP_LPT_SRST 0x14 +#define RK817_CODEC_DTOP_DIGEN_CLKE 0x15 +#define RK817_CODEC_AREF_RTCFG0 0x16 +#define RK817_CODEC_AREF_RTCFG1 0x17 +#define RK817_CODEC_AADC_CFG0 0x18 +#define RK817_CODEC_AADC_CFG1 0x19 +#define RK817_CODEC_DADC_VOLL 0x1a +#define RK817_CODEC_DADC_VOLR 0x1b +#define RK817_CODEC_DADC_SR_ACL0 0x1e +#define RK817_CODEC_DADC_ALC1 0x1f +#define RK817_CODEC_DADC_ALC2 0x20 +#define RK817_CODEC_DADC_NG 0x21 +#define RK817_CODEC_DADC_HPF 0x22 +#define RK817_CODEC_DADC_RVOLL 0x23 +#define RK817_CODEC_DADC_RVOLR 0x24 +#define RK817_CODEC_AMIC_CFG0 0x27 +#define RK817_CODEC_AMIC_CFG1 0x28 +#define RK817_CODEC_DMIC_PGA_GAIN 0x29 +#define RK817_CODEC_DMIC_LMT1 0x2a +#define RK817_CODEC_DMIC_LMT2 0x2b +#define RK817_CODEC_DMIC_NG1 0x2c +#define RK817_CODEC_DMIC_NG2 0x2d +#define RK817_CODEC_ADAC_CFG0 0x2e +#define RK817_CODEC_ADAC_CFG1 0x2f +#define RK817_CODEC_DDAC_POPD_DACST 0x30 +#define RK817_CODEC_DDAC_VOLL 0x31 +#define RK817_CODEC_DDAC_VOLR 0x32 +#define RK817_CODEC_DDAC_SR_LMT0 0x35 +#define RK817_CODEC_DDAC_LMT1 0x36 +#define RK817_CODEC_DDAC_LMT2 0x37 +#define RK817_CODEC_DDAC_MUTE_MIXCTL 0x38 +#define RK817_CODEC_DDAC_RVOLL 0x39 +#define RK817_CODEC_DDAC_RVOLR 0x3a +#define RK817_CODEC_AHP_ANTI0 0x3b +#define RK817_CODEC_AHP_ANTI1 0x3c +#define RK817_CODEC_AHP_CFG0 0x3d +#define RK817_CODEC_AHP_CFG1 0x3e +#define RK817_CODEC_AHP_CP 0x3f +#define RK817_CODEC_ACLASSD_CFG1 0x40 +#define RK817_CODEC_ACLASSD_CFG2 0x41 +#define RK817_CODEC_APLL_CFG0 0x42 +#define RK817_CODEC_APLL_CFG1 0x43 +#define RK817_CODEC_APLL_CFG2 0x44 +#define RK817_CODEC_APLL_CFG3 0x45 +#define RK817_CODEC_APLL_CFG4 0x46 +#define RK817_CODEC_APLL_CFG5 0x47 +#define RK817_CODEC_DI2S_CKM 0x48 +#define RK817_CODEC_DI2S_RSD 0x49 +#define RK817_CODEC_DI2S_RXCR1 0x4a +#define RK817_CODEC_DI2S_RXCR2 0x4b +#define RK817_CODEC_DI2S_RXCMD_TSD 0x4c +#define RK817_CODEC_DI2S_TXCR1 0x4d +#define RK817_CODEC_DI2S_TXCR2 0x4e +#define RK817_CODEC_DI2S_TXCR3_TXCMD 0x4f + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + #define RK817_POWER_EN_REG(i) (0xb1 + (i)) #define RK817_POWER_SLP_EN_REG(i) (0xb5 + (i)) -- 2.25.1

2 3

[PATCH] ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook 845 G8
by Kai-Heng Feng 20 Apr '21

20 Apr '21

On HP EliteBook 845 G8, the audio LEDs can be enabled by ALC285_FIXUP_HP_MUTE_LED. So use it accordingly. In addition to that, the mic captures lots of noises, so also limits the mic boost. The quality of capture audio becomes crystal clear after limiting the mic boost. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7544b77d3f7..c71f1f2b6a8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6427,6 +6427,7 @@ enum { ALC282_FIXUP_ACER_DISABLE_LINEOUT, ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST, ALC256_FIXUP_ACER_HEADSET_MIC, + ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -7901,6 +7902,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC285_FIXUP_HP_MUTE_LED, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8079,6 +8086,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.30.2

2 1

[PATCH v6] sound: rawmidi: Add framing mode
by David Henningsson 20 Apr '21

20 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Changes since v5: Added realtime clock and changed params struct according to Jaroslav's wishes. This version of the patch has been compile tested only. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 30 ++++++++++++- sound/core/rawmidi.c | 86 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 4 +- 4 files changed, 118 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..989e1517332d 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + unsigned int framing; /* whether to frame input data */ + unsigned int clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..773a00c0a1d8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,38 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +#define SNDRV_RAWMIDI_MODE_FRAMING_MASK (7<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_SHIFT 0 +#define SNDRV_RAWMIDI_MODE_FRAMING_NONE (0<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP (1<<0) +#define SNDRV_RAWMIDI_MODE_CLOCK_MASK (7<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_SHIFT 3 +#define SNDRV_RAWMIDI_MODE_CLOCK_NONE (0<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_REALTIME (1<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC (2<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW (3<<3) + +#define SNDRV_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 data[SNDRV_RAWMIDI_FRAMING_DATA_LENGTH]; +} __attribute__((packed)); + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned int mode; /* For input data only, frame incoming data */ + unsigned char reserved[12]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..5d5f4363e887 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -680,9 +680,12 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, bool is_input) { char *newbuf, *oldbuf; + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +723,18 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; + unsigned int clock_type = params->mode & SNDRV_RAWMIDI_MODE_CLOCK_MASK; + + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) + return -EINVAL; + else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) + return -EINVAL; + if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) + return -EINVAL; snd_rawmidi_drain_input(substream); + substream->framing = framing; + substream->clock_type = clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +977,61 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SNDRV_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SNDRV_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SNDRV_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + +static struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + switch (substream->clock_type) { + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW: + ktime_get_raw_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC: + ktime_get_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_REALTIME: + ktime_get_real_ts64(&ts64); + break; + } + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1046,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1057,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1534,6 +1607,7 @@ static __poll_t snd_rawmidi_poll(struct file *file, poll_table *wait) /* */ + static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -1541,6 +1615,8 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; unsigned long buffer_size, avail, xruns; + unsigned int clock_type; + static const char *clock_names[4] = { "none", "realtime", "monotonic", "monotonic raw" }; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1596,6 +1672,14 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; + if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + clock_names[clock_type]); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..68a93443583c 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,8 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned int mode; + unsigned char reserved[12]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +26,7 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.mode, &src->mode) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

3 2

[PATCH] ALSA: ctxfi: Avoid writing uninitialized variable in uaa_to_xfi()
by Yizhuo Zhai 20 Apr '21

20 Apr '21

Inside the function uaa_to_xfi(), variable "l_timer" could be uninitialized if pci_bus_read_config_dword() returns PCIBIOS_BAD_REGISTER_NUMBER. However, it is write back to PCI config space via pci_write_config_dword(), which is potentially unsafe. Signed-off-by: Yizhuo <yzhai003(a)ucr.edu> --- sound/pci/ctxfi/cthw20k1.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 0cea4982ed7d..7c3436499974 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1789,7 +1789,7 @@ static struct capabilities hw_capabilities(struct hw *hw) static int uaa_to_xfi(struct pci_dev *pci) { unsigned int bar0, bar1, bar2, bar3, bar4, bar5; - unsigned int cmd, irq, cl_size, l_timer, pwr; + unsigned int cmd, irq, cl_size, l_timer = ~0, pwr; unsigned int is_uaa; unsigned int data[4] = {0}; unsigned int io_base; -- 2.31.1 -- Kind Regards, *Yizhuo Zhai* *Computer Science, Graduate Student* *University of California, Riverside *

2 1

Headset MIC issue on HP ThinClient
by Kailang 20 Apr '21

20 Apr '21

Hi Takashi, Some HP machines, boot with plugged headset, the headset MIC will lose. Attach patch will solve it. BR, Kailang

2 1

[PATCH] ASoC: rt711-sdca: remove capture switch controls
by shumingf＠realtek.com 20 Apr '21

20 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The settings of the switch control already set by DAPM event. These switch controls might the user confused why it can't disable the capture. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..2a09c305f4e4 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -652,14 +652,6 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), -- 2.29.0

5 10

[PATCH 1/2] ASoC: simple-card-utils: Propagate errors on too many links
by Thierry Reding 20 Apr '21

20 Apr '21

From: Thierry Reding <treding(a)nvidia.com> The DAI counting code doesn't propagate errors when the number of maximum links is exceeded, which causes subsequent initialization code to continue to run and that eventually leads to memory corruption with the code trying to access memory that is out of bounds. Fix this by propgating errors when the maximum number of links is reached, which ensures that the driver fails to load and prevents the memory corruption. Fixes: f2138aed231c ("ASoC: simple-card-utils: enable flexible CPU/Codec/Platform") Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/soc/generic/audio-graph-card.c | 19 +++++++++++-------- sound/soc/generic/simple-card.c | 17 ++++++++++------- 2 files changed, 21 insertions(+), 15 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0582fe296471..2401212281c2 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -539,8 +539,8 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, return ret; } -static void graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li); +static int graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li); int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) { @@ -552,7 +552,10 @@ int audio_graph_parse_of(struct asoc_simple_priv *priv, struct device *dev) card->dev = dev; memset(&li, 0, sizeof(li)); - graph_get_dais_count(priv, &li); + ret = graph_get_dais_count(priv, &li); + if (ret < 0) + return ret; + if (!li.link) return -EINVAL; @@ -657,8 +660,8 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, return 0; } -static void graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li) +static int graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li) { /* * link_num : number of links. @@ -706,9 +709,9 @@ static void graph_get_dais_count(struct asoc_simple_priv *priv, * => 4 DAIs = 2xCPU + 2xCodec * => 1 ccnf = 1xdummy-Codec */ - graph_for_each_link(priv, li, - graph_count_noml, - graph_count_dpcm); + return graph_for_each_link(priv, li, + graph_count_noml, + graph_count_dpcm); } int audio_graph_card_probe(struct snd_soc_card *card) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index bf5ddf1ea65f..1d1c4309b582 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -526,8 +526,8 @@ static int simple_count_dpcm(struct asoc_simple_priv *priv, return 0; } -static void simple_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li) +static int simple_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); struct device_node *top = dev->of_node; @@ -584,12 +584,12 @@ static void simple_get_dais_count(struct asoc_simple_priv *priv, li->num[0].platforms = 1; li->link = 1; - return; + return 0; } - simple_for_each_link(priv, li, - simple_count_noml, - simple_count_dpcm); + return simple_for_each_link(priv, li, + simple_count_noml, + simple_count_dpcm); } static int simple_soc_probe(struct snd_soc_card *card) @@ -628,7 +628,10 @@ static int asoc_simple_probe(struct platform_device *pdev) card->probe = simple_soc_probe; memset(&li, 0, sizeof(li)); - simple_get_dais_count(priv, &li); + ret = simple_get_dais_count(priv, &li); + if (ret < 0) + return ret; + if (!li.link) return -EINVAL; -- 2.30.2

4 9

[PATCH v2] ALSA: usb-audio: Re-apply implicit feedback mode to Pioneer devices
by Takashi Iwai 20 Apr '21

20 Apr '21

Pioneer devices are supposed to be working with the implicit feedback mode, but so far the attempt to apply the implicit feedback caused issues, hence we explicitly skipped the implicit feedback mode for them. Recently, Geraldo discovered that the device actually works if you skip the generic matching of the sync EPs for the capture stream. That is, we should apply the implicit feedback setup for the playback like other similar devices, while we need to return 1 from audioformat_capture_quirk() so that no further matching will be done. And, later on, Olivia reported later that the fiddling with the capture quirk alone doesn't suffice for the test with speaker-test program. This seems to be a similar case like the recently fixed BOSS devices. Indeed, the problem could be addressed by setting playback_first flag, which indicates that the playback URBs have to be sent out at first even in the implicit feedback mode. This patch implements the application of the implicit feedback to Pioneer devices as described in the above. The former skip_pioneer_sync_ep() was dropped, and instead we provide is_pioneer_implicit_fb() to check the Pioneer devices that need the implicit feedback. In the audioformat_implicit_fb_quirk(), simply apply the implicit fb for playback and set chip->playback_first flag if matching, and in audioformat_capture_quirk()(), it returns 1 for skipping the generic EP sync handling. Reported-by: Geraldo <geraldogabriel(a)gmail.com> Tested-by: Olivia Mackintosh <livvy(a)base.nu> Link: https://lore.kernel.org/r/s5ha6pygqfz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/implicit.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 428e3a449f5d..77ffcea294dd 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -230,18 +230,27 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip, ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Playback and capture EPs on Pioneer devices share the same iface/altset + * for the implicit feedback operation */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) +static bool is_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct usb_host_interface *alts) + { struct usb_endpoint_descriptor *epd; + if (USB_ID_VENDOR(chip->usb_id) != 0x2b73 && + USB_ID_VENDOR(chip->usb_id) != 0x08e4) + return false; + if (alts->desc.bInterfaceClass != USB_CLASS_VENDOR_SPEC) + return false; if (alts->desc.bNumEndpoints != 2) - return 0; + return false; + + epd = get_endpoint(alts, 0); + if (!usb_endpoint_is_isoc_out(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC) + return false; epd = get_endpoint(alts, 1); if (!usb_endpoint_is_isoc_in(epd) || @@ -250,8 +259,9 @@ static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, USB_ENDPOINT_USAGE_DATA && (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ + return false; + + return true; } static int __add_generic_implicit_fb(struct snd_usb_audio *chip, @@ -367,12 +377,12 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } /* Pioneer devices with vendor spec class */ - if (attr == USB_ENDPOINT_SYNC_ASYNC && - alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && - (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ - USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) - return 1; + if (is_pioneer_implicit_fb(chip, alts)) { + chip->playback_first = 1; + return add_implicit_fb_sync_ep(chip, fmt, + get_endpoint(alts, 1)->bEndpointAddress, + 1, alts->desc.bInterfaceNumber, + alts); } /* Try the generic implicit fb if available */ @@ -394,6 +404,8 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, if (p && (p->type == IMPLICIT_FB_FIXED || p->type == IMPLICIT_FB_BOTH)) return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); + if (is_pioneer_implicit_fb(chip, alts)) + return 1; /* skip the quirk, also don't handle generic sync EP */ return 0; } -- 2.26.2

2 1

[PATCH 0/5] ASoC: rt286/rt298: Fixes for DMIC2 config and combo jack
by David Ward 19 Apr '21

19 Apr '21

The last two patches in this series fix a longstanding issue that prevented the ALC3263 codec from using a headset mic. This codec can be found on Dell systems including the Latitude 13 7350, Venue 11 Pro 7140, and XPS 13 9343. In fact, there is an ACPI quirk for the XPS 13 9343, which forces it to use legacy HD Audio just to avoid this issue: https://lore.kernel.org/alsa-devel/CAPeXnHv07HkvcHrYFmZMr8OTp7U7F=k_k=LPYnU… This may allow that ACPI quirk to be removed. Either way, the other systems mentioned above do not support this quirk and already use the ASoC driver, so this fix is necessary for headset mic support on those systems. Note: there is likely other handling for this codec that only exists in the HDA driver, but which also belongs in the ASoC driver. Commit 394c97f824fa ("ALSA: hda/realtek - Change EAPD to verb control") describes an issue that does not seem to be resolved in the ASoC driver, to give an example. Other patches in this series are not specific to the ALC3263. These patches set the correct combo jack configuration when headphones are inserted, and fix a misaligned value set in the DMIC2 Configuration Default register. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=114171 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> David Ward (5): ASoC: rt286: Fix upper byte in DMIC2 configuration ASoC: rt286: Configure combo jack for headphones ASoC: rt298: Configure combo jack for headphones ASoC: rt286: Make RT286_SET_GPIO_* readable and writable ASoC: rt286: Generalize support for ALC3263 codec sound/soc/codecs/rt286.c | 34 +++++++++++++++++++++------------- sound/soc/codecs/rt298.c | 9 +++++++-- 2 files changed, 28 insertions(+), 15 deletions(-) -- 2.31.1

3 7

[PATCH 0/5] ASoC: remove more cppcheck warnings
by Pierre-Louis Bossart 19 Apr '21

19 Apr '21

I missed those warnings in the previous rounds, they are mostly trivial and shouldn't change the behavior. Pierre-Louis Bossart (5): ASoC: soc-acpi: remove useless initialization ASoC: soc-core: fix signed/unsigned issue ASoC: soc-core: fix always-false condition ASoC: codecs: lpass-rx-macro: remove useless return ASoC: codecs: rt5682: clarify expression sound/soc/codecs/lpass-rx-macro.c | 2 -- sound/soc/codecs/rt5682.c | 2 +- sound/soc/soc-acpi.c | 2 +- sound/soc/soc-core.c | 7 +------ 4 files changed, 3 insertions(+), 10 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: sigmadsp: Disable cache mechanism for readbacks
by Niklas Carlsson 19 Apr '21

19 Apr '21

From: Niklas Carlsson <niklasc(a)axis.com> The ALSA control readback functionality only works for non-volatile controls, i.e. control values that does not change on their own without driver interaction. This doesn't work for readbacks since the DSP firmware updates the control value. Disable the cache mechanism in the driver if the control name matches the prefix used for readbacks to ensure that the control value is valid. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/sigmadsp.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 76c77dc8ecf7..b992216aee55 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -24,6 +24,8 @@ #define SIGMA_FW_CHUNK_TYPE_CONTROL 1 #define SIGMA_FW_CHUNK_TYPE_SAMPLERATES 2 +#define READBACK_CTRL_NAME "ReadBack" + struct sigmadsp_control { struct list_head head; uint32_t samplerates; @@ -31,6 +33,7 @@ struct sigmadsp_control { unsigned int num_bytes; const char *name; struct snd_kcontrol *kcontrol; + bool is_readback; bool cached; uint8_t cache[]; }; @@ -141,7 +144,8 @@ static int sigmadsp_ctrl_put(struct snd_kcontrol *kcontrol, if (ret == 0) { memcpy(ctrl->cache, data, ctrl->num_bytes); - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; } mutex_unlock(&sigmadsp->lock); @@ -164,7 +168,8 @@ static int sigmadsp_ctrl_get(struct snd_kcontrol *kcontrol, } if (ret == 0) { - ctrl->cached = true; + if (!ctrl->is_readback) + ctrl->cached = true; memcpy(ucontrol->value.bytes.data, ctrl->cache, ctrl->num_bytes); } @@ -231,6 +236,15 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, name[name_len] = '\0'; ctrl->name = name; + /* + * Readbacks doesn't work with non-volatile controls, since the + * firmware updates the control value without driver interaction. Mark + * the readbacks to ensure that the values are not cached. + */ + if (ctrl->name && strncmp(ctrl->name, READBACK_CTRL_NAME, + (sizeof(READBACK_CTRL_NAME) - 1)) == 0) + ctrl->is_readback = true; + ctrl->addr = le16_to_cpu(ctrl_chunk->addr); ctrl->num_bytes = num_bytes; ctrl->samplerates = le32_to_cpu(chunk->samplerates); -- 2.20.1

2 1

[PATCH v2] ASoC: ak4458: check reset control status
by Shengjiu Wang 19 Apr '21

19 Apr '21

From: Viorel Suman <viorel.suman(a)nxp.com> check the return value of ak4458_rstn_control. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - fix "use of undeclared identifier 'ret'" reported by kernel test robot sound/soc/codecs/ak4458.c | 21 ++++++++++++++++----- 1 file changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 85a1d00894a9..3088128816bb 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -334,7 +334,7 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); u8 format, dsdsel0, dsdsel1; - int nfs1, dsd_bclk; + int nfs1, dsd_bclk, ret; nfs1 = params_rate(params); ak4458->fs = nfs1; @@ -419,8 +419,13 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, AK4458_00_CONTROL1, AK4458_DIF_MASK, format); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } @@ -429,6 +434,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); + int ret; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: /* Slave Mode */ @@ -461,8 +467,13 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) ak4458->fmt == SND_SOC_DAIFMT_PDM ? AK4458_DP_MASK : 0); - ak4458_rstn_control(component, 0); - ak4458_rstn_control(component, 1); + ret = ak4458_rstn_control(component, 0); + if (ret) + return ret; + + ret = ak4458_rstn_control(component, 1); + if (ret) + return ret; return 0; } -- 2.27.0

2 1

[v6 2/3] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Apr '21

19 Apr '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in seperate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 560 +++++++++++++++++++++++++++++++++ 3 files changed, 568 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..7624475efa08 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1058,6 +1059,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..6938a7a1f02b --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,560 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct rk808 *rk808; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding for now. + * Additionally, I do not know according to the documentation the units accepted for the clock + * values, so for the moment those are left unvalidated. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +static int rk817_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + /* Set resistor value and charge pump current for PLL. */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + /* Set the PLL feedback clock divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + /* Set the PLL pre-divide value (values not documented). */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + /* Set the PLL VCO output clock divide and PLL divided ratio of PLL High Clk (values not + * documented). + */ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + + return 0; +} + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db, 0.375db/step, for example: + * 0x00: 0dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +/* + * PGA GAIN L/R volume setting + * 27db~-18db, 3db/step, for example: + * 0x0: -18dB + * 0xf: 27dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_gain_tlv, -1800, 2700); + +static const struct snd_kcontrol_new rk817_volume_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_TLV("Mic Capture Gain", RK817_CODEC_DMIC_PGA_GAIN, 4, 0, 0xf, 0, + rk817_gain_tlv), +}; + +/* Since the speaker output and L headphone pin are internally the same, make audio path mutually + * exclusive with a mux. + */ + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + struct rk808 *rk808 = dev_get_drvdata(component->dev->parent); + int ret; + + snd_soc_component_init_regmap(component, rk808->regmap); + rk817->component = component; + + ret = snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + + rk817_init(component); + + /* setting initial pll values so that we can continue to leverage simple-audio-card. + * The values aren't important since no parameters are used. + */ + + snd_soc_component_set_pll(component, 0, 0, 0, 0); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_volume_controls, + .num_controls = ARRAY_SIZE(rk817_volume_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), + .set_pll = rk817_set_component_pll, +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk808 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk808) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + rk817_codec_data->rk808 = rk808; + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->mclk = clk_get(pdev->dev.parent, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = clk_prepare_enable(rk817_codec_data->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "%s() clock prepare error %d\n", + __func__, ret); + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = platform_get_drvdata(pdev); + + clk_disable_unprepare(rk817->mclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1

2 1

[RFC PATCH 0/3] Separate BE DAI HW constraints from FE ones
by Codrin Ciubotariu 19 Apr '21

19 Apr '21

HW constraints are needed to set limitations for HW parameters used to configure the DAIs. All DAIs on the same link must agree upon the HW parameters, so the parameters are affected by the DAIs' features and their limitations. In case of DPCM, the FE DAIs can be used to perform different kind of conversions, such as format or rate changing, bringing the audio stream to a configuration supported by all the DAIs of the BE's link. For this reason, the limitations of the BE DAIs are not always important for the HW parameters between user-space and FE, only for the paratemers between FE and BE DAI links. This brings us to this patch-set, which aims to separate the FE HW constraints from the BE HW constraints. This way, the FE can be used to perform more efficient HW conversions, on the initial audio stream parameters, to parameters supported by the BE DAIs. To achieve this, the first thing needed is to detect whether a HW constraint rule is enforced by a FE or a BE DAI. This means that snd_pcm_hw_rule_add() needs to be able to differentiate between the two type of DAIs. For this, the runtime pointer to struct snd_pcm_runtime is replaced with a pointer to struct snd_pcm_substream, to be able to reach substream->pcm->internal to differentiate between FE and BE DAIs. This change affects many sound drivers (and one gpu drm driver). All these changes are included in the first patch, to have a better overview of the implications created by this change. The second patch adds a new struct snd_pcm_hw_constraints under struct snd_soc_dpcm_runtime, which is used to store the HW constraint rules added by the BE DAIs. This structure is initialized with a subset of the runtime constraint rules which does not include the rules that affect the buffer or period size. snd_pcm_hw_rule_add() will add the BE rules to the new struct snd_pcm_hw_constraints. The third and last patch will apply the BE rule constraints, after the fixup callback. If the fixup HW parameters do not respect the BE constraint rules, the rules will exit with an error. The FE mask and interval constraints are copied to the BE ones, to satisfy the dai_link->dpcm_merged_* flags. The dai_link->dpcm_merged_* flags are used to know if the FE does format or sampling rate conversion. I tested with ad1934 and wm8731 codecs as BEs, with a not-yet-mainlined ASRC as FE, that can also do format conversion. I realize that the change to snd_pcm_hw_rule_add() has a big impact, even though all the drivers use snd_pcm_hw_rule_add() with substream->runtime, so passing substream instead of runtime is not that risky. Codrin Ciubotariu (3): pcm: use substream instead of runtime in snd_pcm_hw_rule_add() ASoC: soc-pcm: add hw_constraints for BE DAI links ASoC: soc-pcm: apply BE HW constraint rules drivers/gpu/drm/vc4/vc4_hdmi.c | 2 +- include/sound/ac97_codec.h | 2 +- include/sound/pcm.h | 29 +++++-- include/sound/pcm_drm_eld.h | 2 +- include/sound/soc-dpcm.h | 1 + sound/arm/aaci.c | 4 +- sound/arm/pxa2xx-pcm-lib.c | 4 +- sound/core/pcm_drm_eld.c | 6 +- sound/core/pcm_lib.c | 66 ++++++++------- sound/core/pcm_native.c | 83 +++++++++++-------- sound/drivers/aloop.c | 8 +- sound/drivers/vx/vx_pcm.c | 8 +- sound/firewire/amdtp-am824.c | 8 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 12 +-- sound/firewire/amdtp-stream.h | 2 +- sound/firewire/bebob/bebob_pcm.c | 6 +- sound/firewire/dice/dice-pcm.c | 6 +- sound/firewire/digi00x/amdtp-dot.c | 6 +- sound/firewire/digi00x/digi00x-pcm.c | 6 +- sound/firewire/digi00x/digi00x.h | 2 +- sound/firewire/fireface/amdtp-ff.c | 6 +- sound/firewire/fireface/ff-pcm.c | 6 +- sound/firewire/fireface/ff.h | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 6 +- sound/firewire/motu/amdtp-motu.c | 6 +- sound/firewire/motu/motu-pcm.c | 6 +- sound/firewire/motu/motu.h | 2 +- sound/firewire/oxfw/oxfw-pcm.c | 6 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/firewire/tascam/tascam.h | 2 +- sound/pci/ac97/ac97_pcm.c | 8 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0_pcm.c | 6 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 8 +- sound/pci/ca0106/ca0106_main.c | 4 +- sound/pci/cmipci.c | 14 ++-- sound/pci/cs4281.c | 4 +- sound/pci/cs46xx/cs46xx_lib.c | 4 +- sound/pci/echoaudio/echoaudio.c | 24 +++--- sound/pci/emu10k1/emu10k1x.c | 4 +- sound/pci/emu10k1/emupcm.c | 10 +-- sound/pci/ens1370.c | 12 +-- sound/pci/es1938.c | 4 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 6 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/hda/patch_hdmi.c | 8 +- sound/pci/hda/patch_si3054.c | 2 +- sound/pci/ice1712/ice1712.c | 8 +- sound/pci/ice1712/ice1724.c | 30 +++---- sound/pci/intel8x0.c | 10 +-- sound/pci/intel8x0m.c | 2 +- sound/pci/lola/lola_pcm.c | 4 +- sound/pci/mixart/mixart.c | 8 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_pcm.c | 8 +- sound/pci/pcxhr/pcxhr.c | 4 +- sound/pci/rme32.c | 14 ++-- sound/pci/rme96.c | 17 ++-- sound/pci/rme9652/hdsp.c | 24 +++--- sound/pci/rme9652/hdspm.c | 12 +-- sound/pci/rme9652/rme9652.c | 20 ++--- sound/pci/sonicvibes.c | 4 +- sound/pci/via82xx.c | 4 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 4 +- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-da7219-max98357a.c | 40 ++++----- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 5 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/codecs/ad193x.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4613.c | 6 +- sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l33.c | 2 +- sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 4 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs4234.c | 4 +- sound/soc/codecs/cs42l73.c | 2 +- sound/soc/codecs/cs43130.c | 4 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8328.c | 2 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/max98090.c | 2 +- sound/soc/codecs/max9867.c | 2 +- sound/soc/codecs/pcm512x.c | 6 +- sound/soc/codecs/sigmadsp.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/uda1334.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8524.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_sai.c | 4 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 9 +- sound/soc/fsl/imx-audmix.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 4 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 12 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 +-- sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/aiu-fifo.c | 4 +- sound/soc/meson/axg-fifo.c | 4 +- sound/soc/sh/rcar/core.c | 6 +- sound/soc/soc-pcm.c | 76 ++++++++++++++++- sound/soc/sti/uniperif_player.c | 4 +- sound/soc/sti/uniperif_reader.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/davinci-mcasp.c | 12 +-- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/uniphier/aio-dma.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/usb/hiface/pcm.c | 2 +- sound/usb/line6/capture.c | 2 +- sound/usb/line6/playback.c | 2 +- sound/usb/misc/ua101.c | 2 +- sound/usb/pcm.c | 24 +++--- 135 files changed, 535 insertions(+), 429 deletions(-) -- 2.27.0

5 20

[PATCH] ALSA: seq: oss: Fix inconsistent indenting
by Yang Li 19 Apr '21

19 Apr '21

Kernel test robot throws below warning -> smatch warnings: sound/core/seq/oss/seq_oss_event.c:297 note_on_event() warn: inconsistent indenting Fixed the inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/core/seq/oss/seq_oss_event.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_event.c b/sound/core/seq/oss/seq_oss_event.c index 7b7c925..9a42713 100644 --- a/sound/core/seq/oss/seq_oss_event.c +++ b/sound/core/seq/oss/seq_oss_event.c @@ -294,7 +294,7 @@ /* set volume to zero -- note off */ // type = SNDRV_SEQ_EVENT_NOTEOFF; //else - if (info->ch[ch].vel) + if (info->ch[ch].vel) /* sample already started -- volume change */ type = SNDRV_SEQ_EVENT_KEYPRESS; else -- 1.8.3.1

3 2

[RESEND PATCH] pinctrl: mediatek: use spin lock in mtk_rmw
by Tzung-Bi Shih 19 Apr '21

19 Apr '21

Commit 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") uses mutex lock in mtk_rmw. However the function is possible called from atomic context. For example call trace: mutex_lock+0x28/0x64 mtk_rmw+0x38/0x80 mtk_hw_set_value+0x100/0x138 mtk_gpio_set+0x48/0x58 gpiod_set_raw_value_commit+0xf4/0x110 gpiod_set_value_nocheck+0x4c/0x80 gpiod_set_value+0x4c/0x6c max98357a_daiops_trigger+0x8c/0x9c soc_pcm_trigger+0x5c/0x10c The max98357a_daiops_trigger() could run in either atomic or non-atomic context. As a result, dmesg shows some similar messages: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:254". Uses spin lock in mtk_rmw instead. Fixes: 42a46434e9b1 ("pinctrl: add lock in mtk_rmw function.") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/pinctrl/mediatek/pinctrl-moore.c | 2 +- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c | 4 ++-- drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h | 2 +- drivers/pinctrl/mediatek/pinctrl-paris.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/pinctrl/mediatek/pinctrl-moore.c b/drivers/pinctrl/mediatek/pinctrl-moore.c index f77921957f15..3a4a23c40a71 100644 --- a/drivers/pinctrl/mediatek/pinctrl-moore.c +++ b/drivers/pinctrl/mediatek/pinctrl-moore.c @@ -619,7 +619,7 @@ int mtk_moore_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); /* Copy from internal struct mtk_pin_desc to register to the core */ pins = devm_kmalloc_array(&pdev->dev, hw->soc->npins, sizeof(*pins), diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c index fcf7c3eeee4a..9ffe01a8ceca 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.c @@ -58,14 +58,14 @@ void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set) { u32 val; - mutex_lock(&pctl->lock); + spin_lock(&pctl->lock); val = mtk_r32(pctl, i, reg); val &= ~mask; val |= set; mtk_w32(pctl, i, reg, val); - mutex_unlock(&pctl->lock); + spin_unlock(&pctl->lock); } static int mtk_hw_pin_field_lookup(struct mtk_pinctrl *hw, diff --git a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h index 65eac708a3b3..13836bb188b7 100644 --- a/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h +++ b/drivers/pinctrl/mediatek/pinctrl-mtk-common-v2.h @@ -252,7 +252,7 @@ struct mtk_pinctrl { struct mtk_pinctrl_group *groups; const char **grp_names; /* lock pin's register resource to avoid multiple threads issue*/ - struct mutex lock; + spinlock_t lock; }; void mtk_rmw(struct mtk_pinctrl *pctl, u8 i, u32 reg, u32 mask, u32 set); diff --git a/drivers/pinctrl/mediatek/pinctrl-paris.c b/drivers/pinctrl/mediatek/pinctrl-paris.c index 48e823f6d293..85db2e4377f0 100644 --- a/drivers/pinctrl/mediatek/pinctrl-paris.c +++ b/drivers/pinctrl/mediatek/pinctrl-paris.c @@ -970,7 +970,7 @@ int mtk_paris_pinctrl_probe(struct platform_device *pdev, hw->nbase = hw->soc->nbase_names; - mutex_init(&hw->lock); + spin_lock_init(&hw->lock); err = mtk_pctrl_build_state(pdev); if (err) { -- 2.31.1.368.gbe11c130af-goog

2 2

[PATCH v5] sound: rawmidi: Add framing mode
by David Henningsson 19 Apr '21

19 Apr '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- This version of the patch has been compile tested only, and compared to v4, has only the changes with no controversy. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 26 ++++++++++++- sound/core/rawmidi.c | 73 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 6 ++- 4 files changed, 103 insertions(+), 4 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..b0057a193c31 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + u8 framing; /* whether to frame input data */ + clockid_t clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..124ac74a13e9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,34 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +enum { + SNDRV_RAWMIDI_FRAMING_NONE = 0, + SNDRV_RAWMIDI_FRAMING_TSTAMP, + SNDRV_RAWMIDI_FRAMING_LAST = SNDRV_RAWMIDI_FRAMING_TSTAMP, +}; + +#define SND_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + u8 frame_type; + u8 length; /* number of valid bytes in data field */ + u8 reserved[2]; + u32 tv_nsec; /* nanoseconds */ + u64 tv_sec; /* seconds */ + u8 data[SND_RAWMIDI_FRAMING_DATA_LENGTH]; +}; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned char framing; /* For input data only, frame incoming data */ + unsigned char clock_type; /* Type of clock to use for framing, same as clockid_t */ + unsigned char reserved[14]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..c3a4940a919d 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -683,6 +683,8 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (params->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,7 +722,16 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + if (params->framing) { + if (params->framing > SNDRV_RAWMIDI_FRAMING_LAST) + return -EINVAL; + /* framing requires a valid clock type */ + if (params->clock_type != CLOCK_MONOTONIC_RAW && params->clock_type != CLOCK_MONOTONIC) + return -EINVAL; + } snd_rawmidi_drain_input(substream); + substream->framing = params->framing; + substream->clock_type = params->clock_type; return resize_runtime_buffer(substream->runtime, params, true); } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +974,56 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SND_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SND_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SND_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return dest_frames * frame_size; +} + +struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + if (substream->framing != SNDRV_RAWMIDI_FRAMING_TSTAMP) + return ts64; + if (substream->clock_type == CLOCK_MONOTONIC_RAW) + ktime_get_raw_ts64(&ts64); + else + ktime_get_ts64(&ts64); + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1038,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1049,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1596,6 +1661,12 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_FRAMING_TSTAMP) { + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + substream->clock_type == CLOCK_MONOTONIC_RAW ? "monotonic raw" : "monotonic"); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..2603d2dd8abb 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,9 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned char framing; + unsigned char clock_type; + unsigned char reserved[14]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +27,8 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.framing, &src->framing) || + get_user(params.clock_type, &src->clock_type) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

4 7

[PATCH] ALSA: hda/realtek: GA503 use same quirks as GA401
by Luke D Jones 19 Apr '21

19 Apr '21

The GA503 has almost exactly the same default setup as the GA401 model with the same issues. The GA401 quirks solve all the issues so we will use the full quirk chain. Signed-off-by: Luke D Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290645516313..9a5546df1e6a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8019,6 +8019,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), + SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), -- 2.31.1

2 1

[PATCH] ALSA: usb-audio: DJM-750: ensure format is set
by Olivia Mackintosh 19 Apr '21

19 Apr '21

Add case statement to set sample-rate for the DJM-750 Pioneer mixer. This was included as part of another patch but I think it has been archived on Patchwork and hasn't been merged. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 63f27ce64bcb..8b8bee3c3dd6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1511,6 +1511,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; + case USB_ID(0x08e4, 0x017f): /* Pioneer DJM-750 */ case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; -- 2.31.1

2 1

[PATCH v2] Pioneer devices: engage implicit feedback sync for playback
by Geraldo 19 Apr '21

19 Apr '21

Dear Linux users of Pioneer gear, please disregard v1 of this patch and test this instead if at all possible. My initial assessment that we needed to let the capture EP be opened twice was clearly proven wrong by further testing. This is good because if we really needed that it would require a lot of reengineering inside ALSA. One thing that still boggles my mind though is why we need a special Pioneer case inside snd_usb_parse_implicit_fb_quirk that returns 1 like a capture quirk was found. This is a highly experimental patch on top of 5.12-rc6 that's supposed to engage implicit feedback sync on the playback for Pioneer devices. Without implicit feedback sync the inputs started glitching due to clock drift in about an hour in my Pioneer DJ DDJ-SR2. Once again I'd like to thank Takashi Iwai. He's the true author of the bulk of this code, I just adapted it and coerced it into working. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- implicit.c.git 2021-04-04 20:51:57.226754632 -0300 +++ implicit.c 2021-04-05 10:02:41.059816581 -0300 @@ -177,30 +177,31 @@ static int add_roland_implicit_fb(struct ifnum, alts); } -/* Playback and capture EPs on Pioneer devices share the same iface/altset, - * but they don't seem working with the implicit fb mode well, hence we - * just return as if the sync were already set up. +/* Pioneer devices: playback and capture streams sharing the same iface/altset */ -static int skip_pioneer_sync_ep(struct snd_usb_audio *chip, - struct audioformat *fmt, - struct usb_host_interface *alts) -{ - struct usb_endpoint_descriptor *epd; - - if (alts->desc.bNumEndpoints != 2) - return 0; - - epd = get_endpoint(alts, 1); - if (!usb_endpoint_is_isoc_in(epd) || - (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || - ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_DATA && - (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != - USB_ENDPOINT_USAGE_IMPLICIT_FB)) - return 0; - return 1; /* don't handle with the implicit fb, just skip sync EP */ +static int add_pioneer_implicit_fb(struct snd_usb_audio *chip, + struct audioformat *fmt, + struct usb_host_interface *alts) +{ + struct usb_endpoint_descriptor *epd; + + if (alts->desc.bNumEndpoints != 2) + return 0; + + epd = get_endpoint(alts, 1); + + if (!usb_endpoint_is_isoc_in(epd) || + (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC || + ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_DATA && + (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) != + USB_ENDPOINT_USAGE_IMPLICIT_FB)) + return 0; + return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1, + alts->desc.bInterfaceNumber, alts); } + static int __add_generic_implicit_fb(struct snd_usb_audio *chip, struct audioformat *fmt, int iface, int altset) @@ -306,7 +307,7 @@ static int audioformat_implicit_fb_quirk alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) { - if (skip_pioneer_sync_ep(chip, fmt, alts)) + if (add_pioneer_implicit_fb(chip, fmt, alts)) return 1; } @@ -339,8 +340,20 @@ int snd_usb_parse_implicit_fb_quirk(stru struct audioformat *fmt, struct usb_host_interface *alts) { - if (fmt->endpoint & USB_DIR_IN) - return audioformat_capture_quirk(chip, fmt, alts); + bool isPioneer; + + if (alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC && + (USB_ID_VENDOR(chip->usb_id) == 0x2b73 || /* Pioneer */ + USB_ID_VENDOR(chip->usb_id) == 0x08e4 /* Pioneer */)) + isPioneer = true; + + if (fmt->endpoint & USB_DIR_IN) { + if (isPioneer == true) + return 1; + else + return audioformat_capture_quirk(chip, fmt, alts); + } + else return audioformat_implicit_fb_quirk(chip, fmt, alts); }

4 12

[RFC] ucm2: rt711-sdca: remove 'rt711 FU0F Capture Switch' setting
by GitHub pull_request - edited 19 Apr '21

19 Apr '21

alsa-project/alsa-ucm-conf pull request #89 was edited from libinyang: The rt711-sdca codec driver removes the 'rt711 FU0F Capture Switch' kcontrol. So we need to remove the setting of 'rt711 FU0F Capture Switch' correspondingly. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: rt711-sdca: remove 'rt711 FU0F Capture Switch' setting
by GitHub pull_request - opened 19 Apr '21

19 Apr '21

alsa-project/alsa-ucm-conf pull request #89 was opened from libinyang: The rt711-sdca codec driver removes the 'rt711 FU0F Capture Switch' kcontrol. So we need to remove the setting of 'rt711 FU0F Capture Switch' correspondingly. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/89.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] soundwire: only use CLOCK_STOP_MODE0 and handle -ENODATA
by Bard Liao 19 Apr '21

19 Apr '21

Existing devices and implementations only support the required CLOCK_STOP_MODE0. All the code related to CLOCK_STOP_MODE1 has not been tested and is highly questionable, with a clear confusion between CLOCK_STOP_MODE1 and the simple clock stop state machine. This patch removes all usages of CLOCK_STOP_MODE1 - which has no impact on any solution - and fixes the use of the simple clock stop state machine. The resulting code should be a lot more symmetrical and easier to maintain. Note that CLOCK_STOP_MODE1 is not supported in the SoundWire Device Class specification so it's rather unlikely that we need to re-add this mode later. If a device lost sync and can no longer ACK a command, it may not be able to enter a lower-power state but it will still be able to resync when the clock restarts. In those cases, we want to continue with the clock stop sequence. This patch modifies the behavior during clock stop sequences to only log errors unrelated to -ENODATA/Command_Ignored. The flow is also modified so that loops continue to prepare/deprepare other devices even when one seems to have lost sync. When resuming the clocks, all issues are logged with a dev_warn(), previously only some of them were checked. This is the only part that now differs between the clock stop entry and clock stop exit sequences: while we don't want to stop the suspend flow, we do want information on potential issues while resuming, as they may have ripple effects. For consistency the log messages are also modified to be unique and self-explanatory. Errors in sdw_slave_clk_stop_callback() were removed, they are now handled in the caller. Pierre-Louis Bossart (4): soundwire: bus: only use CLOCK_STOP_MODE0 and fix confusions soundwire: add missing kernel-doc description soundwire: bus: handle -ENODATA errors in clock stop/start sequences soundwire: bus: add missing \n in dynamic debug drivers/soundwire/bus.c | 155 +++++++++++++++------------------- include/linux/soundwire/sdw.h | 3 +- 2 files changed, 70 insertions(+), 88 deletions(-) -- 2.17.1

1 4

[PATCH] soundwire: cadence_master: always set CMD_ACCEPT
by Bard Liao 19 Apr '21

19 Apr '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The Cadence IP can be configured in two different ways to deal with CMD_IGNORED replies to broadcast commands. The CMD_ACCEPT bitfield controls whether the command is discarded or if the IP proceeds with the change (typically a bank switch or clock stop command). The existing code seems to be inconsistent: a) For some historical reason, we set this CMD_ACCEPT bitfield during the initialization, but we don't during a resume from a clock-stoppped state. b) In addition, the loop used in the clock-stop sequence is quite racy, it's possible that a device has lost sync but it's still tagged as ATTACHED. c) If somehow a Device loses sync and is unable to ack a broadcast command, we do not have an error handling mechanism anyways. The IP should go ahead and let the Device regain sync at a later time. Make sure the CMD_ACCEPT bit is always set. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 21 ++------------------- 1 file changed, 2 insertions(+), 19 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 192dac10f0c2..25950422b085 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1428,20 +1428,6 @@ int sdw_cdns_clock_stop(struct sdw_cdns *cdns, bool block_wake) } } - /* - * This CMD_ACCEPT should be used when there are no devices - * attached on the link when entering clock stop mode. If this is - * not set and there is a broadcast write then the command ignored - * will be treated as a failure - */ - if (!slave_present) - cdns_updatel(cdns, CDNS_MCP_CONTROL, - CDNS_MCP_CONTROL_CMD_ACCEPT, - CDNS_MCP_CONTROL_CMD_ACCEPT); - else - cdns_updatel(cdns, CDNS_MCP_CONTROL, - CDNS_MCP_CONTROL_CMD_ACCEPT, 0); - /* commit changes */ ret = cdns_config_update(cdns); if (ret < 0) { @@ -1508,11 +1494,8 @@ int sdw_cdns_clock_restart(struct sdw_cdns *cdns, bool bus_reset) cdns_updatel(cdns, CDNS_MCP_CONTROL, CDNS_MCP_CONTROL_BLOCK_WAKEUP, 0); - /* - * clear CMD_ACCEPT so that the command ignored - * will be treated as a failure during a broadcast write - */ - cdns_updatel(cdns, CDNS_MCP_CONTROL, CDNS_MCP_CONTROL_CMD_ACCEPT, 0); + cdns_updatel(cdns, CDNS_MCP_CONTROL, CDNS_MCP_CONTROL_CMD_ACCEPT, + CDNS_MCP_CONTROL_CMD_ACCEPT); if (!bus_reset) { -- 2.17.1

1 0

[PATCH v3] soundwire: intel: move to auxiliary bus
by Bard Liao 19 Apr '21

19 Apr '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Now that the auxiliary_bus exists, there's no reason to use platform devices as children of a PCI device any longer. This patch refactors the code by extending a basic auxiliary device with Intel link-specific structures that need to be passed between controller and link levels. This refactoring is much cleaner with no need for cross-pointers between device and link structures. Note that the auxiliary bus API has separate init and add steps, which requires more attention in the error unwinding paths. The main loop needs to deal with kfree() and auxiliary_device_uninit() for the current iteration before jumping to the common label which releases everything allocated in prior iterations. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - add link_dev_register for all kzalloc, device_init, and device_add. v3: - Modify the function description of sdw_intel_probe() which was missing in previous version. --- drivers/soundwire/Kconfig | 1 + drivers/soundwire/intel.c | 56 ++++--- drivers/soundwire/intel.h | 14 +- drivers/soundwire/intel_init.c | 232 +++++++++++++++++++--------- include/linux/soundwire/sdw_intel.h | 6 +- 5 files changed, 202 insertions(+), 107 deletions(-) diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index 016e74230bb7..2b7795233282 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -25,6 +25,7 @@ config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE select SOUNDWIRE_GENERIC_ALLOCATION + select AUXILIARY_BUS depends on ACPI && SND_SOC help SoundWire Intel Master driver. diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fd95f94630b1..c11e3d8cd308 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -11,7 +11,7 @@ #include <linux/module.h> #include <linux/interrupt.h> #include <linux/io.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <sound/pcm_params.h> #include <linux/pm_runtime.h> #include <sound/soc.h> @@ -1327,11 +1327,14 @@ static int intel_init(struct sdw_intel *sdw) } /* - * probe and init + * probe and init (aux_dev_id argument is required by function prototype but not used) */ -static int intel_master_probe(struct platform_device *pdev) +static int intel_link_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *aux_dev_id) + { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); struct sdw_intel *sdw; struct sdw_cdns *cdns; struct sdw_bus *bus; @@ -1344,14 +1347,14 @@ static int intel_master_probe(struct platform_device *pdev) cdns = &sdw->cdns; bus = &cdns->bus; - sdw->instance = pdev->id; - sdw->link_res = dev_get_platdata(dev); + sdw->instance = auxdev->id; + sdw->link_res = &ldev->link_res; cdns->dev = dev; cdns->registers = sdw->link_res->registers; cdns->instance = sdw->instance; cdns->msg_count = 0; - bus->link_id = pdev->id; + bus->link_id = auxdev->id; sdw_cdns_probe(cdns); @@ -1384,10 +1387,10 @@ static int intel_master_probe(struct platform_device *pdev) return 0; } -int intel_master_startup(struct platform_device *pdev) +int intel_link_startup(struct auxiliary_device *auxdev) { struct sdw_cdns_stream_config config; - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1524,9 +1527,9 @@ int intel_master_startup(struct platform_device *pdev) return ret; } -static int intel_master_remove(struct platform_device *pdev) +static void intel_link_remove(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_cdns *cdns = dev_get_drvdata(dev); struct sdw_intel *sdw = cdns_to_intel(cdns); struct sdw_bus *bus = &cdns->bus; @@ -1542,19 +1545,17 @@ static int intel_master_remove(struct platform_device *pdev) snd_soc_unregister_component(dev); } sdw_bus_master_delete(bus); - - return 0; } -int intel_master_process_wakeen_event(struct platform_device *pdev) +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev) { - struct device *dev = &pdev->dev; + struct device *dev = &auxdev->dev; struct sdw_intel *sdw; struct sdw_bus *bus; void __iomem *shim; u16 wake_sts; - sdw = platform_get_drvdata(pdev); + sdw = dev_get_drvdata(dev); bus = &sdw->cdns.bus; if (bus->prop.hw_disabled) { @@ -1976,17 +1977,22 @@ static const struct dev_pm_ops intel_pm = { SET_RUNTIME_PM_OPS(intel_suspend_runtime, intel_resume_runtime, NULL) }; -static struct platform_driver sdw_intel_drv = { - .probe = intel_master_probe, - .remove = intel_master_remove, +static const struct auxiliary_device_id intel_link_id_table[] = { + { .name = "soundwire_intel.link" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, intel_link_id_table); + +static struct auxiliary_driver sdw_intel_drv = { + .probe = intel_link_probe, + .remove = intel_link_remove, .driver = { - .name = "intel-sdw", + /* auxiliary_driver_register() sets .name to be the modname */ .pm = &intel_pm, - } + }, + .id_table = intel_link_id_table }; - -module_platform_driver(sdw_intel_drv); +module_auxiliary_driver(sdw_intel_drv); MODULE_LICENSE("Dual BSD/GPL"); -MODULE_ALIAS("platform:intel-sdw"); -MODULE_DESCRIPTION("Intel Soundwire Master Driver"); +MODULE_DESCRIPTION("Intel Soundwire Link Driver"); diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 06bac8ba14e9..0b47b148da3f 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -7,7 +7,6 @@ /** * struct sdw_intel_link_res - Soundwire Intel link resource structure, * typically populated by the controller driver. - * @pdev: platform_device * @mmio_base: mmio base of SoundWire registers * @registers: Link IO registers base * @shim: Audio shim pointer @@ -23,7 +22,6 @@ * @list: used to walk-through all masters exposed by the same controller */ struct sdw_intel_link_res { - struct platform_device *pdev; void __iomem *mmio_base; /* not strictly needed, useful for debug */ void __iomem *registers; void __iomem *shim; @@ -48,7 +46,15 @@ struct sdw_intel { #endif }; -int intel_master_startup(struct platform_device *pdev); -int intel_master_process_wakeen_event(struct platform_device *pdev); +int intel_link_startup(struct auxiliary_device *auxdev); +int intel_link_process_wakeen_event(struct auxiliary_device *auxdev); + +struct sdw_intel_link_dev { + struct auxiliary_device auxdev; + struct sdw_intel_link_res link_res; +}; + +#define auxiliary_dev_to_sdw_intel_link_dev(auxiliary_dev) \ + container_of(auxiliary_dev, struct sdw_intel_link_dev, auxdev) #endif /* __SDW_INTEL_LOCAL_H */ diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index 05b726cdfebc..c75865f583df 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -12,7 +12,7 @@ #include <linux/interrupt.h> #include <linux/io.h> #include <linux/module.h> -#include <linux/platform_device.h> +#include <linux/auxiliary_bus.h> #include <linux/pm_runtime.h> #include <linux/soundwire/sdw_intel.h> #include "cadence_master.h" @@ -24,28 +24,108 @@ #define SDW_LINK_BASE 0x30000 #define SDW_LINK_SIZE 0x10000 +static void intel_link_dev_release(struct device *dev) +{ + struct auxiliary_device *auxdev = to_auxiliary_dev(dev); + struct sdw_intel_link_dev *ldev = auxiliary_dev_to_sdw_intel_link_dev(auxdev); + + kfree(ldev); +} + +/* alloc, init and add link devices */ +static struct sdw_intel_link_dev *intel_link_dev_register(struct sdw_intel_res *res, + struct sdw_intel_ctx *ctx, + struct fwnode_handle *fwnode, + const char *name, + int link_id) +{ + struct sdw_intel_link_dev *ldev; + struct sdw_intel_link_res *link; + struct auxiliary_device *auxdev; + int ret; + + ldev = kzalloc(sizeof(*ldev), GFP_KERNEL); + if (!ldev) + return ERR_PTR(-ENOMEM); + + auxdev = &ldev->auxdev; + auxdev->name = name; + auxdev->dev.parent = res->parent; + auxdev->dev.fwnode = fwnode; + auxdev->dev.release = intel_link_dev_release; + + /* we don't use an IDA since we already have a link ID */ + auxdev->id = link_id; + + /* + * keep a handle on the allocated memory, to be used in all other functions. + * Since the same pattern is used to skip links that are not enabled, there is + * no need to check if ctx->ldev[i] is NULL later on. + */ + ctx->ldev[link_id] = ldev; + + /* Add link information used in the driver probe */ + link = &ldev->link_res; + link->mmio_base = res->mmio_base; + link->registers = res->mmio_base + SDW_LINK_BASE + + (SDW_LINK_SIZE * link_id); + link->shim = res->mmio_base + SDW_SHIM_BASE; + link->alh = res->mmio_base + SDW_ALH_BASE; + + link->ops = res->ops; + link->dev = res->dev; + + link->clock_stop_quirks = res->clock_stop_quirks; + link->shim_lock = &ctx->shim_lock; + link->shim_mask = &ctx->shim_mask; + link->link_mask = ctx->link_mask; + + /* now follow the two-step init/add sequence */ + ret = auxiliary_device_init(auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to initialize link dev %s link_id %d\n", + name, link_id); + kfree(ldev); + return ERR_PTR(ret); + } + + ret = auxiliary_device_add(&ldev->auxdev); + if (ret < 0) { + dev_err(res->parent, "failed to add link dev %s link_id %d\n", + ldev->auxdev.name, link_id); + /* ldev will be freed with the put_device() and .release sequence */ + auxiliary_device_uninit(&ldev->auxdev); + return ERR_PTR(ret); + } + + return ldev; +} + +static void intel_link_dev_unregister(struct sdw_intel_link_dev *ldev) +{ + auxiliary_device_delete(&ldev->auxdev); + auxiliary_device_uninit(&ldev->auxdev); +} + static int sdw_intel_cleanup(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link = ctx->links; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!link) - return 0; - link_mask = ctx->link_mask; - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - if (link->pdev) { - pm_runtime_disable(&link->pdev->dev); - platform_device_unregister(link->pdev); - } + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) - pm_runtime_put_noidle(link->dev); + pm_runtime_disable(&ldev->auxdev.dev); + intel_link_dev_unregister(ldev); + + if (!ldev->link_res.clock_stop_quirks) + pm_runtime_put_noidle(ldev->link_res.dev); } return 0; @@ -91,9 +171,8 @@ EXPORT_SYMBOL_NS(sdw_intel_thread, SOUNDWIRE_INTEL_INIT); static struct sdw_intel_ctx *sdw_intel_probe_controller(struct sdw_intel_res *res) { - struct platform_device_info pdevinfo; - struct platform_device *pdev; struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; struct sdw_intel_ctx *ctx; struct acpi_device *adev; struct sdw_slave *slave; @@ -116,67 +195,60 @@ static struct sdw_intel_ctx count = res->count; dev_dbg(&adev->dev, "Creating %d SDW Link devices\n", count); - ctx = devm_kzalloc(&adev->dev, sizeof(*ctx), GFP_KERNEL); + /* + * we need to alloc/free memory manually and can't use devm: + * this routine may be called from a workqueue, and not from + * the parent .probe. + * If devm_ was used, the memory might never be freed on errors. + */ + ctx = kzalloc(sizeof(*ctx), GFP_KERNEL); if (!ctx) return NULL; ctx->count = count; - ctx->links = devm_kcalloc(&adev->dev, ctx->count, - sizeof(*ctx->links), GFP_KERNEL); - if (!ctx->links) + + /* + * allocate the array of pointers. The link-specific data is allocated + * as part of the first loop below and released with the auxiliary_device_uninit(). + * If some links are disabled, the link pointer will remain NULL. Given that the + * number of links is small, this is simpler than using a list to keep track of links. + */ + ctx->ldev = kcalloc(ctx->count, sizeof(*ctx->ldev), GFP_KERNEL); + if (!ctx->ldev) { + kfree(ctx); return NULL; + } - ctx->count = count; ctx->mmio_base = res->mmio_base; ctx->link_mask = res->link_mask; ctx->handle = res->handle; mutex_init(&ctx->shim_lock); - link = ctx->links; link_mask = ctx->link_mask; INIT_LIST_HEAD(&ctx->link_list); - /* Create SDW Master devices */ - for (i = 0; i < count; i++, link++) { - if (!(link_mask & BIT(i))) { - dev_dbg(&adev->dev, - "Link %d masked, will not be enabled\n", i); + for (i = 0; i < count; i++) { + if (!(link_mask & BIT(i))) continue; - } - link->mmio_base = res->mmio_base; - link->registers = res->mmio_base + SDW_LINK_BASE - + (SDW_LINK_SIZE * i); - link->shim = res->mmio_base + SDW_SHIM_BASE; - link->alh = res->mmio_base + SDW_ALH_BASE; - - link->ops = res->ops; - link->dev = res->dev; - - link->clock_stop_quirks = res->clock_stop_quirks; - link->shim_lock = &ctx->shim_lock; - link->shim_mask = &ctx->shim_mask; - link->link_mask = link_mask; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - - pdevinfo.parent = res->parent; - pdevinfo.name = "intel-sdw"; - pdevinfo.id = i; - pdevinfo.fwnode = acpi_fwnode_handle(adev); - pdevinfo.data = link; - pdevinfo.size_data = sizeof(*link); - - pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(pdev)) { - dev_err(&adev->dev, - "platform device creation failed: %ld\n", - PTR_ERR(pdev)); + /* + * init and add a device for each link + * + * The name of the device will be soundwire_intel.link.[i], + * with the "soundwire_intel" module prefix automatically added + * by the auxiliary bus core. + */ + ldev = intel_link_dev_register(res, + ctx, + acpi_fwnode_handle(adev), + "link", + i); + if (IS_ERR(ldev)) goto err; - } - link->pdev = pdev; - link->cdns = platform_get_drvdata(pdev); + + link = &ldev->link_res; + link->cdns = dev_get_drvdata(&ldev->auxdev.dev); list_add_tail(&link->list, &ctx->link_list); bus = &link->cdns->bus; @@ -185,8 +257,7 @@ static struct sdw_intel_ctx num_slaves++; } - ctx->ids = devm_kcalloc(&adev->dev, num_slaves, - sizeof(*ctx->ids), GFP_KERNEL); + ctx->ids = kcalloc(num_slaves, sizeof(*ctx->ids), GFP_KERNEL); if (!ctx->ids) goto err; @@ -204,8 +275,14 @@ static struct sdw_intel_ctx return ctx; err: - ctx->count = i; - sdw_intel_cleanup(ctx); + while (i--) { + if (!(link_mask & BIT(i))) + continue; + ldev = ctx->ldev[i]; + intel_link_dev_unregister(ldev); + } + kfree(ctx->ldev); + kfree(ctx); return NULL; } @@ -213,7 +290,7 @@ static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { struct acpi_device *adev; - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; @@ -232,27 +309,28 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) return -EINVAL; } - if (!ctx->links) + if (!ctx->ldev) return -EINVAL; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_startup(link->pdev); + ldev = ctx->ldev[i]; - if (!link->clock_stop_quirks) { + intel_link_startup(&ldev->auxdev); + + if (!ldev->link_res.clock_stop_quirks) { /* * we need to prevent the parent PCI device * from entering pm_runtime suspend, so that * power rails to the SoundWire IP are not * turned off. */ - pm_runtime_get_noresume(link->dev); + pm_runtime_get_noresume(ldev->link_res.dev); } } @@ -263,8 +341,8 @@ sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) * sdw_intel_probe() - SoundWire Intel probe routine * @res: resource data * - * This registers a platform device for each Master handled by the controller, - * and SoundWire Master and Slave devices will be created by the platform + * This registers an auxiliary device for each Master handled by the controller, + * and SoundWire Master and Slave devices will be created by the auxiliary * device probe. All the information necessary is stored in the context, and * the res argument pointer can be freed after this step. * This function will be called after sdw_intel_acpi_scan() by SOF probe. @@ -297,27 +375,31 @@ EXPORT_SYMBOL_NS(sdw_intel_startup, SOUNDWIRE_INTEL_INIT); void sdw_intel_exit(struct sdw_intel_ctx *ctx) { sdw_intel_cleanup(ctx); + kfree(ctx->ids); + kfree(ctx->ldev); + kfree(ctx); } EXPORT_SYMBOL_NS(sdw_intel_exit, SOUNDWIRE_INTEL_INIT); void sdw_intel_process_wakeen_event(struct sdw_intel_ctx *ctx) { - struct sdw_intel_link_res *link; + struct sdw_intel_link_dev *ldev; u32 link_mask; int i; - if (!ctx->links) + if (!ctx->ldev) return; - link = ctx->links; link_mask = ctx->link_mask; /* Startup SDW Master devices */ - for (i = 0; i < ctx->count; i++, link++) { + for (i = 0; i < ctx->count; i++) { if (!(link_mask & BIT(i))) continue; - intel_master_process_wakeen_event(link->pdev); + ldev = ctx->ldev[i]; + + intel_link_process_wakeen_event(&ldev->auxdev); } } EXPORT_SYMBOL_NS(sdw_intel_process_wakeen_event, SOUNDWIRE_INTEL_INIT); diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 3a5446ac014a..1ebea7764011 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -58,7 +58,7 @@ struct sdw_intel_acpi_info { u32 link_mask; }; -struct sdw_intel_link_res; +struct sdw_intel_link_dev; /* Intel clock-stop/pm_runtime quirk definitions */ @@ -109,7 +109,7 @@ struct sdw_intel_slave_id { * Controller * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle - * @links: information for each link (controller-specific and kept + * @ldev: information for each link (controller-specific and kept * opaque here) * @ids: array of slave_id, representing Slaves exposed across all enabled * links @@ -123,7 +123,7 @@ struct sdw_intel_ctx { u32 link_mask; int num_slaves; acpi_handle handle; - struct sdw_intel_link_res *links; + struct sdw_intel_link_dev **ldev; struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ -- 2.17.1

1 0

[PATCH] soundwire: add slave device to linked list after device_register()
by Bard Liao 19 Apr '21

19 Apr '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We currently add the slave device to a linked list before device_register(), and then remove it if device_register() fails. It's not clear why this sequence was necessary, this patch moves the linked list management to after the device_register(). Suggested-by: Keyon Jie <yang.jie(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/slave.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 112b21967c7a..0c92db2e1ddc 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -65,9 +65,6 @@ int sdw_slave_add(struct sdw_bus *bus, for (i = 0; i < SDW_MAX_PORTS; i++) init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); - list_add_tail(&slave->node, &bus->slaves); - mutex_unlock(&bus->bus_lock); ret = device_register(&slave->dev); if (ret) { @@ -77,13 +74,15 @@ int sdw_slave_add(struct sdw_bus *bus, * On err, don't free but drop ref as this will be freed * when release method is invoked. */ - mutex_lock(&bus->bus_lock); - list_del(&slave->node); - mutex_unlock(&bus->bus_lock); put_device(&slave->dev); return ret; } + + mutex_lock(&bus->bus_lock); + list_add_tail(&slave->node, &bus->slaves); + mutex_unlock(&bus->bus_lock); + sdw_slave_debugfs_init(slave); return ret; -- 2.17.1

4 3

[PATCH] soundwire: dmi-quirks: remove duplicate initialization
by Bard Liao 19 Apr '21

19 Apr '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> cppcheck warning: drivers/soundwire/dmi-quirks.c:85:12: style: Redundant initialization for 'map'. The initialized value is overwritten before it is read. [redundantInitialization] for (map = dmi_id->driver_data; map->adr; map++) { ^ drivers/soundwire/dmi-quirks.c:83:25: note: map is initialized struct adr_remap *map = dmi_id->driver_data; ^ drivers/soundwire/dmi-quirks.c:85:12: note: map is overwritten for (map = dmi_id->driver_data; map->adr; map++) { ^ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 82061c1d9835..5db0a2443a1d 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -80,7 +80,7 @@ u64 sdw_dmi_override_adr(struct sdw_bus *bus, u64 addr) /* check if any address remap quirk applies */ dmi_id = dmi_first_match(adr_remap_quirk_table); if (dmi_id) { - struct adr_remap *map = dmi_id->driver_data; + struct adr_remap *map; for (map = dmi_id->driver_data; map->adr; map++) { if (map->adr == addr) { -- 2.17.1

1 0

[PATCH v3] ALSA: control: Add memory consumption limit to user controls
by Takashi Sakamoto 18 Apr '21

18 Apr '21

ALSA control interface allows users to add arbitrary control elements (called "user controls" or "user elements"), and its resource usage is limited just by the max number of control sets (currently 32). This limit, however, is quite loose: each allocation of control set may have 1028 elements, and each element may have up to 512 bytes (ILP32) or 1024 bytes (LP64) of value data. Moreover, each control set may contain the enum strings and TLV data, which can be up to 64kB and 128kB, respectively. Totally, the whole memory consumption may go over 38MB -- it's quite large, and we'd rather like to reduce the size. OTOH, there have been other requests even to increase the max number of user elements; e.g. ALSA firewire stack require the more user controls, hence we want to raise the bar, too. For satisfying both requirements, this patch changes the management of user controls: instead of setting the upper limit of the number of user controls, we check the actual memory allocation size and set the upper limit of the total allocation in bytes. As long as the memory consumption stays below the limit, more user controls are allowed than the current limit 32. At the same time, we set the lower limit (8MB) as default than the current theoretical limit, in order to lower the risk of DoS. As a compromise for lowering the default limit, now the actual memory limit is defined as a module option, 'max_user_ctl_alloc_size', so that user can increase/decrease the limit if really needed, too. Co-developed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Tested-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- v1->v2: Drop alloc_size field from user_element, calculate at private_free v2->v3: Rebase. Fix boundary error. Obsolete macro usage relying on modern compiler optimization. Change comment style by modern coding convention. Rename module parameter so that users get it easily. Patch comment improvements. --- include/sound/core.h | 2 +- sound/core/control.c | 75 ++++++++++++++++++++++++++++++-------------- 2 files changed, 52 insertions(+), 25 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 2e24f194ef70..1f9aef0adbc9 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -100,7 +100,7 @@ struct snd_card { struct rw_semaphore controls_rwsem; /* controls list lock */ rwlock_t ctl_files_rwlock; /* ctl_files list lock */ int controls_count; /* count of all controls */ - int user_ctl_count; /* count of all user controls */ + size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ struct list_head ctl_files; /* active control files */ diff --git a/sound/core/control.c b/sound/core/control.c index 20d707d4ef40..a076c08c21b6 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -7,6 +7,7 @@ #include <linux/threads.h> #include <linux/interrupt.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/slab.h> #include <linux/vmalloc.h> #include <linux/time.h> @@ -18,8 +19,11 @@ #include <sound/info.h> #include <sound/control.h> -/* max number of user-defined controls */ -#define MAX_USER_CONTROLS 32 +// Max allocation size for user controls. +static int max_user_ctl_alloc_size = 8 * 1024 * 1024; +module_param_named(max_user_ctl_alloc_size, max_user_ctl_alloc_size, int, 0444); +MODULE_PARM_DESC(max_user_ctl_alloc_size, "Max allocation size for user controls"); + #define MAX_CONTROL_COUNT 1028 struct snd_kctl_ioctl { @@ -561,9 +565,6 @@ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, goto error; } ret = snd_ctl_remove(card, kctl); - if (ret < 0) - goto error; - card->user_ctl_count--; error: up_write(&card->controls_rwsem); return ret; @@ -1265,6 +1266,12 @@ struct user_element { void *priv_data; /* private data (like strings for enumerated type) */ }; +// check whether the addition (in bytes) of user ctl element may overflow the limit. +static bool check_user_elem_overflow(struct snd_card *card, ssize_t add) +{ + return (ssize_t)card->user_ctl_alloc_size + add > max_user_ctl_alloc_size; +} + static int snd_ctl_elem_user_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1342,6 +1349,10 @@ static int replace_user_tlv(struct snd_kcontrol *kctl, unsigned int __user *buf, if (size > 1024 * 128) /* sane value */ return -EINVAL; + // does the TLV size change cause overflow? + if (check_user_elem_overflow(ue->card, (ssize_t)(size - ue->tlv_data_size))) + return -ENOMEM; + container = vmemdup_user(buf, size); if (IS_ERR(container)) return PTR_ERR(container); @@ -1359,11 +1370,16 @@ static int replace_user_tlv(struct snd_kcontrol *kctl, unsigned int __user *buf, for (i = 0; i < kctl->count; ++i) kctl->vd[i].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; mask = SNDRV_CTL_EVENT_MASK_INFO; + } else { + ue->card->user_ctl_alloc_size -= ue->tlv_data_size; + ue->tlv_data_size = 0; + kvfree(ue->tlv_data); } - kvfree(ue->tlv_data); ue->tlv_data = container; ue->tlv_data_size = size; + // decremented at private_free. + ue->card->user_ctl_alloc_size += size; mask |= SNDRV_CTL_EVENT_MASK_TLV; for (i = 0; i < kctl->count; ++i) @@ -1405,16 +1421,17 @@ static int snd_ctl_elem_init_enum_names(struct user_element *ue) unsigned int i; const uintptr_t user_ptrval = ue->info.value.enumerated.names_ptr; - if (ue->info.value.enumerated.names_length > 64 * 1024) + buf_len = ue->info.value.enumerated.names_length; + if (buf_len > 64 * 1024) return -EINVAL; - names = vmemdup_user((const void __user *)user_ptrval, - ue->info.value.enumerated.names_length); + if (check_user_elem_overflow(ue->card, buf_len)) + return -ENOMEM; + names = vmemdup_user((const void __user *)user_ptrval, buf_len); if (IS_ERR(names)) return PTR_ERR(names); /* check that there are enough valid names */ - buf_len = ue->info.value.enumerated.names_length; p = names; for (i = 0; i < ue->info.value.enumerated.items; ++i) { name_len = strnlen(p, buf_len); @@ -1428,14 +1445,27 @@ static int snd_ctl_elem_init_enum_names(struct user_element *ue) ue->priv_data = names; ue->info.value.enumerated.names_ptr = 0; + // increment the allocation size; decremented again at private_free. + ue->card->user_ctl_alloc_size += ue->info.value.enumerated.names_length; return 0; } +static size_t compute_user_elem_size(size_t size, unsigned int count) +{ + return sizeof(struct user_element) + size * count; +} + static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) { struct user_element *ue = kcontrol->private_data; + // decrement the allocation size. + ue->card->user_ctl_alloc_size -= compute_user_elem_size(ue->elem_data_size, kcontrol->count); + ue->card->user_ctl_alloc_size -= ue->tlv_data_size; + if (ue->priv_data) + ue->card->user_ctl_alloc_size -= ue->info.value.enumerated.names_length; + kvfree(ue->tlv_data); kvfree(ue->priv_data); kfree(ue); @@ -1449,6 +1479,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, unsigned int count; unsigned int access; long private_size; + size_t alloc_size; struct user_element *ue; unsigned int offset; int err; @@ -1466,13 +1497,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, return err; } - /* - * The number of userspace controls are counted control by control, - * not element by element. - */ - if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) - return -ENOMEM; - /* Check the number of elements for this userspace control. */ count = info->owner; if (count == 0) @@ -1503,6 +1527,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; private_size = value_sizes[info->type] * info->count; + alloc_size = compute_user_elem_size(private_size, count); + + if (check_user_elem_overflow(card, alloc_size)) + return -ENOMEM; /* * Keep memory object for this userspace control. After passing this @@ -1514,16 +1542,18 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (err < 0) return err; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); - kctl->private_data = kzalloc(sizeof(struct user_element) + private_size * count, - GFP_KERNEL); - if (kctl->private_data == NULL) { + ue = kzalloc(alloc_size, GFP_KERNEL); + if (!ue) { kfree(kctl); return -ENOMEM; } + kctl->private_data = ue; kctl->private_free = snd_ctl_elem_user_free; + // increment the allocated size; decremented again at private_free. + card->user_ctl_alloc_size += alloc_size; + /* Set private data for this userspace control. */ - ue = (struct user_element *)kctl->private_data; ue->card = card; ue->info = *info; ue->info.access = 0; @@ -1565,9 +1595,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, * applications because the field originally means PID of a process * which locks the element. */ - - card->user_ctl_count++; - unlock: up_write(&card->controls_rwsem); return err; -- 2.27.0

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[PATCH] ALSA: usb-audio: Add support for many Roland devices' implicit feedback quirks
by Lucas 18 Apr '21

18 Apr '21

It makes USB audio capture and playback possible and pristine on my Roland INTEGRA-7, Boutique D-05, and R-26, along with many more I've encountered people having had issues with over the last decade or so. Signed-off-by: Lucas Endres <jaffa225man(a)gmail.com> --- diff -Nurp linux-5.11.9.orig-2021-04-13/sound/usb/implicit.c linux-5.11.9.roland1/sound/usb/implicit.c --- linux-5.11.9.orig-2021-04-13/sound/usb/implicit.c 2021-04-13 14:23:34.743782890 -0500 +++ linux-5.11.9.roland1/sound/usb/implicit.c 2021-04-15 19:35:20.053444538 -0500 @@ -79,13 +79,72 @@ static const struct snd_usb_implicit_fb_ /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a6, 0x0d, 0x01), /* Roland JUNO-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00a9, 0x0d, 0x01), /* Roland MC-808 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ad, 0x0d, 0x01), /* Roland SH-201 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b2, 0x0d, 0x01), /* Roland VG-99 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00b3, 0x0d, 0x01), /* Roland VG-99 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c2, 0x0d, 0x01), /* Roland SonicCell */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c4, 0x0d, 0x01), /* Edirol M-16DX */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c5, 0x0d, 0x01), /* Roland SP-555 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00c7, 0x0d, 0x01), /* Roland V-Synth GT */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00d1, 0x0d, 0x01), /* Roland Music Atelier */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00da, 0x0d, 0x01), /* BOSS GT-10 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00db, 0x0d, 0x01), /* BOSS GT-10 Guitar Effects Processor */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00dc, 0x0d, 0x01), /* BOSS GT-10B */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00de, 0x0d, 0x01), /* Roland Fantom-G */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00eb, 0x0d, 0x01), /* Roland VS-100 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00f8, 0x0d, 0x01), /* Roland JUNO Series */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fc, 0x0d, 0x01), /* Roland VS-700C */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fd, 0x0d, 0x01), /* Roland VS-700 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00fe, 0x0d, 0x01), /* Roland VS-700 M1 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x00ff, 0x0d, 0x01), /* Roland VS-700 M2 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0100, 0x0d, 0x01), /* Roland VS-700 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0101, 0x0d, 0x01), /* Roland VS-700 M2 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0102, 0x0d, 0x01), /* Roland VB-99 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0109, 0x0d, 0x01), /* BOSS eBand JS-8 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0111, 0x0d, 0x01), /* Roland GAIA SH-01 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0113, 0x0d, 0x01), /* BOSS ME-25 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0114, 0x0d, 0x01), /* Roland SD-50 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0117, 0x0d, 0x01), /* Roland VS-20 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0119, 0x0d, 0x01), /* Roland OCTAPAD SPD-30 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x011c, 0x0d, 0x01), /* Roland Lucina AX-09 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x011e, 0x0d, 0x01), /* BOSS BR-800 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0120, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0121, 0x0d, 0x01), /* Roland OCTA-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0123, 0x0d, 0x01), /* Roland JUNO-Gi */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0124, 0x0d, 0x01), /* Roland M-300 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0127, 0x0d, 0x01), /* Roland GR-55 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x012b, 0x0d, 0x01), /* Roland DUO-CAPTURE */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x012f, 0x0d, 0x01), /* Roland QUAD-CAPTURE */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0132, 0x0d, 0x01), /* Roland TRI-CAPTURE */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0134, 0x0d, 0x01), /* Roland V-Mixer */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0137, 0x0d, 0x01), /* Roland DUO-CAPTURE Advanced Mode*/ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0138, 0x0d, 0x01), /* BOSS RC-300 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x013a, 0x0d, 0x01), /* Roland JUPITER-80 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x013e, 0x0d, 0x01), /* Roland R-26 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0145, 0x0d, 0x01), /* Roland SPD-SX */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x014b, 0x0d, 0x01), /* BOSS eBand JS-10 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x014d, 0x0d, 0x01), /* BOSS GT-100 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0150, 0x0d, 0x01), /* Roland TD-15 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0151, 0x0d, 0x01), /* Roland TD-11 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0154, 0x0d, 0x01), /* Roland JUPITER-50 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0158, 0x0d, 0x01), /* Roland TD-30 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x0159, 0x0d, 0x01), /* Roland DUO-CAPTURE EX */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x015b, 0x0d, 0x01), /* Roland INTEGRA-7 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x015d, 0x0d, 0x01), /* Roland R-88 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x017a, 0x0d, 0x01), /* Roland VT-3 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x017c, 0x0d, 0x01), /* Roland TR-8 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01b5, 0x0d, 0x01), /* Roland Boutique JP-08 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01fd, 0x0d, 0x01), /* Roland Boutique SH-01A */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01ff, 0x0d, 0x01), /* Roland Boutique D-05 */ IMPLICIT_FB_BOTH_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */

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