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

[alsa-devel] [PATCH 0/3] i.MX8M support for FSL SAI
by Lucas Stach 22 Jul '19

22 Jul '19

Hi all, those are the changes that are absolutely needed to support the i.MX8M SAI block. The most noticable difference is the moved register layout, due to the hardware guys adding two version registers at the start of the register map and considerable bigger FIFO sizes. Downstream has a lot more changes to support new features, but those 3 patches at least allows me to play back audio on i.MX8M based systems. Regards, Lucas Lucas Stach (3): ASoC: fsl_sai: add of_match data ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth ASoC: fsl_sai: add i.MX8M support .../devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 45 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 8 +++- 3 files changed, 44 insertions(+), 11 deletions(-) -- 2.20.1

5 19

[alsa-devel] [PATCH] SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection
by Enric Balletbo i Serra 22 Jul '19

22 Jul '19

The TS3A227E says that the headset keypress detection needs the MICBIAS power in order to report the key events to ensure proper operation The headset keypress detection needs the MICBIAS power in order to report the key events all the time as long as MIC is present. So MICBIAS pin is forced on when a MICROPHONE is detected. On Veyron Minnie I observed that if the MICBIAS power is not present and the key press detection is activated (just because it is enabled when you insert a headset), it randomly reports a keypress on insert. E.g. (KEY_PLAYPAUSE) Event: (SW_HEADPHONE_INSERT), value 1 Event: (SW_MICROPHONE_INSERT), value 1 Event: -------------- SYN_REPORT ------------ Event: (KEY_PLAYPAUSE), value 1 Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying effect that the media player starts a play/pause loop. Note that, although most of the time the key reported is the one associated with BTN_0, not always this is true. On my tests I also saw different keys reported Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- Some notes about the patch Steps to test (userspace GNOME3): 1. Play audio using a media player 2. Make sure the Internal MIC is selected as audio input device and doesn't switches when you insert the headset. 3. Insert a headset (with buttons) Audio switches to headphones and you can hear that enters on a loop play/pause if the KEY_PLAYPAUSE was reported. Also you can check that press to any headset button doesn't work. The part where the datasheet says that power must be supplied in order to have keypress detection work is in page 44 of [1] " The TS3A227E can monitor the microphone line of a 4-pole headset to detect up to 4 key presses/releases and report the key press events back to the host. The key press detection must be activated manually by setting the KP Enable bit of the Device Settings 2 register. To ensure proper operation the MICBIAS voltage must be applied to MICP before enabling key press detection. " [1] http://www.ti.com/lit/ds/symlink/ts3a227e.pdf sound/soc/rockchip/rockchip_max98090.c | 32 ++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index c5fc24675a33..782e534d4c0d 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; +static int rk_jack_event(struct notifier_block *nb, unsigned long event, + void *data) +{ + struct snd_soc_jack *jack = (struct snd_soc_jack *)data; + struct snd_soc_dapm_context *dapm = &jack->card->dapm; + + if (event & SND_JACK_MICROPHONE) + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + else + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct notifier_block rk_jack_nb = { + .notifier_call = rk_jack_event, +}; + +static int rk_init(struct snd_soc_pcm_runtime *runtime) +{ + /* + * The jack has already been created in the rk_98090_headset_init() + * function. + */ + snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb); + + return 0; +} + static int rk_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi, static struct snd_soc_dai_link rk_dailink = { .name = "max98090", .stream_name = "Audio", + .init = rk_init, .ops = &rk_aif1_ops, /* set max98090 as slave */ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks
by Charles Keepax 22 Jul '19

22 Jul '19

DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24c050a1..f73882cf0031d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, list_add_tail(&widget->work_list, list); if (custom_stop_condition && custom_stop_condition(widget, dir)) { - widget->endpoints[dir] = 1; - return widget->endpoints[dir]; + list = NULL; + custom_stop_condition = NULL; } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { @@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, * * Optionally, can be supplied with a function acting as a stopping condition. * This function takes the dapm widget currently being examined and the walk - * direction as an arguments, it should return true if the walk should be - * stopped and false otherwise. + * direction as an arguments, it should return true if widgets from that point + * in the graph onwards should not be added to the widget list. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: cs42xx8: Fix MFREQ selection issue for async mode
by shengjiu.wang＠nxp.com 22 Jul '19

22 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> When sample rate of TX is different with sample rate of RX in async mode, the MFreq selection will be wrong. For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz. Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX and RX instance, the correct value of MFreq is 2 for both TX and RX. But original method will cause MFreq = 0 for TX, MFreq = 2 for RX. If TX is started after RX, RX will be impacted, RX work abnormal with MFreq = 0. This patch is to select proper MFreq value according to TX rate and RX rate. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++++++++++++++++------ 1 file changed, 97 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index b377cddaf2e6..804969495754 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -47,6 +47,7 @@ struct cs42xx8_priv { unsigned long sysclk; u32 tx_channels; struct gpio_desc *gpiod_reset; + u32 rate[2]; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { }; struct cs42xx8_ratios { - unsigned int ratio; - unsigned char speed; - unsigned char mclk; + unsigned int mfreq; + unsigned int min_mclk; + unsigned int max_mclk; + unsigned int ratio[3]; }; +/* + * According to reference mannual, define the cs42xx8_ratio struct + * MFreq2 | MFreq1 | MFreq0 | Description | SSM | DSM | QSM | + * 0 | 0 | 0 |1.029MHz to 12.8MHz | 256 | 128 | 64 | + * 0 | 0 | 1 |1.536MHz to 19.2MHz | 384 | 192 | 96 | + * 0 | 1 | 0 |2.048MHz to 25.6MHz | 512 | 256 | 128 | + * 0 | 1 | 1 |3.072MHz to 38.4MHz | 768 | 384 | 192 | + * 1 | x | x |4.096MHz to 51.2MHz |1024 | 512 | 256 | + */ static const struct cs42xx8_ratios cs42xx8_ratios[] = { - { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, - { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, - { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, - { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, - { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, - { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, - { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, - { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, - { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } + { 0, 1029000, 12800000, {256, 128, 64} }, + { 2, 1536000, 19200000, {384, 192, 96} }, + { 4, 2048000, 25600000, {512, 256, 128} }, + { 6, 3072000, 38400000, {768, 384, 192} }, + { 8, 4096000, 51200000, {1024, 512, 256} }, }; static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 ratio = cs42xx8->sysclk / params_rate(params); - u32 i, fm, val, mask; + u32 ratio[2]; + u32 rate[2]; + u32 fm[2]; + u32 i, val, mask; + bool condition1, condition2; if (tx) cs42xx8->tx_channels = params_channels(params); + rate[tx] = params_rate(params); + rate[!tx] = cs42xx8->rate[!tx]; + + ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0; + ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0; + + /* Get functional mode for tx and rx according to rate */ + for (i = 0; i < 2; i++) { + if (cs42xx8->slave_mode) { + fm[i] = CS42XX8_FM_AUTO; + } else { + if (rate[i] < 50000) { + fm[i] = CS42XX8_FM_SINGLE; + } else if (rate[i] > 50000 && rate[i] < 100000) { + fm[i] = CS42XX8_FM_DOUBLE; + } else if (rate[i] > 100000 && rate[i] < 200000) { + fm[i] = CS42XX8_FM_QUAD; + } else { + dev_err(component->dev, + "unsupported sample rate\n"); + return -EINVAL; + } + } + } + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { - if (cs42xx8_ratios[i].ratio == ratio) + /* Is the ratio[tx] valid ? */ + condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[tx] || + cs42xx8_ratios[i].ratio[1] == ratio[tx] || + cs42xx8_ratios[i].ratio[2] == ratio[tx]) : + (cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) && + cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk && + cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk; + + if (!ratio[tx]) + condition1 = true; + + /* Is the ratio[!tx] valid ? */ + condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[!tx] || + cs42xx8_ratios[i].ratio[1] == ratio[!tx] || + cs42xx8_ratios[i].ratio[2] == ratio[!tx]) : + (cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx])); + + if (!ratio[!tx]) + condition2 = true; + + /* + * Both ratio[tx] and ratio[!tx] is valid, then we get + * a proper MFreq. + */ + if (condition1 && condition2) break; } @@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - mask = CS42XX8_FUNCMOD_MFREQ_MASK; - val = cs42xx8_ratios[i].mclk; + cs42xx8->rate[tx] = params_rate(params); - fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mfreq; regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, - CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val); + + return 0; +} + +static int cs42xx8_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + /* Clear stored rate */ + cs42xx8->rate[tx] = 0; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx), + CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO)); return 0; } @@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = { .set_fmt = cs42xx8_set_dai_fmt, .set_sysclk = cs42xx8_set_dai_sysclk, .hw_params = cs42xx8_hw_params, + .hw_free = cs42xx8_hw_free, .digital_mute = cs42xx8_digital_mute, }; -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing
by Codrin Ciubotariu 22 Jul '19

22 Jul '19

Using regmap_multi_reg_write() when we set the default values for our registers makes the code smaller and easier to read. Suggested-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/ad193x.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 80dab5df9633..fb04c9379b71 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,15 +413,10 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .ops = &ad193x_dai_ops, }; -struct ad193x_reg_default { - unsigned int reg; - unsigned int val; -}; - /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct ad193x_reg_default reg_init[] = { + const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -437,21 +432,17 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct ad193x_reg_default reg_adc_init[] = { + const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ }; - int i; - for (i = 0; i < ARRAY_SIZE(reg_init); i++) - regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val); + regmap_multi_reg_write(ad193x->regmap, reg_init, ARRAY_SIZE(reg_init)); if (ad193x_has_adc(ad193x)) { - for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) { - regmap_write(ad193x->regmap, reg_adc_init[i].reg, - reg_adc_init[i].val); - } + regmap_multi_reg_write(ad193x->regmap, reg_adc_init, + ARRAY_SIZE(reg_adc_init)); } } -- 2.20.1

3 2

[alsa-devel] [PATCH 00/12] treewide: Fix GENMASK misuses
by Joe Perches 22 Jul '19

22 Jul '19

These GENMASK uses are inverted argument order and the actual masks produced are incorrect. Fix them. Add checkpatch tests to help avoid more misuses too. Joe Perches (12): checkpatch: Add GENMASK tests clocksource/drivers/npcm: Fix misuse of GENMASK macro drm: aspeed_gfx: Fix misuse of GENMASK macro iio: adc: max9611: Fix misuse of GENMASK macro irqchip/gic-v3-its: Fix misuse of GENMASK macro mmc: meson-mx-sdio: Fix misuse of GENMASK macro net: ethernet: mediatek: Fix misuses of GENMASK macro net: stmmac: Fix misuses of GENMASK macro rtw88: Fix misuse of GENMASK macro phy: amlogic: G12A: Fix misuse of GENMASK macro staging: media: cedrus: Fix misuse of GENMASK macro ASoC: wcd9335: Fix misuse of GENMASK macro drivers/clocksource/timer-npcm7xx.c | 2 +- drivers/gpu/drm/aspeed/aspeed_gfx.h | 2 +- drivers/iio/adc/max9611.c | 2 +- drivers/irqchip/irq-gic-v3-its.c | 2 +- drivers/mmc/host/meson-mx-sdio.c | 2 +- drivers/net/ethernet/mediatek/mtk_eth_soc.h | 2 +- drivers/net/ethernet/mediatek/mtk_sgmii.c | 2 +- drivers/net/ethernet/stmicro/stmmac/descs.h | 2 +- drivers/net/ethernet/stmicro/stmmac/dwmac-sun8i.c | 4 ++-- drivers/net/wireless/realtek/rtw88/rtw8822b.c | 2 +- drivers/phy/amlogic/phy-meson-g12a-usb2.c | 2 +- drivers/staging/media/sunxi/cedrus/cedrus_regs.h | 2 +- scripts/checkpatch.pl | 15 +++++++++++++++ sound/soc/codecs/wcd-clsh-v2.c | 2 +- 14 files changed, 29 insertions(+), 14 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH v2] ASoC: SOF: Intel: hda: Make hdac_device device-managed
by Ranjani Sridharan 22 Jul '19

22 Jul '19

snd_hdac_ext_bus_device_exit() has been recently modified to no longer free the hdac device. SOF allocates memory for hdac_device and hda_hda_priv with kzalloc. Make them device-managed instead so that they will be freed when the SOF driver is unloaded. Because of the above change, hda_codec is device-managed and it will be freed when the ASoC device is removed. Freeing the codec in snd_hda_codec_dev_release() leads to kernel panic while unloading and reloading the ASoC driver. So, avoid freeing the hda_codec for ASoC driver. This is done in the same patch to avoid bisect failure. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- sound/soc/sof/intel/hda-codec.c | 6 ++---- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6c51b8363f8b..a2e23d7e768f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -846,7 +846,13 @@ static void snd_hda_codec_dev_release(struct device *dev) snd_hda_sysfs_clear(codec); kfree(codec->modelname); kfree(codec->wcaps); - kfree(codec); + + /* + * In the case of ASoC HD-audio, hda_codec is device managed. + * It will be freed when the ASoC device is removed. + */ + if (codec->core.type == HDA_DEV_LEGACY) + kfree(codec); } #define DEV_NAME_LEN 31 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index b8b37f082309..0d8437b080bf 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -62,8 +62,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) address, resp); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hda_priv = kzalloc(sizeof(*hda_priv), GFP_KERNEL); + hda_priv = devm_kzalloc(sdev->dev, sizeof(*hda_priv), GFP_KERNEL); if (!hda_priv) return -ENOMEM; @@ -82,8 +81,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address) return 0; #else - /* snd_hdac_ext_bus_device_exit will use kfree to free hdev */ - hdev = kzalloc(sizeof(*hdev), GFP_KERNEL); + hdev = devm_kzalloc(sdev->dev, sizeof(*hdev), GFP_KERNEL); if (!hdev) return -ENOMEM; -- 2.17.1

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[alsa-devel] [Patch v2] sound: soc: codecs: wcd9335: add irqflag IRQF_ONESHOT flag
by Hariprasad Kelam 22 Jul '19

22 Jul '19

Add IRQF_ONESHOT to ensure "Interrupt is not reenabled after the hardirq handler finished". fixes below issue reported by coccicheck sound/soc/codecs/wcd9335.c:4068:8-33: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- v1 : add IRQF_ONESHOT and introduce irqflags variable v2 : remove irqflags variable sound/soc/codecs/wcd9335.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1bbbe42..9566027 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4062,7 +4062,8 @@ static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) ret = devm_request_threaded_irq(wcd->dev, irq, NULL, wcd9335_irqs[i].handler, - IRQF_TRIGGER_RISING, + IRQF_TRIGGER_RISING | + IRQF_ONESHOT, wcd9335_irqs[i].name, wcd); if (ret) { dev_err(wcd->dev, "Failed to request %s\n", -- 2.7.4

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[alsa-devel] [PATCH] sound: soc: bcm: cygnus-pcm: Unneeded variable: "ret".
by Hariprasad Kelam 22 Jul '19

22 Jul '19

This patch fixes below issues reported by coccicheck sound/soc/bcm/cygnus-pcm.c:642:5-8: Unneeded variable: "ret". Return "0" on line 650 sound/soc/bcm/cygnus-pcm.c:671:5-8: Unneeded variable: "ret". Return "0" on line 696 We cannot change return type of these functions as they are callback functions of snd_pcm_ops Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/bcm/cygnus-pcm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 123ecf5..8966b02 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -639,7 +639,6 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct cygnus_aio_port *aio; - int ret = 0; aio = cygnus_dai_get_dma_data(substream); dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum); @@ -647,7 +646,7 @@ static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - return ret; + return 0; } static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream) @@ -668,7 +667,6 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct cygnus_aio_port *aio; unsigned long bufsize, periodsize; - int ret = 0; bool is_play; u32 start; struct ringbuf_regs *p_rbuf = NULL; @@ -693,7 +691,7 @@ static int cygnus_pcm_prepare(struct snd_pcm_substream *substream) ringbuf_set_initial(aio->cygaud->audio, p_rbuf, is_play, start, periodsize, bufsize); - return ret; + return 0; } static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream) -- 2.7.4

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[alsa-devel] [PATCH] sound: soc: codecs: mt6358: change return type of mt6358_codec_init_reg
by Hariprasad Kelam 22 Jul '19

22 Jul '19

As mt6358_codec_init_reg function always returns 0 , change return type from int to void. fixes below issue reported by coccicheck sound/soc/codecs/mt6358.c:2260:5-8: Unneeded variable: "ret". Return "0" on line 2289 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/soc/codecs/mt6358.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/mt6358.c b/sound/soc/codecs/mt6358.c index 50b3fc5..c17250a 100644 --- a/sound/soc/codecs/mt6358.c +++ b/sound/soc/codecs/mt6358.c @@ -2255,10 +2255,8 @@ static struct snd_soc_dai_driver mt6358_dai_driver[] = { }, }; -static int mt6358_codec_init_reg(struct mt6358_priv *priv) +static void mt6358_codec_init_reg(struct mt6358_priv *priv) { - int ret = 0; - /* Disable HeadphoneL/HeadphoneR short circuit protection */ regmap_update_bits(priv->regmap, MT6358_AUDDEC_ANA_CON0, RG_AUDHPLSCDISABLE_VAUDP15_MASK_SFT, @@ -2285,8 +2283,6 @@ static int mt6358_codec_init_reg(struct mt6358_priv *priv) /* set gpio */ playback_gpio_reset(priv); capture_gpio_reset(priv); - - return ret; } static int mt6358_codec_probe(struct snd_soc_component *cmpnt) -- 2.7.4

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