January 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "IIO: ADC: add stm32 DFSDM support for PDM microphone" to the asoc tree
by Mark Brown 10 Jan '18

10 Jan '18

The patch IIO: ADC: add stm32 DFSDM support for PDM microphone has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eca949800d2dd761bc0c73b793e4e9ad4a997469 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Date: Wed, 10 Jan 2018 11:13:12 +0100 Subject: [PATCH] IIO: ADC: add stm32 DFSDM support for PDM microphone This code offers a way to handle PDM audio microphones in ASOC framework. Audio driver should use consumer API. A specific management is implemented for DMA, with a callback, to allows to handle audio buffers efficiently. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 | 16 + drivers/iio/adc/stm32-dfsdm-adc.c | 502 ++++++++++++++++++++- include/linux/iio/adc/stm32-dfsdm-adc.h | 18 + 3 files changed, 529 insertions(+), 7 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 create mode 100644 include/linux/iio/adc/stm32-dfsdm-adc.h diff --git a/Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 b/Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 new file mode 100644 index 000000000000..da9822309f07 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 @@ -0,0 +1,16 @@ +What: /sys/bus/iio/devices/iio:deviceX/in_voltage_spi_clk_freq +KernelVersion: 4.14 +Contact: arnaud.pouliquen(a)st.com +Description: + For audio purpose only. + Used by audio driver to set/get the spi input frequency. + This is mandatory if DFSDM is slave on SPI bus, to + provide information on the SPI clock frequency during runtime + Notice that the SPI frequency should be a multiple of sample + frequency to ensure the precision. + if DFSDM input is SPI master + Reading SPI clkout frequency, + error on writing + If DFSDM input is SPI Slave: + Reading returns value previously set. + Writing value before starting conversions. \ No newline at end of file diff --git a/drivers/iio/adc/stm32-dfsdm-adc.c b/drivers/iio/adc/stm32-dfsdm-adc.c index 68b5920e92cb..b03ca3f94331 100644 --- a/drivers/iio/adc/stm32-dfsdm-adc.c +++ b/drivers/iio/adc/stm32-dfsdm-adc.c @@ -6,19 +6,23 @@ * Author: Arnaud Pouliquen <arnaud.pouliquen(a)st.com>. */ +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> #include <linux/interrupt.h> #include <linux/iio/buffer.h> #include <linux/iio/hw-consumer.h> #include <linux/iio/iio.h> #include <linux/iio/sysfs.h> #include <linux/module.h> -#include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/regmap.h> #include <linux/slab.h> #include "stm32-dfsdm.h" +#define DFSDM_DMA_BUFFER_SIZE (4 * PAGE_SIZE) + /* Conversion timeout */ #define DFSDM_TIMEOUT_US 100000 #define DFSDM_TIMEOUT (msecs_to_jiffies(DFSDM_TIMEOUT_US / 1000)) @@ -58,6 +62,18 @@ struct stm32_dfsdm_adc { struct completion completion; u32 *buffer; + /* Audio specific */ + unsigned int spi_freq; /* SPI bus clock frequency */ + unsigned int sample_freq; /* Sample frequency after filter decimation */ + int (*cb)(const void *data, size_t size, void *cb_priv); + void *cb_priv; + + /* DMA */ + u8 *rx_buf; + unsigned int bufi; /* Buffer current position */ + unsigned int buf_sz; /* Buffer size */ + struct dma_chan *dma_chan; + dma_addr_t dma_buf; }; struct stm32_dfsdm_str2field { @@ -351,10 +367,63 @@ int stm32_dfsdm_channel_parse_of(struct stm32_dfsdm *dfsdm, return 0; } +static ssize_t dfsdm_adc_audio_get_spiclk(struct iio_dev *indio_dev, + uintptr_t priv, + const struct iio_chan_spec *chan, + char *buf) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + + return snprintf(buf, PAGE_SIZE, "%d\n", adc->spi_freq); +} + +static ssize_t dfsdm_adc_audio_set_spiclk(struct iio_dev *indio_dev, + uintptr_t priv, + const struct iio_chan_spec *chan, + const char *buf, size_t len) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + struct stm32_dfsdm_filter *fl = &adc->dfsdm->fl_list[adc->fl_id]; + struct stm32_dfsdm_channel *ch = &adc->dfsdm->ch_list[adc->ch_id]; + unsigned int sample_freq = adc->sample_freq; + unsigned int spi_freq; + int ret; + + dev_err(&indio_dev->dev, "enter %s\n", __func__); + /* If DFSDM is master on SPI, SPI freq can not be updated */ + if (ch->src != DFSDM_CHANNEL_SPI_CLOCK_EXTERNAL) + return -EPERM; + + ret = kstrtoint(buf, 0, &spi_freq); + if (ret) + return ret; + + if (!spi_freq) + return -EINVAL; + + if (sample_freq) { + if (spi_freq % sample_freq) + dev_warn(&indio_dev->dev, + "Sampling rate not accurate (%d)\n", + spi_freq / (spi_freq / sample_freq)); + + ret = stm32_dfsdm_set_osrs(fl, 0, (spi_freq / sample_freq)); + if (ret < 0) { + dev_err(&indio_dev->dev, + "No filter parameters that match!\n"); + return ret; + } + } + adc->spi_freq = spi_freq; + + return len; +} + static int stm32_dfsdm_start_conv(struct stm32_dfsdm_adc *adc, bool dma) { struct regmap *regmap = adc->dfsdm->regmap; int ret; + unsigned int dma_en = 0, cont_en = 0; ret = stm32_dfsdm_start_channel(adc->dfsdm, adc->ch_id); if (ret < 0) @@ -365,6 +434,24 @@ static int stm32_dfsdm_start_conv(struct stm32_dfsdm_adc *adc, bool dma) if (ret < 0) goto stop_channels; + if (dma) { + /* Enable DMA transfer*/ + dma_en = DFSDM_CR1_RDMAEN(1); + /* Enable conversion triggered by SPI clock*/ + cont_en = DFSDM_CR1_RCONT(1); + } + /* Enable DMA transfer*/ + ret = regmap_update_bits(regmap, DFSDM_CR1(adc->fl_id), + DFSDM_CR1_RDMAEN_MASK, dma_en); + if (ret < 0) + goto stop_channels; + + /* Enable conversion triggered by SPI clock*/ + ret = regmap_update_bits(regmap, DFSDM_CR1(adc->fl_id), + DFSDM_CR1_RCONT_MASK, cont_en); + if (ret < 0) + goto stop_channels; + ret = stm32_dfsdm_start_filter(adc->dfsdm, adc->fl_id); if (ret < 0) goto stop_channels; @@ -398,6 +485,231 @@ static void stm32_dfsdm_stop_conv(struct stm32_dfsdm_adc *adc) stm32_dfsdm_stop_channel(adc->dfsdm, adc->ch_id); } +static int stm32_dfsdm_set_watermark(struct iio_dev *indio_dev, + unsigned int val) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + unsigned int watermark = DFSDM_DMA_BUFFER_SIZE / 2; + + /* + * DMA cyclic transfers are used, buffer is split into two periods. + * There should be : + * - always one buffer (period) DMA is working on + * - one buffer (period) driver pushed to ASoC side. + */ + watermark = min(watermark, val * (unsigned int)(sizeof(u32))); + adc->buf_sz = watermark * 2; + + return 0; +} + +static unsigned int stm32_dfsdm_adc_dma_residue(struct stm32_dfsdm_adc *adc) +{ + struct dma_tx_state state; + enum dma_status status; + + status = dmaengine_tx_status(adc->dma_chan, + adc->dma_chan->cookie, + &state); + if (status == DMA_IN_PROGRESS) { + /* Residue is size in bytes from end of buffer */ + unsigned int i = adc->buf_sz - state.residue; + unsigned int size; + + /* Return available bytes */ + if (i >= adc->bufi) + size = i - adc->bufi; + else + size = adc->buf_sz + i - adc->bufi; + + return size; + } + + return 0; +} + +static void stm32_dfsdm_audio_dma_buffer_done(void *data) +{ + struct iio_dev *indio_dev = data; + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + int available = stm32_dfsdm_adc_dma_residue(adc); + size_t old_pos; + + /* + * FIXME: In Kernel interface does not support cyclic DMA buffer,and + * offers only an interface to push data samples per samples. + * For this reason IIO buffer interface is not used and interface is + * bypassed using a private callback registered by ASoC. + * This should be a temporary solution waiting a cyclic DMA engine + * support in IIO. + */ + + dev_dbg(&indio_dev->dev, "%s: pos = %d, available = %d\n", __func__, + adc->bufi, available); + old_pos = adc->bufi; + + while (available >= indio_dev->scan_bytes) { + u32 *buffer = (u32 *)&adc->rx_buf[adc->bufi]; + + /* Mask 8 LSB that contains the channel ID */ + *buffer = (*buffer & 0xFFFFFF00) << 8; + available -= indio_dev->scan_bytes; + adc->bufi += indio_dev->scan_bytes; + if (adc->bufi >= adc->buf_sz) { + if (adc->cb) + adc->cb(&adc->rx_buf[old_pos], + adc->buf_sz - old_pos, adc->cb_priv); + adc->bufi = 0; + old_pos = 0; + } + } + if (adc->cb) + adc->cb(&adc->rx_buf[old_pos], adc->bufi - old_pos, + adc->cb_priv); +} + +static int stm32_dfsdm_adc_dma_start(struct iio_dev *indio_dev) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + int ret; + + if (!adc->dma_chan) + return -EINVAL; + + dev_dbg(&indio_dev->dev, "%s size=%d watermark=%d\n", __func__, + adc->buf_sz, adc->buf_sz / 2); + + /* Prepare a DMA cyclic transaction */ + desc = dmaengine_prep_dma_cyclic(adc->dma_chan, + adc->dma_buf, + adc->buf_sz, adc->buf_sz / 2, + DMA_DEV_TO_MEM, + DMA_PREP_INTERRUPT); + if (!desc) + return -EBUSY; + + desc->callback = stm32_dfsdm_audio_dma_buffer_done; + desc->callback_param = indio_dev; + + cookie = dmaengine_submit(desc); + ret = dma_submit_error(cookie); + if (ret) { + dmaengine_terminate_all(adc->dma_chan); + return ret; + } + + /* Issue pending DMA requests */ + dma_async_issue_pending(adc->dma_chan); + + return 0; +} + +static int stm32_dfsdm_postenable(struct iio_dev *indio_dev) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + int ret; + + /* Reset adc buffer index */ + adc->bufi = 0; + + ret = stm32_dfsdm_start_dfsdm(adc->dfsdm); + if (ret < 0) + return ret; + + ret = stm32_dfsdm_start_conv(adc, true); + if (ret) { + dev_err(&indio_dev->dev, "Can't start conversion\n"); + goto stop_dfsdm; + } + + if (adc->dma_chan) { + ret = stm32_dfsdm_adc_dma_start(indio_dev); + if (ret) { + dev_err(&indio_dev->dev, "Can't start DMA\n"); + goto err_stop_conv; + } + } + + return 0; + +err_stop_conv: + stm32_dfsdm_stop_conv(adc); +stop_dfsdm: + stm32_dfsdm_stop_dfsdm(adc->dfsdm); + + return ret; +} + +static int stm32_dfsdm_predisable(struct iio_dev *indio_dev) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + + if (adc->dma_chan) + dmaengine_terminate_all(adc->dma_chan); + + stm32_dfsdm_stop_conv(adc); + + stm32_dfsdm_stop_dfsdm(adc->dfsdm); + + return 0; +} + +static const struct iio_buffer_setup_ops stm32_dfsdm_buffer_setup_ops = { + .postenable = &stm32_dfsdm_postenable, + .predisable = &stm32_dfsdm_predisable, +}; + +/** + * stm32_dfsdm_get_buff_cb() - register a callback that will be called when + * DMA transfer period is achieved. + * + * @iio_dev: Handle to IIO device. + * @cb: Pointer to callback function: + * - data: pointer to data buffer + * - size: size in byte of the data buffer + * - private: pointer to consumer private structure. + * @private: Pointer to consumer private structure. + */ +int stm32_dfsdm_get_buff_cb(struct iio_dev *iio_dev, + int (*cb)(const void *data, size_t size, + void *private), + void *private) +{ + struct stm32_dfsdm_adc *adc; + + if (!iio_dev) + return -EINVAL; + adc = iio_priv(iio_dev); + + adc->cb = cb; + adc->cb_priv = private; + + return 0; +} +EXPORT_SYMBOL_GPL(stm32_dfsdm_get_buff_cb); + +/** + * stm32_dfsdm_release_buff_cb - unregister buffer callback + * + * @iio_dev: Handle to IIO device. + */ +int stm32_dfsdm_release_buff_cb(struct iio_dev *iio_dev) +{ + struct stm32_dfsdm_adc *adc; + + if (!iio_dev) + return -EINVAL; + adc = iio_priv(iio_dev); + + adc->cb = NULL; + adc->cb_priv = NULL; + + return 0; +} +EXPORT_SYMBOL_GPL(stm32_dfsdm_release_buff_cb); + static int stm32_dfsdm_single_conv(struct iio_dev *indio_dev, const struct iio_chan_spec *chan, int *res) { @@ -453,15 +765,41 @@ static int stm32_dfsdm_write_raw(struct iio_dev *indio_dev, { struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); struct stm32_dfsdm_filter *fl = &adc->dfsdm->fl_list[adc->fl_id]; + struct stm32_dfsdm_channel *ch = &adc->dfsdm->ch_list[adc->ch_id]; + unsigned int spi_freq = adc->spi_freq; int ret = -EINVAL; - if (mask == IIO_CHAN_INFO_OVERSAMPLING_RATIO) { + switch (mask) { + case IIO_CHAN_INFO_OVERSAMPLING_RATIO: ret = stm32_dfsdm_set_osrs(fl, 0, val); if (!ret) adc->oversamp = val; + + return ret; + + case IIO_CHAN_INFO_SAMP_FREQ: + if (!val) + return -EINVAL; + if (ch->src != DFSDM_CHANNEL_SPI_CLOCK_EXTERNAL) + spi_freq = adc->dfsdm->spi_master_freq; + + if (spi_freq % val) + dev_warn(&indio_dev->dev, + "Sampling rate not accurate (%d)\n", + spi_freq / (spi_freq / val)); + + ret = stm32_dfsdm_set_osrs(fl, 0, (spi_freq / val)); + if (ret < 0) { + dev_err(&indio_dev->dev, + "Not able to find parameter that match!\n"); + return ret; + } + adc->sample_freq = val; + + return 0; } - return ret; + return -EINVAL; } static int stm32_dfsdm_read_raw(struct iio_dev *indio_dev, @@ -493,12 +831,23 @@ static int stm32_dfsdm_read_raw(struct iio_dev *indio_dev, case IIO_CHAN_INFO_OVERSAMPLING_RATIO: *val = adc->oversamp; + return IIO_VAL_INT; + + case IIO_CHAN_INFO_SAMP_FREQ: + *val = adc->sample_freq; + return IIO_VAL_INT; } return -EINVAL; } +static const struct iio_info stm32_dfsdm_info_audio = { + .hwfifo_set_watermark = stm32_dfsdm_set_watermark, + .read_raw = stm32_dfsdm_read_raw, + .write_raw = stm32_dfsdm_write_raw, +}; + static const struct iio_info stm32_dfsdm_info_adc = { .read_raw = stm32_dfsdm_read_raw, .write_raw = stm32_dfsdm_write_raw, @@ -531,6 +880,70 @@ static irqreturn_t stm32_dfsdm_irq(int irq, void *arg) return IRQ_HANDLED; } +/* + * Define external info for SPI Frequency and audio sampling rate that can be + * configured by ASoC driver through consumer.h API + */ +static const struct iio_chan_spec_ext_info dfsdm_adc_audio_ext_info[] = { + /* spi_clk_freq : clock freq on SPI/manchester bus used by channel */ + { + .name = "spi_clk_freq", + .shared = IIO_SHARED_BY_TYPE, + .read = dfsdm_adc_audio_get_spiclk, + .write = dfsdm_adc_audio_set_spiclk, + }, + {}, +}; + +static void stm32_dfsdm_dma_release(struct iio_dev *indio_dev) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + + if (adc->dma_chan) { + dma_free_coherent(adc->dma_chan->device->dev, + DFSDM_DMA_BUFFER_SIZE, + adc->rx_buf, adc->dma_buf); + dma_release_channel(adc->dma_chan); + } +} + +static int stm32_dfsdm_dma_request(struct iio_dev *indio_dev) +{ + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + struct dma_slave_config config = { + .src_addr = (dma_addr_t)adc->dfsdm->phys_base + + DFSDM_RDATAR(adc->fl_id), + .src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + }; + int ret; + + adc->dma_chan = dma_request_slave_channel(&indio_dev->dev, "rx"); + if (!adc->dma_chan) + return -EINVAL; + + adc->rx_buf = dma_alloc_coherent(adc->dma_chan->device->dev, + DFSDM_DMA_BUFFER_SIZE, + &adc->dma_buf, GFP_KERNEL); + if (!adc->rx_buf) { + ret = -ENOMEM; + goto err_release; + } + + ret = dmaengine_slave_config(adc->dma_chan, &config); + if (ret) + goto err_free; + + return 0; + +err_free: + dma_free_coherent(adc->dma_chan->device->dev, DFSDM_DMA_BUFFER_SIZE, + adc->rx_buf, adc->dma_buf); +err_release: + dma_release_channel(adc->dma_chan); + + return ret; +} + static int stm32_dfsdm_adc_chan_init_one(struct iio_dev *indio_dev, struct iio_chan_spec *ch) { @@ -551,7 +964,12 @@ static int stm32_dfsdm_adc_chan_init_one(struct iio_dev *indio_dev, ch->info_mask_separate = BIT(IIO_CHAN_INFO_RAW); ch->info_mask_shared_by_all = BIT(IIO_CHAN_INFO_OVERSAMPLING_RATIO); - ch->scan_type.sign = 'u'; + if (adc->dev_data->type == DFSDM_AUDIO) { + ch->scan_type.sign = 's'; + ch->ext_info = dfsdm_adc_audio_ext_info; + } else { + ch->scan_type.sign = 'u'; + } ch->scan_type.realbits = 24; ch->scan_type.storagebits = 32; adc->ch_id = ch->channel; @@ -560,6 +978,39 @@ static int stm32_dfsdm_adc_chan_init_one(struct iio_dev *indio_dev, &adc->dfsdm->ch_list[ch->channel]); } +static int stm32_dfsdm_audio_init(struct iio_dev *indio_dev) +{ + struct iio_chan_spec *ch; + struct stm32_dfsdm_adc *adc = iio_priv(indio_dev); + struct stm32_dfsdm_channel *d_ch; + int ret; + + indio_dev->modes |= INDIO_BUFFER_SOFTWARE; + indio_dev->setup_ops = &stm32_dfsdm_buffer_setup_ops; + + ch = devm_kzalloc(&indio_dev->dev, sizeof(*ch), GFP_KERNEL); + if (!ch) + return -ENOMEM; + + ch->scan_index = 0; + + ret = stm32_dfsdm_adc_chan_init_one(indio_dev, ch); + if (ret < 0) { + dev_err(&indio_dev->dev, "Channels init failed\n"); + return ret; + } + ch->info_mask_separate = BIT(IIO_CHAN_INFO_SAMP_FREQ); + + d_ch = &adc->dfsdm->ch_list[adc->ch_id]; + if (d_ch->src != DFSDM_CHANNEL_SPI_CLOCK_EXTERNAL) + adc->spi_freq = adc->dfsdm->spi_master_freq; + + indio_dev->num_channels = 1; + indio_dev->channels = ch; + + return stm32_dfsdm_dma_request(indio_dev); +} + static int stm32_dfsdm_adc_init(struct iio_dev *indio_dev) { struct iio_chan_spec *ch; @@ -612,11 +1063,20 @@ static const struct stm32_dfsdm_dev_data stm32h7_dfsdm_adc_data = { .init = stm32_dfsdm_adc_init, }; +static const struct stm32_dfsdm_dev_data stm32h7_dfsdm_audio_data = { + .type = DFSDM_AUDIO, + .init = stm32_dfsdm_audio_init, +}; + static const struct of_device_id stm32_dfsdm_adc_match[] = { { .compatible = "st,stm32-dfsdm-adc", .data = &stm32h7_dfsdm_adc_data, }, + { + .compatible = "st,stm32-dfsdm-dmic", + .data = &stm32h7_dfsdm_audio_data, + }, {} }; @@ -667,8 +1127,13 @@ static int stm32_dfsdm_adc_probe(struct platform_device *pdev) name = devm_kzalloc(dev, sizeof("dfsdm-adc0"), GFP_KERNEL); if (!name) return -ENOMEM; - iio->info = &stm32_dfsdm_info_adc; - snprintf(name, sizeof("dfsdm-adc0"), "dfsdm-adc%d", adc->fl_id); + if (dev_data->type == DFSDM_AUDIO) { + iio->info = &stm32_dfsdm_info_audio; + snprintf(name, sizeof("dfsdm-pdm0"), "dfsdm-pdm%d", adc->fl_id); + } else { + iio->info = &stm32_dfsdm_info_adc; + snprintf(name, sizeof("dfsdm-adc0"), "dfsdm-adc%d", adc->fl_id); + } iio->name = name; /* @@ -700,7 +1165,27 @@ static int stm32_dfsdm_adc_probe(struct platform_device *pdev) if (ret < 0) return ret; - return iio_device_register(iio); + ret = iio_device_register(iio); + if (ret < 0) + goto err_cleanup; + + dev_err(dev, "of_platform_populate\n"); + if (dev_data->type == DFSDM_AUDIO) { + ret = of_platform_populate(np, NULL, NULL, dev); + if (ret < 0) { + dev_err(dev, "Failed to find an audio DAI\n"); + goto err_unregister; + } + } + + return 0; + +err_unregister: + iio_device_unregister(iio); +err_cleanup: + stm32_dfsdm_dma_release(iio); + + return ret; } static int stm32_dfsdm_adc_remove(struct platform_device *pdev) @@ -708,7 +1193,10 @@ static int stm32_dfsdm_adc_remove(struct platform_device *pdev) struct stm32_dfsdm_adc *adc = platform_get_drvdata(pdev); struct iio_dev *indio_dev = iio_priv_to_dev(adc); + if (adc->dev_data->type == DFSDM_AUDIO) + of_platform_depopulate(&pdev->dev); iio_device_unregister(indio_dev); + stm32_dfsdm_dma_release(indio_dev); return 0; } diff --git a/include/linux/iio/adc/stm32-dfsdm-adc.h b/include/linux/iio/adc/stm32-dfsdm-adc.h new file mode 100644 index 000000000000..e7dc7a542a4e --- /dev/null +++ b/include/linux/iio/adc/stm32-dfsdm-adc.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * This file discribe the STM32 DFSDM IIO driver API for audio part + * + * Copyright (C) 2017, STMicroelectronics - All Rights Reserved + * Author(s): Arnaud Pouliquen <arnaud.pouliquen(a)st.com>. + */ + +#ifndef STM32_DFSDM_ADC_H +#define STM32_DFSDM_ADC_H + +int stm32_dfsdm_get_buff_cb(struct iio_dev *iio_dev, + int (*cb)(const void *data, size_t size, + void *private), + void *private); +int stm32_dfsdm_release_buff_cb(struct iio_dev *iio_dev); + +#endif -- 2.15.1

1 0

[alsa-devel] Applied "IIO: consumer: allow to set buffer sizes" to the asoc tree
by Mark Brown 10 Jan '18

10 Jan '18

The patch IIO: consumer: allow to set buffer sizes has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 16cbca06fa8288a4f58426fc898b141e12ee8008 Mon Sep 17 00:00:00 2001 From: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Date: Wed, 10 Jan 2018 11:13:13 +0100 Subject: [PATCH] IIO: consumer: allow to set buffer sizes Add iio consumer API to set buffer size and watermark according to sysfs API. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/iio/buffer/industrialio-buffer-cb.c | 11 +++++++++++ include/linux/iio/consumer.h | 11 +++++++++++ 2 files changed, 22 insertions(+) diff --git a/drivers/iio/buffer/industrialio-buffer-cb.c b/drivers/iio/buffer/industrialio-buffer-cb.c index 4847534700e7..ea63c838eeae 100644 --- a/drivers/iio/buffer/industrialio-buffer-cb.c +++ b/drivers/iio/buffer/industrialio-buffer-cb.c @@ -104,6 +104,17 @@ struct iio_cb_buffer *iio_channel_get_all_cb(struct device *dev, } EXPORT_SYMBOL_GPL(iio_channel_get_all_cb); +int iio_channel_cb_set_buffer_watermark(struct iio_cb_buffer *cb_buff, + size_t watermark) +{ + if (!watermark) + return -EINVAL; + cb_buff->buffer.watermark = watermark; + + return 0; +} +EXPORT_SYMBOL_GPL(iio_channel_cb_set_buffer_watermark); + int iio_channel_start_all_cb(struct iio_cb_buffer *cb_buff) { return iio_update_buffers(cb_buff->indio_dev, &cb_buff->buffer, diff --git a/include/linux/iio/consumer.h b/include/linux/iio/consumer.h index 2017f35db17c..9887f4f8e2a8 100644 --- a/include/linux/iio/consumer.h +++ b/include/linux/iio/consumer.h @@ -133,6 +133,17 @@ struct iio_cb_buffer *iio_channel_get_all_cb(struct device *dev, int (*cb)(const void *data, void *private), void *private); +/** + * iio_channel_cb_set_buffer_watermark() - set the buffer watermark. + * @cb_buffer: The callback buffer from whom we want the channel + * information. + * @watermark: buffer watermark in bytes. + * + * This function allows to configure the buffer watermark. + */ +int iio_channel_cb_set_buffer_watermark(struct iio_cb_buffer *cb_buffer, + size_t watermark); + /** * iio_channel_release_all_cb() - release and unregister the callback. * @cb_buffer: The callback buffer that was allocated. -- 2.15.1

1 0

[alsa-devel] [PATCH v8 00/13] Add STM32 DFSDM support
by Arnaud Pouliquen 10 Jan '18

10 Jan '18

Hello New version based on minor comments from Jonathan. Main deltas V8 vs V7: - Few typos fixes. - Function return optimizations in sound/soc/stm/stm32_adfsdm.c. Main deltas V7 vs V6: - Replaces the custom license information text with the appropriate SPDX identifier. - Few fixes in sound/soc/stm/stm32_adfsdm.c and stm32-dfsdm-core.c. - Add missing #interrupt-cells in binding examples. - Integrate last Jonathan's comments. Main deltas V6 vs V5: - Fix warning reported by kbuild test in : include/linux/iio/consumer.h sound/soc/stm/stm32_adfsdm.c Main deltas V5 vs V4: - Integrate ASOC DAI as a subnode of the DFSDM. - Add in kernel consumer interface to allow to manipulate attribute. Thanks, Arnaud Arnaud Pouliquen (12): docs: driver-api: add iio hw consumer section IIO: hw_consumer: add devm_iio_hw_consumer_alloc IIO: inkern: API for manipulating channel attributes IIO: Add DT bindings for sigma delta adc modulator IIO: ADC: add sigma delta modulator support IIO: add DT bindings for stm32 DFSDM filter IIO: ADC: add stm32 DFSDM core support IIO: ADC: add STM32 DFSDM sigma delta ADC support IIO: ADC: add stm32 DFSDM support for PDM microphone IIO: consumer: allow to set buffer sizes ASoC: add bindings for stm32 DFSDM filter ASoC: stm32: add DFSDM DAI support Lars-Peter Clausen (1): iio: Add hardware consumer buffer support .../ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 | 16 + .../bindings/iio/adc/sigma-delta-modulator.txt | 13 + .../bindings/iio/adc/st,stm32-dfsdm-adc.txt | 128 ++ .../devicetree/bindings/sound/st,stm32-adfsdm.txt | 63 + Documentation/driver-api/iio/hw-consumer.rst | 51 + Documentation/driver-api/iio/index.rst | 1 + drivers/iio/adc/Kconfig | 37 + drivers/iio/adc/Makefile | 3 + drivers/iio/adc/sd_adc_modulator.c | 68 ++ drivers/iio/adc/stm32-dfsdm-adc.c | 1220 ++++++++++++++++++++ drivers/iio/adc/stm32-dfsdm-core.c | 309 +++++ drivers/iio/adc/stm32-dfsdm.h | 310 +++++ drivers/iio/buffer/Kconfig | 10 + drivers/iio/buffer/Makefile | 1 + drivers/iio/buffer/industrialio-buffer-cb.c | 11 + drivers/iio/buffer/industrialio-hw-consumer.c | 247 ++++ drivers/iio/inkern.c | 17 +- include/linux/iio/adc/stm32-dfsdm-adc.h | 18 + include/linux/iio/consumer.h | 37 + include/linux/iio/hw-consumer.h | 21 + include/linux/iio/iio.h | 28 - include/linux/iio/types.h | 28 + sound/soc/stm/Kconfig | 11 + sound/soc/stm/Makefile | 3 + sound/soc/stm/stm32_adfsdm.c | 347 ++++++ 25 files changed, 2965 insertions(+), 33 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-iio-dfsdm-adc-stm32 create mode 100644 Documentation/devicetree/bindings/iio/adc/sigma-delta-modulator.txt create mode 100644 Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-adfsdm.txt create mode 100644 Documentation/driver-api/iio/hw-consumer.rst create mode 100644 drivers/iio/adc/sd_adc_modulator.c create mode 100644 drivers/iio/adc/stm32-dfsdm-adc.c create mode 100644 drivers/iio/adc/stm32-dfsdm-core.c create mode 100644 drivers/iio/adc/stm32-dfsdm.h create mode 100644 drivers/iio/buffer/industrialio-hw-consumer.c create mode 100644 include/linux/iio/adc/stm32-dfsdm-adc.h create mode 100644 include/linux/iio/hw-consumer.h create mode 100644 sound/soc/stm/stm32_adfsdm.c -- 2.7.4

3 16

[alsa-devel] [PATCH] ASoC: mt8173: remove unnecessary micbias widget in route
by Bard Liao 10 Jan '18

10 Jan '18

The micbias1/2 are connected to route as SUPPLY usage. It was not take effect since they were MICBIAS type. To keep the same register settings, we have to remove it once the micbias1/2 widget is converted to SUPPLY type. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 -- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 -- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 -- 3 files changed, 6 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 99c1521..5a9a548 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -37,8 +37,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_rt5514_routes[] = { {"Sub DMIC1R", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, }; diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index 42de84c..b724808 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -40,8 +40,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_rt5676_routes[] = { {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, {"Headphone", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650 */ - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, {"Sub AIF2RX", NULL, "Headset Mic"}, /* IF2 DAC from 5650 */ diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index e69c141..40ebefd 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -51,8 +51,6 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, {"Headphone", NULL, "HPOR"}, - {"Headset Mic", NULL, "micbias1"}, - {"Headset Mic", NULL, "micbias2"}, {"IN1P", NULL, "Headset Mic"}, {"IN1N", NULL, "Headset Mic"}, }; -- 2.7.4

2 1

[alsa-devel] ASoC: sgtl5000: recording issue at 8 kHz with fsl-imx28
by Fischer, Christian 10 Jan '18

10 Jan '18

Hello guys, trying to record from a SGTL5000 with 8 kHz sampling rate gives a very bad audio quality. All other sampling rates working fine. Initially posted at NXP forum: https://community.nxp.com/message/975131 Tested with imx28, sgtl5000 and Kernel 4.14.12. I've attached a recording with 8 kHz, capture-vol was 67, mic-gain 33, AVC-off and Capture-ZC off. Recorded with "arecord -D "hw:0,1" -t wav -f S16_LE -c 2 -r 8000 test.wav". Sounds like there are missing samples.. The microphone is powered from mic-bias and decoupled with a 100nF capacitor. Can anybody help me? Thank you guys! Kind regards, Christian Fischer

1 0

[alsa-devel] [ANNOUNCE] SOF 1.0 Release
by Liam Girdwood 10 Jan '18

10 Jan '18

I'd like to announce the 1.0 release of Sound Open Firmware (SOF) is available for download on git.alsa-project.org. SOF is an open source BSD licensed audio DSP firmware that currently runs on Cadence xtensa architecture DSPs found on Intel Baytrail, Braswell and Cherrytrail based devices, however SOF is architecture and platform independent meaning it's not tightly coupled to xtensa DSPs or Intel CPUs/SoCs. The 1.0 release supports limited playback and capture topologies alongside volume, mixing and sample format conversion. Information about SOF can be found on the ALSA SOF wiki here :- https://www.alsa-project.org/main/index.php/Firmware Development for SOF 1.1 has started which will include support for Haswell, Broadwell and Apollolake platform alongside support for SRC and more complex topologies. Liam

4 4

Re: [alsa-devel] Headphone issues iMac 14.1 (A1418)
by Takashi Iwai 10 Jan '18

10 Jan '18

On Wed, 10 Jan 2018 09:46:27 +0100, Freaky wrote: > > Sorry, forgot to ask, is the patch going to be included upstream and/or > would you like us to test it? The official patch is below and I queued it now. It'll be likely included in 4.15-final after rc8. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Apply the existing quirk to iMac 14,1 iMac 14,1 requires the same quirk as iMac 12,2, using GPIO 2 and 3 for headphone and speaker output amps. Add the codec SSID quirk entry (106b:0600) accordingly. BugLink: http://lkml.kernel.org/r/CAEw6Zyteav09VGHRfD5QwsfuWv5a43r0tFBNbfcHXoNrxVz7e… Reported-by: Freaky <freaky2000(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 80bbadc83721..d6e079f4ec09 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -408,6 +408,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /*SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27),*/ /* codec SSID */ + SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), -- 2.15.1

1 0

[alsa-devel] BUG: soft lockup (2)
by syzbot 10 Jan '18

10 Jan '18

syzkaller has found reproducer for the following crash on e1915c8195b38393005be9b74bfa6a3a367c83b3 git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. C reproducer is attached syzkaller reproducer is attached. See https://goo.gl/kgGztJ for information about syzkaller reproducers IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f76f3c62dfadce022fd1c1deff15a61e09ac708a(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. watchdog: BUG: soft lockup - CPU#0 stuck for 135s! [syzkaller670324:3527] Modules linked in: irq event stamp: 2531226 hardirqs last enabled at (2531225): [<00000000f1ec093f>] snd_pcm_stream_unlock_irq+0x78/0xe0 sound/core/pcm_native.c:166 hardirqs last disabled at (2531226): [<000000003c6ef1cd>] apic_timer_interrupt+0xa4/0xb0 arch/x86/entry/entry_64.S:920 softirqs last enabled at (41848): [<0000000081bd5f03>] __do_softirq+0x7a0/0xb85 kernel/softirq.c:311 softirqs last disabled at (41829): [<00000000d02c6d52>] invoke_softirq kernel/softirq.c:365 [inline] softirqs last disabled at (41829): [<00000000d02c6d52>] irq_exit+0x1cc/0x200 kernel/softirq.c:405 CPU: 0 PID: 3527 Comm: syzkaller670324 Not tainted 4.15.0-rc6+ #158 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:memcpy+0x45/0x50 mm/kasan/kasan.c:305 RSP: 0018:ffff8801bf6676f0 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff11 RAX: ffffc9000137ba06 RBX: 0000000000000002 RCX: 0000000000000000 RDX: 0000000000000002 RSI: ffff8801bf6677da RDI: ffffc9000137ba08 RBP: ffff8801bf667708 R08: fffff5200026f741 R09: fffff5200026f741 R10: 0000000000000001 R11: fffff5200026f740 R12: ffffc9000137ba06 R13: ffff8801bf6677d8 R14: dffffc0000000000 R15: ffffc9000137ba06 FS: 0000000000000000(0000) GS:ffff8801db200000(0063) knlGS:00000000f7ec6b40 CS: 0010 DS: 002b ES: 002b CR0: 0000000080050033 CR2: 0000000020735ee0 CR3: 00000001bfba8002 CR4: 00000000001606f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: memcpy include/linux/string.h:344 [inline] cvt_s16_to_native sound/core/oss/mulaw.c:164 [inline] mulaw_decode+0x52f/0x770 sound/core/oss/mulaw.c:195 mulaw_transfer+0x222/0x270 sound/core/oss/mulaw.c:273 snd_pcm_plug_write_transfer+0x22d/0x420 sound/core/oss/pcm_plugin.c:611 snd_pcm_oss_write2+0x260/0x420 sound/core/oss/pcm_oss.c:1311 snd_pcm_oss_write1 sound/core/oss/pcm_oss.c:1372 [inline] snd_pcm_oss_write+0x5fe/0x830 sound/core/oss/pcm_oss.c:2646 __vfs_write+0xef/0x970 fs/read_write.c:480 vfs_write+0x189/0x510 fs/read_write.c:544 SYSC_write fs/read_write.c:589 [inline] SyS_write+0xef/0x220 fs/read_write.c:581 do_syscall_32_irqs_on arch/x86/entry/common.c:327 [inline] do_fast_syscall_32+0x3ee/0xf9d arch/x86/entry/common.c:389 entry_SYSENTER_compat+0x54/0x63 arch/x86/entry/entry_64_compat.S:129 RIP: 0023:0xf7f0cc79 RSP: 002b:00000000f7ec61fc EFLAGS: 00000246 ORIG_RAX: 0000000000000004 RAX: ffffffffffffffda RBX: 0000000000000005 RCX: 0000000020735ee0 RDX: 00000000fffffee4 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 00000000003d0f00 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Code: 89 de 31 d2 e8 9d f9 ff ff 48 8b 4d 08 48 89 de 4c 89 e7 ba 01 00 00 00 e8 89 f9 ff ff 48 89 da 4c 89 ee 4c 89 e7 e8 7b eb c0 03 <5b> 41 5c 41 5d 5d c3 0f 1f 40 00 89 f1 b8 00 10 00 00 55 48 d3

2 1

[alsa-devel] snd_hda_intel: coil whine noises after resume
by Francisco G. 10 Jan '18

10 Jan '18

Hello. Some users included myself have noticed coil-whine-like noises coming through the headphones after the computer resumes from sleep. This noise is clearly associated to the laptop activity, e.g. when scrolling through a webpage or compiling something, the processor can be heard doing the work. Interestingly, the coil whine that can be heard from the computer with naked ears (which is a common problem with this laptop) is also increased after resume. The problem has been observed on a Dell XPS 13 9360 with an i7-8550U CPU by myself, as well as with an i5-8250U by a user in the Arch Linux forums <https://bbs.archlinux.org/viewtopic.php?pid=1759588#p1759588>. This user has also noticed that reloading the snd_hda_intel module fixes the issue. Please let me know of any inormation that would be needed to debug this problem.

2 5

[alsa-devel] [PATCH v5][RESEND] ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
by Kuninori Morimoto 10 Jan '18

10 Jan '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Power Down Release Command (PMVR, PMDAC, RSTN, PMDA1-PMDA6) which are located on PW_MGMT1 / PW_MGMT3 register must be write again after at least 5 LRCK cycle or later on each command. Otherwise, Playback volume will be 0dB. Basically, it should be 1. PowerDownRelease by Power Management1 <= call 1.x after 5LRCK 1.x Dummy write to Power Management1 2. PowerDownRelease by Power Management3 <= call 2.x after 5LRCK 2.x Dummy write to Power Management3 To avoid too many dummy write, this patch is merging these. 1. PowerDownRelease by Power Management1 2. PowerDownRelease by Power Management3 <= call after 5LRCK 2.x Dummy write to Power Management1/3 <= merge dummy write This patch adds dummy write when Playback Start timing. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- 2week passed. resend sound/soc/codecs/ak4613.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index ee9e822..d85c61e 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -15,6 +15,7 @@ */ #include <linux/clk.h> +#include <linux/delay.h> #include <linux/i2c.h> #include <linux/slab.h> #include <linux/of_device.h> @@ -95,6 +96,9 @@ struct ak4613_priv { struct mutex lock; const struct ak4613_interface *iface; struct snd_pcm_hw_constraint_list constraint; + struct work_struct dummy_write_work; + struct snd_soc_component *component; + unsigned int rate; unsigned int sysclk; unsigned int fmt; @@ -392,6 +396,7 @@ static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } + priv->rate = rate; /* * FIXME @@ -467,11 +472,83 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, return 0; } +static void ak4613_dummy_write(struct work_struct *work) +{ + struct ak4613_priv *priv = container_of(work, + struct ak4613_priv, + dummy_write_work); + struct snd_soc_component *component = priv->component; + unsigned int mgmt1; + unsigned int mgmt3; + + /* + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * + * Note + * + * To avoid extra delay, we want to avoid preemption here, + * but we can't. Because it uses I2C access which is using IRQ + * and sleep. Thus, delay might be more than 5 LR clocks + * see also + * ak4613_dai_trigger() + */ + udelay(5000000 / priv->rate); + + snd_soc_component_read(component, PW_MGMT1, &mgmt1); + snd_soc_component_read(component, PW_MGMT3, &mgmt3); + + snd_soc_component_write(component, PW_MGMT1, mgmt1); + snd_soc_component_write(component, PW_MGMT3, mgmt3); +} + +static int ak4613_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * FIXME + * + * PW_MGMT1 / PW_MGMT3 needs dummy write at least after 5 LR clocks + * from Power Down Release. Otherwise, Playback volume will be 0dB. + * To avoid complex multiple delay/dummy_write method from + * ak4613_set_bias_level() / SND_SOC_DAPM_DAC_E("DACx", ...), + * call it once here. + * + * But, unfortunately, we can't "write" here because here is atomic + * context (It uses I2C access for writing). + * Thus, use schedule_work() to switching to normal context + * immediately. + * + * Note + * + * Calling ak4613_dummy_write() function might be delayed. + * In such case, ak4613 volume might be temporarily 0dB when + * beggining of playback. + * see also + * ak4613_dummy_write() + */ + + if ((cmd != SNDRV_PCM_TRIGGER_START) && + (cmd != SNDRV_PCM_TRIGGER_RESUME)) + return 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + + priv->component = &codec->component; + schedule_work(&priv->dummy_write_work); + + return 0; +} + static const struct snd_soc_dai_ops ak4613_dai_ops = { .startup = ak4613_dai_startup, .shutdown = ak4613_dai_shutdown, .set_sysclk = ak4613_dai_set_sysclk, .set_fmt = ak4613_dai_set_fmt, + .trigger = ak4613_dai_trigger, .hw_params = ak4613_dai_hw_params, }; @@ -591,6 +668,7 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, priv->iface = NULL; priv->cnt = 0; priv->sysclk = 0; + INIT_WORK(&priv->dummy_write_work, ak4613_dummy_write); mutex_init(&priv->lock); -- 1.9.1

1 0