September 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v7] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 01 Oct '19

01 Oct '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

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[alsa-devel] [PATCH] ASoC: stm32: sai: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: wcd9335: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd9335.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: rt5663: clean up indentation issues
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } -- 2.20.1

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[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

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[alsa-devel] [PATCH] ASoC: core: Clarify usage of ignore_machine
by Daniel Baluta 01 Oct '19

01 Oct '19

For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..e32a45f6bd88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1853,7 +1853,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1874,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ -- 2.17.1

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[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 1/5] ASoC: amd: Registering device endpoints using MFD framework
by Ravulapati Vishnu vardhan rao 30 Sep '19

30 Sep '19

Removed platform based endpoint registering. Now Registering DMA and multiple I2S BT playback endpoint devices automatically by using MFD framework. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Reviewed-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 54 ++++++++------- sound/soc/amd/raven/pci-acp3x.c | 148 +++++++++++++++++++++++++++++----------- 2 files changed, 139 insertions(+), 63 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..3a27fc8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,33 +7,41 @@ #include "chip_offset_byte.h" -#define ACP3x_PHY_BASE_ADDRESS 0x1240000 -#define ACP3x_I2S_MODE 0 -#define ACP3x_REG_START 0x1240000 -#define ACP3x_REG_END 0x1250200 -#define I2S_MODE 0x04 -#define BT_TX_THRESHOLD 26 -#define BT_RX_THRESHOLD 25 -#define ACP3x_POWER_ON 0x00 -#define ACP3x_POWER_ON_IN_PROGRESS 0x01 -#define ACP3x_POWER_OFF 0x02 -#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_DEVS 3 +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 +#define I2S_MODE 0x04 +#define I2S_RX_THRESHOLD 27 +#define I2S_TX_THRESHOLD 28 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP_ERR_INTR_MASK 29 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 #define ACP_SRAM_PTE_OFFSET 0x02050000 -#define PAGE_SIZE_4K_ENABLE 0x2 +#define PAGE_SIZE_4K_ENABLE 0x2 #define MEM_WINDOW_START 0x4000000 -#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 -#define CAPTURE_FIFO_ADDR_OFFSET 0x500 - -#define PLAYBACK_MIN_NUM_PERIODS 2 -#define PLAYBACK_MAX_NUM_PERIODS 8 -#define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 4096 -#define CAPTURE_MIN_NUM_PERIODS 2 -#define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 16384 -#define CAPTURE_MIN_PERIOD_SIZE 4096 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..c0b8fad 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -9,23 +9,49 @@ #include <linux/io.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/mfd/core.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> #include "acp3x.h" +struct i2s_platform_data { + unsigned int cap; + int channel; + u32 snd_rates; +}; struct acp3x_dev_data { + struct device *parent; + struct mfd_cell *cell; + struct resource *res; void __iomem *acp3x_base; bool acp3x_audio_mode; - struct resource *res; struct platform_device *pdev; }; + +static struct device *get_mfd_cell_dev(const char *device_name, int r) +{ + char auto_dev_name[25]; + struct device *dev; + + snprintf(auto_dev_name, sizeof(auto_dev_name), + "%s.%d.auto", device_name, r); + dev = bus_find_device_by_name(&platform_bus_type, + NULL, auto_dev_name); + dev_info(dev, "device %s added\n", auto_dev_name); + return dev; +} + static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + uint64_t addr; + int val, i, r; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct device *dev; + struct i2s_platform_data *i2s_pdata; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -63,47 +89,82 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - + adata->parent = &pci->dev; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, - GFP_KERNEL); - if (!adata->res) { + sizeof(struct resource) * 4, + GFP_KERNEL); + adata->cell = devm_kzalloc(&pci->dev, + sizeof(struct mfd_cell) * 3, + GFP_KERNEL); + if (!adata->cell) { ret = -ENOMEM; goto unmap_mmio; } - adata->res[0].name = "acp3x_i2s_iomem"; - adata->res[0].flags = IORESOURCE_MEM; - adata->res[0].start = addr; - adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; - - adata->acp3x_audio_mode = ACP3x_I2S_MODE; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + i2s_pdata = devm_kzalloc(&pci->dev, + sizeof(struct i2s_platform_data) * 3, + GFP_KERNEL); + if (i2s_pdata == NULL) { + kfree(adata->res); + kfree(adata->cell); + return -ENOMEM; } + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + + (ACP3x_REG_END - ACP3x_REG_START); + i2s_pdata[0].cap = 0; + i2s_pdata[0].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[1].name = "acp3x_i2s_sp_play_cap"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + i2s_pdata[1].cap = 0; + i2s_pdata[1].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[2].name = "acp3x_i2s_bt_play_cap"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + i2s_pdata[2].cap = 0; + i2s_pdata[2].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + adata->cell[0].name = "acp3x_rv_i2s_dma"; + adata->cell[0].num_resources = 4; + adata->cell[0].resources = &adata->res[0]; + adata->cell[0].platform_data = &irqflags; + adata->cell[0].pdata_size = sizeof(irqflags); + + adata->cell[1].name = "acp3x_i2s_playcap"; + adata->cell[1].num_resources = 1; + adata->cell[1].resources = &adata->res[1]; + adata->cell[1].platform_data = &i2s_pdata[0]; + adata->cell[1].pdata_size = + sizeof(struct i2s_platform_data); + + adata->cell[2].name = "acp3x_i2s_playcap"; + adata->cell[2].num_resources = 1; + adata->cell[2].resources = &adata->res[2]; + adata->cell[2].platform_data = &i2s_pdata[1]; + adata->cell[2].pdata_size = + sizeof(struct i2s_platform_data); + r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); + for (i = 0; i < 3 ; i++) + dev = get_mfd_cell_dev(adata->cell[i].name, i); break; + default: dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; @@ -112,23 +173,29 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); + /*ignore device status and return driver probe error*/ + return -ENODEV; release_regions: pci_release_regions(pci); + /*ignore device status and return driver probe error*/ + return -ENODEV; disable_pci: pci_disable_device(pci); - - return ret; + /*ignore device status and return driver probe error*/ + return -ENODEV; } static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); - - platform_device_unregister(adata->pdev); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -151,6 +218,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

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Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 28 Sep '19

28 Sep '19

On 9/26/19 9:55 PM, Navid Emamdoost wrote: > On Wed, Sep 25, 2019 at 12:05:28PM -0500, Pierre-Louis Bossart wrote: >> On 9/25/19 11:19 AM, Navid Emamdoost wrote: >>> In snd_skl_parse_uuids if allocation for module->instance_id fails, the >>> allocated memory for module shoulde be released. I changes the >>> allocation for module to use devm_kzalloc to be resource_managed >>> allocation and avoid the release in error path. >> >> if you use devm_, don't you need to fix the error path as well then, I see a >> kfree(uuid) in skl_freeup_uuid_list(). >> >> I am not very familiar with this code but the error seems to be that the >> list_add_tail() is called after the module->instance_id is allocated, so >> there is a risk that the module allocated earlier is not freed (since it's >> not yet added to the list). Freeing the module as done in patch 1 works, >> using devm_ without fixing the error path does not seem correct to me. >> > Thanks for the feedback, then it's your call if you can accept patch 1 as > fix. Cezary, it's really your call. > Navid. >>> >>> Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> >>> --- >>> Changes in v2: >>> - Changed the allocation for module from kzalloc to devm_kzalloc >>> --- >>> sound/soc/intel/skylake/skl-sst-utils.c | 2 +- >>> 1 file changed, 1 insertion(+), 1 deletion(-) >>> >>> diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c >>> index d43cbf4a71ef..ac37f04b0eea 100644 >>> --- a/sound/soc/intel/skylake/skl-sst-utils.c >>> +++ b/sound/soc/intel/skylake/skl-sst-utils.c >>> @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, >>> */ >>> for (i = 0; i < num_entry; i++, mod_entry++) { >>> - module = kzalloc(sizeof(*module), GFP_KERNEL); >>> + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); >>> if (!module) { >>> ret = -ENOMEM; >>> goto free_uuid_list; >>> >>

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