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

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: enable IIR filter
by Shunli Wang 20 May '19

20 May '19

This patch is to enable IIR filter when DMIC of PMIC is enabled. Signed-off-by: Shunli Wang <shunli.wang(a)mediatek.com> --- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 017d7d1d9148..2b758a18c2ea 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -176,9 +176,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; @@ -426,6 +423,17 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); + /* enable iir */ + ul_src_con0 |= (1 << UL_IIR_ON_TMP_CTL_SFT) & + UL_IIR_ON_TMP_CTL_MASK_SFT; + + /* 35Hz @ 48k */ + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_02_01, 0x00000000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_04_03, 0x00003FB8); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_06_05, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_08_07, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_10_09, 0x0000C048); + regmap_write(afe->regmap, AFE_ADDA_UL_SRC_CON0, ul_src_con0); /* mtkaif_rxif_data_mode = 0, amic */ -- 2.18.0

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[alsa-devel] [PATCH] ALSA: dice: add stream format parameters for PreSonus FireStudio
by Takashi Sakamoto 20 May '19

20 May '19

FireStudio was launched by PreSonus 2009. This model consists of three ICs for its packet processing on IEEE 1394 bus: - Texus Instruments TSB41AB2 for physical layer of IEEE 1394 bus - WaveFront semiconductor, Dice II STD ASIC for link layer of IEEE 1394 bus and protocol layer - Xilinx Spartan XG3S500E FPGA for signal processing This model don't support TCAT extended application protocol. For such devices, ALSA dice driver needs to have hard-coded parameters for stream formats. This commit adds hard-coded table for this model. As a result, sampling transfer frequencies of 88.2/96.0 kHz are supported. I note that this patch can be backported to Linux kernel v4.18 and later. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04042eda bus_info_length 4, crc_length 4, crc 11994 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000a9204 company_id 000a92 | 410 023a8b7f device_id 04023a8b7f | EUI-64 000a9204023a8b7f root directory ----------------------------------------------------------------- 414 000661b6 directory_length 6, crc 25014 418 03000a92 vendor 41c 8100000a --> descriptor leaf at 444 420 17000008 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 00041c75 directory_length 4, crc 7285 434 12000a92 specifier id 438 13000001 version 43c 17000008 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ----------------------------------------------------------------- 444 00047c11 leaf_length 4, crc 31761 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 50726553 "PreS" 454 6f6e7573 "onus" descriptor leaf at 458 ----------------------------------------------------------------- 458 0005d7b3 leaf_length 5, crc 55219 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 46495245 "FIRE" 468 53545544 "STUD" 46c 494f0000 "IO" descriptor leaf at 470 ----------------------------------------------------------------- 470 0005d7b3 leaf_length 5, crc 55219 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 46495245 "FIRE" 480 53545544 "STUD" 484 494f0000 "IO" Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-presonus.c | 62 +++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 9 +++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 73 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-presonus.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 37062a233f6a..7efca3648c6a 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,4 +1,4 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ - dice-alesis.o dice-extension.o dice-mytek.o + dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-presonus.c b/sound/firewire/dice/dice-presonus.c new file mode 100644 index 000000000000..503f462a83f4 --- /dev/null +++ b/sound/firewire/dice/dice-presonus.c @@ -0,0 +1,62 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-presonus.c - a part of driver for DICE based devices +// +// Copyright (c) 2019 Takashi Sakamoto +// +// Licensed under the terms of the GNU General Public License, version 2. + +#include "dice.h" + +struct dice_presonus_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +static const struct dice_presonus_spec dice_presonus_firesutio = { + .tx_pcm_chs = {{16, 16, 0}, {10, 2, 0} }, + .rx_pcm_chs = {{16, 16, 0}, {10, 2, 0} }, + .has_midi = true, +}; + +int snd_dice_detect_presonus_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_presonus_spec *spec; + } *entry, entries[] = { + {0x000008, &dice_presonus_firesutio}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index eee184b05d93..ac600e061d7b 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -19,6 +19,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_MAUDIO 0x000d6c #define OUI_MYTEK 0x001ee8 #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. +#define OUI_PRESONUS 0x000a92 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 @@ -371,6 +372,14 @@ static const struct ieee1394_device_id dice_id_table[] = { .vendor_id = OUI_SSL, .model_id = 0x000070, }, + // Presonus FireStudio. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_PRESONUS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_presonus_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 83353a3559e8..9699adc2a96d 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -227,5 +227,6 @@ int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice); int snd_dice_detect_alesis_formats(struct snd_dice *dice); int snd_dice_detect_extension_formats(struct snd_dice *dice); int snd_dice_detect_mytek_formats(struct snd_dice *dice); +int snd_dice_detect_presonus_formats(struct snd_dice *dice); #endif -- 2.20.1

2 1

[alsa-devel] [asoc:for-5.3 34/45] sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits'
by kbuild test robot 20 May '19

20 May '19

Hi Shunli, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 head: bfa8130f50a63563eae10ef933fe01b50b3e87a0 commit: 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d [34/45] ASoC: Mediatek: add memory interface data align config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/mediatek/common/mtk-afe-fe-dai.c: In function 'mtk_afe_fe_prepare': >> sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits' mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/common/mtk-afe-fe-dai.c:19:12: note: declared here static int mtk_regmap_update_bits(struct regmap *map, int reg, ^~~~~~~~~~~~~~~~~~~~~~ vim +/mtk_regmap_update_bits +268 sound/soc/mediatek/common/mtk-afe-fe-dai.c 236 237 int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, 238 struct snd_soc_dai *dai) 239 { 240 struct snd_soc_pcm_runtime *rtd = substream->private_data; 241 struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); 242 struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; 243 int hd_audio = 0; 244 int hd_align = 1; 245 246 /* set hd mode */ 247 switch (substream->runtime->format) { 248 case SNDRV_PCM_FORMAT_S16_LE: 249 hd_audio = 0; 250 break; 251 case SNDRV_PCM_FORMAT_S32_LE: 252 hd_audio = 1; 253 hd_align = 1; 254 break; 255 case SNDRV_PCM_FORMAT_S24_LE: 256 hd_audio = 1; 257 hd_align = 0; 258 break; 259 default: 260 dev_err(afe->dev, "%s() error: unsupported format %d\n", 261 __func__, substream->runtime->format); 262 break; 263 } 264 265 mtk_regmap_update_bits(afe->regmap, memif->data->hd_reg, 266 1, hd_audio, memif->data->hd_shift); 267 > 268 mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, 269 memif->data->hd_align_mshift, 270 hd_align ? memif->data->hd_align_mshift : 0); 271 272 return 0; 273 } 274 EXPORT_SYMBOL_GPL(mtk_afe_fe_prepare); 275 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH][RFC] ASoC: soc-pcm: fixup try_module_get() calling timing
by Kuninori Morimoto 20 May '19

20 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_open() try to call try_module_get() based on component->driver->module_get_upon_open. But, it should be always called, not relatead to .open callback. It should be called at (A) istead of (B) => (A) if (!component->driver->ops || !component->driver->ops->open) continue; => (B) if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { ... } ret = component->driver->ops->open(substream); Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy if someone can confirm it. sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7737e00..7b4cda6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -440,10 +440,6 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, component = rtdcom->component; *last = component; - if (!component->driver->ops || - !component->driver->ops->open) - continue; - if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { dev_err(component->dev, @@ -452,6 +448,10 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, return -ENODEV; } + if (!component->driver->ops || + !component->driver->ops->open) + continue; + ret = component->driver->ops->open(substream); if (ret < 0) { dev_err(component->dev, -- 2.7.4

3 4

[alsa-devel] [PATCH][RFC] ASoC: soc-pcm: fixup module_put() calling timing
by Kuninori Morimoto 18 May '19

18 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_close() try to call module_put() based on component->driver->module_get_upon_open. But, it should be called even though it doesn't have .close callback. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy if someone can confirm it. sound/soc/soc-pcm.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7b4cda6..e24eab3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -477,11 +477,9 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, if (component == last) break; - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); + if (component->driver->ops && + component->driver->ops->close) + component->driver->ops->close(substream); if (component->driver->module_get_upon_open) module_put(component->dev->driver->owner); -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: fix typos in code comments
by Weitao Hou 18 May '19

18 May '19

fix lenght to length Signed-off-by: Weitao Hou <houweitaoo(a)gmail.com> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4f85cb19a309..e8141a33a55e 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1194,7 +1194,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * q6asm_write_async() - non blocking write * * @ac: audio client pointer - * @len: lenght in bytes + * @len: length in bytes * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw * @wflags: flags associated with write -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: fsl_esai: fix the channel swap issue after xrun
by S.j. Wang 17 May '19

17 May '19

There is chip errata ERR008000, the reference doc is (https://www.nxp.com/docs/en/errata/IMX6DQCE.pdf), The issue is "While using ESAI transmit or receive and an underrun/overrun happens, channel swap may occur. The only recovery mechanism is to reset the ESAI." In this commit add a tasklet to handle reset of ESAI after xrun happens Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 166 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 166 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 10d2210c91ef..149972894c95 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -52,17 +52,20 @@ struct fsl_esai { struct clk *extalclk; struct clk *fsysclk; struct clk *spbaclk; + struct tasklet_struct task; u32 fifo_depth; u32 slot_width; u32 slots; u32 tx_mask; u32 rx_mask; + u32 tx_channels; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; bool sck_div[2]; bool slave_mode; bool synchronous; + bool reset_at_xrun; char name[32]; }; @@ -71,8 +74,14 @@ static irqreturn_t esai_isr(int irq, void *devid) struct fsl_esai *esai_priv = (struct fsl_esai *)devid; struct platform_device *pdev = esai_priv->pdev; u32 esr; + u32 saisr; regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr); + + if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) + && esai_priv->reset_at_xrun) + tasklet_schedule(&esai_priv->task); if (esr & ESAI_ESR_TINIT_MASK) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); @@ -552,6 +561,9 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 mask; + if (tx) + esai_priv->tx_channels = channels; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -585,10 +597,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), @@ -618,6 +636,145 @@ static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; +static void fsl_esai_reset(unsigned long arg) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)arg; + u32 saisr; + u32 tsma, tsmb, rsma, rsmb, tcr, rcr, tfcr, rfcr; + int i; + + /* + * stop the tx & rx + */ + regmap_read(esai_priv->regmap, REG_ESAI_TSMA, &tsma); + regmap_read(esai_priv->regmap, REG_ESAI_TSMB, &tsmb); + regmap_read(esai_priv->regmap, REG_ESAI_RSMA, &rsma); + regmap_read(esai_priv->regmap, REG_ESAI_RSMB, &rsmb); + + regmap_read(esai_priv->regmap, REG_ESAI_TCR, &tcr); + regmap_read(esai_priv->regmap, REG_ESAI_RCR, &rcr); + + regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); + regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK | ESAI_xCR_TE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK | ESAI_xCR_RE_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMB_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMB_xS_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR, 0); + + /* + * reset the esai, and restore the registers + */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK, + ESAI_ECR_ESAIEN | ESAI_ECR_ERST); + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK, + ESAI_ECR_ESAIEN); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xPR_MASK, + ESAI_xCR_xPR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xPR_MASK, + ESAI_xCR_xPR); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, 0); + + /* + * Add fifo reset here, because the regcache_sync will + * write one more data to ETDR. + * Which will cause channel shift. + */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + regcache_mark_dirty(esai_priv->regmap); + regcache_sync(esai_priv->regmap); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xPR_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xPR_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, + ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, + ESAI_PCRC_PC(ESAI_GPIO)); + + regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr); + + /* + * restart tx / rx, if they already enabled + */ + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFEN_MASK, tfcr & ESAI_xFCR_xFEN); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFEN_MASK, rfcr & ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; i < esai_priv->tx_channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_TE_MASK, + ESAI_xCR_TE_MASK & tcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_RE_MASK, + ESAI_xCR_RE_MASK & rcr); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMB_xS_MASK, + ESAI_xSMB_xS_MASK & tsmb); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, + ESAI_xSMA_xS_MASK & tsma); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMB_xS_MASK, + ESAI_xSMB_xS_MASK & rsmb); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, + ESAI_xSMA_xS_MASK & rsma); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE & tcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE & rcr); +} + static int fsl_esai_dai_probe(struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); @@ -787,6 +944,10 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); + if (of_device_is_compatible(np, "fsl,vf610-esai") || + of_device_is_compatible(np, "fsl,imx35-esai")) + esai_priv->reset_at_xrun = true; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); @@ -899,6 +1060,8 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + tasklet_init(&esai_priv->task, fsl_esai_reset, (unsigned long)esai_priv); + pm_runtime_enable(&pdev->dev); regcache_cache_only(esai_priv->regmap, true); @@ -912,7 +1075,10 @@ static int fsl_esai_probe(struct platform_device *pdev) static int fsl_esai_remove(struct platform_device *pdev) { + struct fsl_esai *esai_priv = platform_get_drvdata(pdev); + pm_runtime_disable(&pdev->dev); + tasklet_kill(&esai_priv->task); return 0; } -- 2.21.0

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[alsa-devel] [PATCH] sound: open brace should be on the previous line
by parna.naveenkumar＠gmail.com 17 May '19

17 May '19

From: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> Resolved open brace { should be on the previous line checkpatch.pl error. While addressing this error, also corrected the affected code for below mentioned checkpatch errors. ERROR: spaces required around that '<' (ctx:VxV) ERROR: spaces required around that '==' (ctx:VxV) ERROR: space required before the open parenthesis '(' Signed-off-by: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> --- sound/sound_core.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 40ad000c2e3c..ce794a2afc6b 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -111,8 +111,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,8 +150,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -173,8 +171,7 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, while (*list && (*list)->unit_minor<n) list=&((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ if(*list==NULL || (*list)->unit_minor>n) break; @@ -219,11 +216,9 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) { - while(*list) - { + while (*list) { struct sound_unit *p=*list; - if(p->unit_minor==unit) - { + if (p->unit_minor == unit) { *list=p->next; return p; } @@ -528,8 +523,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit) struct sound_unit *s; s=chains[chain]; - while(s && s->unit_minor <= unit) - { + while (s && s->unit_minor <= unit) { if(s->unit_minor==unit) return s; s=s->next; @@ -545,8 +539,7 @@ static int soundcore_open(struct inode *inode, struct file *file) const struct file_operations *new_fops = NULL; chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ unit&=0xF0; unit|=3; chain=3; -- 2.17.1

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[alsa-devel] [PATCH V2] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 17 May '19

17 May '19

Handle the reset GPIO and reset the device every time we start it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio sound/soc/codecs/cs42xx8.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 28a4ac36c4f8..b377cddaf2e6 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; u32 tx_channels; + struct gpio_desc *gpiod_reset; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -467,6 +469,13 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) return -EINVAL; } + cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(cs42xx8->gpiod_reset)) + return PTR_ERR(cs42xx8->gpiod_reset); + + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + cs42xx8->clk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs42xx8->clk)) { dev_err(dev, "failed to get the clock: %ld\n", @@ -547,6 +556,8 @@ static int cs42xx8_runtime_resume(struct device *dev) return ret; } + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); if (ret) { @@ -586,6 +597,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 1); + clk_disable_unprepare(cs42xx8->clk); return 0; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..28a4ac36c4f8 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev) msleep(5); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { -- 2.21.0

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