December 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] DT soundcard driver with special clock routing
by Petr Kulhavy 23 Dec '15

23 Dec '15

Hi, I'm developing a DT-based driver for an ARM SoC device with the following special clock routing: - WM8758 codec attached to the CPU DAI via I2S - the codec is I2S clock master - the CPU feeds the codec with 12MHz clock (outside of the I2S) - the codec converts the 12MHz internally to 48kHz*256 or 44.1kHz *256 clock and provides it to I2S as MCLK So the special set-up needed here is that the codec needs to be configured to: a) generate the proper MCLK b) enable the respective GPIO pin as clock output What is the best way of representing this set-up using device tree? I'm wondering if this can be done by adding a specific option in the codec DT binding for enabling the clock output and then use the simple audio card. Or do I need to write a specific soundcard driver that sets up the codec registers? I would like to avoid that if possible and use existing code... What would you recommend? Thanks -- Petr Kulhavy, MSc System Architect *BARIX* petr(a)barix.com <mailto:petr@barix.com> | Skype: brain.barix Barix AG, Seefeldstrasse 303 | 8008 Zurich, Switzerland T +41 43 43322 11 | www.barix.com <http://www.barix.com> You have received this email because of your relationship Barix AG and its affiliated companies. Barix AG and its affiliated companies do not sell or exchange email addresses, or any other personal contact information provided by you with any third parties. All email distributions are managed and controlled by Barix AG and its affiliated companies. Barix AG, Seefeldstr. 303, 8008 Zürich, Switzerland. Company Reg. No: CH-020.3.023.869-8, VAT Reg. No: CHE-105.687.663.

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[alsa-devel] [PATCH] ASoC: rt5651: add ACPI and OF support
by Bard Liao 23 Dec '15

23 Dec '15

Add required tables and the binding document for ACPI and OF matching. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5651.txt | 41 ++++++++++++++++++++++ sound/soc/codecs/rt5651.c | 31 ++++++++++++++++ 2 files changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5651.txt diff --git a/Documentation/devicetree/bindings/sound/rt5651.txt b/Documentation/devicetree/bindings/sound/rt5651.txt new file mode 100644 index 0000000..3875233 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5651.txt @@ -0,0 +1,41 @@ +RT5651 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5651". + +- reg : The I2C address of the device. + +Optional properties: + +- realtek,in2-differential + Boolean. Indicate MIC2 input are differential, rather than single-ended. + +- realtek,dmic-en + Boolean. true if dmic is used. + +Pins on the device (for linking into audio routes) for RT5651: + + * DMIC L1 + * DMIC R1 + * IN1P + * IN2P + * IN2N + * IN3P + * HPOL + * HPOR + * LOUTL + * LOUTR + * PDML + * PDMR + +Example: + +codec: rt5651@1a { + compatible = "realtek,rt5651"; + reg = <0x1a>; + realtek,dmic-en = "true"; + realtek,in2-diff = "false"; +}; diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 1d40318..7a61970 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -18,6 +18,7 @@ #include <linux/regmap.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1735,12 +1736,38 @@ static const struct regmap_config rt5651_regmap = { .num_ranges = ARRAY_SIZE(rt5651_ranges), }; +#if defined(CONFIG_OF) +static const struct of_device_id rt5651_of_match[] = { + { .compatible = "realtek,rt5651", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5651_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5651_acpi_match[] = { + { "10EC5651", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5651_acpi_match); +#endif + static const struct i2c_device_id rt5651_i2c_id[] = { { "rt5651", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); +static int rt5651_parse_dt(struct rt5651_priv *rt5651, struct device_node *np) +{ + rt5651->pdata.in2_diff = of_property_read_bool(np, + "realtek,in2-differential"); + rt5651->pdata.dmic_en = of_property_read_bool(np, + "realtek,dmic-en"); + + return 0; +} + static int rt5651_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1757,6 +1784,8 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, if (pdata) rt5651->pdata = *pdata; + else if (i2c->dev.of_node) + rt5651_parse_dt(rt5651, i2c->dev.of_node); rt5651->regmap = devm_regmap_init_i2c(i2c, &rt5651_regmap); if (IS_ERR(rt5651->regmap)) { @@ -1806,6 +1835,8 @@ static int rt5651_i2c_remove(struct i2c_client *i2c) static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", + .acpi_match_table = ACPI_PTR(rt5651_acpi_match), + .of_match_table = of_match_ptr(rt5651_of_match), }, .probe = rt5651_i2c_probe, .remove = rt5651_i2c_remove, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/6] ASoC: da7219: AB Silicon related updates and small fixes.
by Adam Thomson 23 Dec '15

23 Dec '15

This is a small collection of updates & fixes, put together for ease. For changes relating to AB silicon, previous silicon revision is obsolete and was unavailable outside of Dialog, so does not require driver support in the future. Changes are based on v4.4-rc6 kernel. Adam Thomson (6): ASoC: da7219: Disable regulators on probe() failure ASoC: da7219: Fix Sidetone to work regardless of DAI capture ASoC: da7219: Update REFERENCES reg default, in-line with HW ASoC: da7219: Remove internal LDO features of codec ASoC: da7219: Add support for 1.6V micbias level ASoC: da7219: Remove support for 32KHz PLL mode Documentation/devicetree/bindings/sound/da7219.txt | 8 +- include/sound/da7219.h | 14 +--- sound/soc/codecs/da7219.c | 87 +++++++--------------- sound/soc/codecs/da7219.h | 9 --- 4 files changed, 32 insertions(+), 86 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH v4 1/2] ASoC: wm_adsp: Add a handler for the compressed IRQ
by Charles Keepax 23 Dec '15

23 Dec '15

Here support is added for responding to DSP IRQs that are used to indicate data being available on the DSP. The idea is that we check the amount of data available upon receipt of an IRQ and on subsequent calls to the pointer callback we recheck once less than one fragment is available (to avoid excessive SPI traffic), if there is truely less than one fragment available we ack the last IRQ and wait for a new one. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v3: - Make spurious IRQs return IRQ_NONE - Add a comment to explain the usage of wm_adsp_buffer_ack_irq Thanks, Charles sound/soc/codecs/wm5110.c | 27 +++++++ sound/soc/codecs/wm_adsp.c | 192 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 3 + 3 files changed, 222 insertions(+) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c364096..dde94c4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2177,10 +2177,23 @@ static int wm5110_open(struct snd_compr_stream *stream) return wm_adsp_compr_open(&priv->core.adsp[n_adsp], stream); } +static irqreturn_t wm5110_adsp2_irq(int irq, void *data) +{ + struct wm5110_priv *florida = data; + int ret; + + ret = wm_adsp_compr_handle_irq(&florida->core.adsp[2]); + if (ret == -ENODEV) + return IRQ_NONE; + + return IRQ_HANDLED; +} + static int wm5110_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int i, ret; priv->core.arizona->dapm = dapm; @@ -2189,6 +2202,14 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, + "ADSP2 Compressed IRQ", wm5110_adsp2_irq, + priv); + if (ret != 0) { + dev_err(codec->dev, "Failed to request DSP IRQ: %d\n", ret); + return ret; + } + for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); if (ret) @@ -2209,12 +2230,15 @@ err_adsp2_codec_probe: for (--i; i >= 0; --i) wm_adsp2_codec_remove(&priv->core.adsp[i], codec); + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + return ret; } static int wm5110_codec_remove(struct snd_soc_codec *codec) { struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int i; for (i = 0; i < WM5110_NUM_ADSP; ++i) @@ -2222,6 +2246,8 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) priv->core.arizona->dapm = NULL; + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); + return 0; } @@ -2273,6 +2299,7 @@ static struct snd_compr_ops wm5110_compr_ops = { .set_params = wm_adsp_compr_set_params, .get_caps = wm_adsp_compr_get_caps, .trigger = wm_adsp_compr_trigger, + .pointer = wm_adsp_compr_pointer, }; static struct snd_soc_platform_driver wm5110_compr_platform = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ac879d1..edf96b5 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -279,6 +279,11 @@ struct wm_adsp_compr_buf { struct wm_adsp_buffer_region *regions; u32 host_buf_ptr; + + u32 error; + u32 irq_ack; + int read_index; + int avail; }; struct wm_adsp_compr { @@ -287,6 +292,8 @@ struct wm_adsp_compr { struct snd_compr_stream *stream; struct snd_compressed_buffer size; + + unsigned int copied_total; }; #define WM_ADSP_DATA_WORD_SIZE 3 @@ -2436,6 +2443,11 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, return -EINVAL; } +static inline unsigned int wm_adsp_compr_frag_words(struct wm_adsp_compr *compr) +{ + return compr->size.fragment_size / WM_ADSP_DATA_WORD_SIZE; +} + int wm_adsp_compr_set_params(struct snd_compr_stream *stream, struct snd_compr_params *params) { @@ -2622,6 +2634,8 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) return -ENOMEM; buf->dsp = dsp; + buf->read_index = -1; + buf->irq_ack = 0xFFFFFFFF; ret = wm_adsp_buffer_locate(buf); if (ret < 0) { @@ -2705,6 +2719,16 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) ret); break; } + + /* Trigger the IRQ at one fragment of data */ + ret = wm_adsp_buffer_write(compr->buf, + HOST_BUFFER_FIELD(high_water_mark), + wm_adsp_compr_frag_words(compr)); + if (ret < 0) { + adsp_err(dsp, "Failed to set high water mark: %d\n", + ret); + break; + } break; case SNDRV_PCM_TRIGGER_STOP: break; @@ -2719,4 +2743,172 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) } EXPORT_SYMBOL_GPL(wm_adsp_compr_trigger); +static inline int wm_adsp_buffer_size(struct wm_adsp_compr_buf *buf) +{ + int last_region = wm_adsp_fw[buf->dsp->fw].caps->num_regions - 1; + + return buf->regions[last_region].cumulative_size; +} + +static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) +{ + u32 next_read_index, next_write_index; + int write_index, read_index, avail; + int ret; + + /* Only sync read index if we haven't already read a valid index */ + if (buf->read_index < 0) { + ret = wm_adsp_buffer_read(buf, + HOST_BUFFER_FIELD(next_read_index), + &next_read_index); + if (ret < 0) + return ret; + + read_index = sign_extend32(next_read_index, 23); + + if (read_index < 0) { + adsp_dbg(buf->dsp, "Avail check on unstarted stream\n"); + return 0; + } + + buf->read_index = read_index; + } + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(next_write_index), + &next_write_index); + if (ret < 0) + return ret; + + write_index = sign_extend32(next_write_index, 23); + + avail = write_index - buf->read_index; + if (avail < 0) + avail += wm_adsp_buffer_size(buf); + + adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", + buf->read_index, write_index, avail); + + buf->avail = avail; + + return 0; +} + +int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) +{ + struct wm_adsp_compr_buf *buf = dsp->buffer; + int ret = 0; + + mutex_lock(&dsp->pwr_lock); + + if (!buf) { + adsp_err(dsp, "Spurious buffer IRQ\n"); + ret = -ENODEV; + goto out; + } + + adsp_dbg(dsp, "Handling buffer IRQ\n"); + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); + if (ret < 0) { + adsp_err(dsp, "Failed to check buffer error: %d\n", ret); + goto out; + } + if (buf->error != 0) { + adsp_err(dsp, "Buffer error occurred: %d\n", buf->error); + ret = -EIO; + goto out; + } + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), + &buf->irq_ack); + if (ret < 0) { + adsp_err(dsp, "Failed to get irq_count: %d\n", ret); + goto out; + } + + ret = wm_adsp_buffer_update_avail(buf); + if (ret < 0) { + adsp_err(dsp, "Error reading avail: %d\n", ret); + goto out; + } + +out: + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_handle_irq); + +/* + * We only need to ack the IRQ on the firmware end when we wish to receive + * another, as such we do this in response to running out of data in the + * buffer, rather than in direct response to the IRQ itself. + */ +static int wm_adsp_buffer_ack_irq(struct wm_adsp_compr_buf *buf) +{ + if (buf->irq_ack & 0x01) + return 0; + + adsp_dbg(buf->dsp, "Acking buffer IRQ(0x%x)\n", buf->irq_ack); + + buf->irq_ack |= 0x01; + + return wm_adsp_buffer_write(buf, HOST_BUFFER_FIELD(irq_ack), + buf->irq_ack); +} + +int wm_adsp_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp) +{ + struct wm_adsp_compr *compr = stream->runtime->private_data; + struct wm_adsp_compr_buf *buf = compr->buf; + struct wm_adsp *dsp = compr->dsp; + int ret = 0; + + adsp_dbg(dsp, "Pointer request\n"); + + mutex_lock(&dsp->pwr_lock); + + if (!compr->buf) { + ret = -ENXIO; + goto out; + } + + if (compr->buf->error) { + ret = -EIO; + goto out; + } + + if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_update_avail(buf); + if (ret < 0) { + adsp_err(dsp, "Error reading avail: %d\n", ret); + goto out; + } + + /* + * If we really have less than 1 fragment available ack the + * last DSP IRQ and rely on the IRQ to inform us once a whole + * fragment is available. + */ + if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_ack_irq(buf); + if (ret < 0) { + adsp_err(dsp, "Failed to ack buffer IRQ: %d\n", + ret); + goto out; + } + } + } + + tstamp->copied_total = compr->copied_total; + tstamp->copied_total += buf->avail * WM_ADSP_DATA_WORD_SIZE; + +out: + mutex_unlock(&dsp->pwr_lock); + + return ret; +} +EXPORT_SYMBOL_GPL(wm_adsp_compr_pointer); + MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 43af093..522fa1a 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -112,5 +112,8 @@ extern int wm_adsp_compr_set_params(struct snd_compr_stream *stream, extern int wm_adsp_compr_get_caps(struct snd_compr_stream *stream, struct snd_compr_caps *caps); extern int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd); +extern int wm_adsp_compr_handle_irq(struct wm_adsp *dsp); +extern int wm_adsp_compr_pointer(struct snd_compr_stream *stream, + struct snd_compr_tstamp *tstamp); #endif -- 2.1.4

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[alsa-devel] [PATCH 52/54] sound: soc: wm8903: Be sure to clamp return value
by Linus Walleij 23 Dec '15

23 Dec '15

As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: as mentioned in 00/54: either apply this directly or ACK it and I will take it into the GPIO tree. --- sound/soc/codecs/wm8903.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 2512def0d349..a82b8bc2cfc0 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1804,7 +1804,7 @@ static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg); - return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; + return !!((reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT); } static int wm8903_gpio_direction_out(struct gpio_chip *chip, -- 2.4.3

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[alsa-devel] [GIT PULL] ASoC updates for v4.5
by Mark Brown 23 Dec '15

23 Dec '15

The following changes since commit 4ef7675344d687a0ef5b0d7c0cee12da005870c0: Linux 4.4-rc6 (2015-12-20 16:06:09 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.5 for you to fetch changes up to 822ad70a2f5c420da5baa9f4354e6b7813ca6da9: Merge remote-tracking branch 'asoc/topic/rt5677' into asoc-next (2015-12-23 00:38:14 +0000) ---------------------------------------------------------------- ASoC: Updates for v4.5 This is quite a busy release on the driver front with a lot of new drivers being added but comparatively quiet on the core side with only one big change going in and that a fairly straightforward refactoring. - Conversion of the array of DAI links to a list by Mengdong Lin, supporting dynamically adding and removing DAI links. - Some more fixes for the topology code, though it is still not final and ready for enabling in production. We really need to get to the point where that can be done. - A pile of changes for Intel SkyLake drivers which hopefully deliver some useful initial functionality for systems with this chipset, though there is more work still to come. - New drivers for a number of Imagination Technologies IPs. - Lots of new features and cleanups for the Renesas drivers. - ANC support for WM5110. - New driver for Atmel class D speaker drivers. - New drivers for Cirrus CS47L24 and WM1831. - New driver for Dialog DA7128. - New drivers for Realtek RT5659 and RT56156. - New driver for Rockchip RK3036. - New driver for TI PC3168A ---------------------------------------------------------------- Adam Thomson (9): ASoC: da7218: Add bindings documentation for DA7218 audio codec ASoC: da7218: Add da7218 codec driver ASoC: da7218: Enable mic level detection reporting to user-space ASoC: da7219: Fix Sidetone to work regardless of DAI capture ASoC: da7219: Disable regulators on probe() failure ASoC: da7219: Update REFERENCES reg default, in-line with HW ASoC: da7219: Remove internal LDO features of codec ASoC: da7219: Add support for 1.6V micbias level ASoC: da7219: Remove support for 32KHz PLL mode Anatol Pomozov (1): ASoC: rt5677: use 'active low' logic for reset pin Arnd Bergmann (8): ASoC: samsung: pass DMA channels as pointers ASoC: s3c24xx-i2s: pass DMA channels as platform data ASoC: pxa: remove incorrect do_div() call ASoC: sh: fix fsi build warnings for 64 bit ASoC: samsung: pass filter function as pointer ASoC: rcar: remove unused variable ASoC: fsl: use correct format string for dma_addr_t ASoC: da7218: avoid 64-bit compile warning Axel Lin (3): ASoC: da7219: Use logical instead of bitwise OR for boolean expression ASoC: wm8904: Make undocumented registers non-readable ASoC: rt5616: Return error if device ID mismatch Bard Liao (7): ASoC: rt298: fix remove unnedded clk setting ASoC: rt5645: Separate regmap for rt5645 and rt5650 ASoC: rt5659: add rt5659 codec driver ASoC: rt5645: merge DMI tables of google projects ASoC: rt5616: add rt5616 codec driver ASoC: rt5616: rename some alsa control names ASoC: rt5677: set PLL_CTRL2 non-volatile Ben Zhang (1): ASoC: rt5677: Reconfigure PLL1 after resume Caesar Wang (5): ASoC: rockchip: i2s: change bclk and lrck according to sample rates ASoC: rockchip-max98090: Allow more sample rates ASoC: rockchip-rt5645: Allow more sample rates ASoC: rt5616: add an of_match table ASoC: rt5616: add devicetree document for rt5616 Charles Keepax (17): ASoC: wm5110: Add DAPM/routing hookup for the ANC block ASoC: wm_adsp: Expand the list of available firmwares ASoC: arizona: Add 32uS delay after putting FLL into freerun ASoC: dapm: Make enable/disable_pin work with always on widgets ASoC: arizona: Correct types of mixer texts and values ASoC: arizona: Fix type of clock rate pointer in arizona_set_sysclk ASoC: wm_adsp: Replace debugfs lock with more general DSP power lock ASoC: wm_adsp: Add power lock for firmware change control ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors ASoC: wm_adsp: Add locking to DSP firmware controls ASoC: wm_adsp: Mimic legacy behaviour of reading controls when DSP is on ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together Damien.Horsley (17): ASoC: img: Add binding document for I2S input controller ASoC: img: Add driver for I2S input controller ASoC: img: Add binding document for I2S output controller ASoC: img: Add driver for I2S output controller ASoC: img: Add binding document for parallel output controller ASoC: img: Add driver for parallel output controller ASoC: img: Add binding document for SPDIF input controller ASoC: img: Add driver for SPDIF input controller ASoC: img: Add documentation for SPDIF in controls ASoC: img: Add binding document for SPDIF output controller ASoC: img: Add driver for SPDIF output controller ASoC: img: parallel out: Add missing initialiser ASoC: Add SOC_DOUBLE_STS macro ASoC: pcm3168a: Add binding document for pcm3168a codec ASoC: pcm3168a: Add driver for pcm3168a codec ASoC: img: Add binding document for Pistachio internal DAC ASoC: img: Add driver for Pistachio internal DAC Dan Carpenter (3): ASoC: Intel: pass correct parameter in sst_alloc_stream_mrfld() ASoC: Intel: Skylake: Fix a couple signedness bugs ASoC: Intel: Skylake: pointer math issue Dharageswari R (2): ASoC: Intel: Skylake: Add support for Loadable modules ASoC: Intel: Skylake: Add support for Mic Select module Fang, Yang A (3): ASoC: Intel: Skylake: add adi + nau8825 machine driver entry ASoc: Intel: boards: fix dapm map of nau88l25_ssm4567 machine ASoc: Intel: boards: Add HDMI/DP links for nau88l25_ssm4567 machine Geert Uytterhoeven (2): ASoC: rsnd: Add missing initialization of ADG req_rate ASoC: rsrc-card: Clarify compatible value Hans de Goede (3): ASoC: sun4i-codec: Rename codec dapm widgets and routes ASoC: sun4i-codec: Add support for PA gpio pin ASoC: sun4i-codec: Use proper output for external amp routes Hardik T Shah (3): ASoC: Intel: Skylake: Add multiple pin formats ASoC: Intel: Skylake: Update the topology interface structure ASoC: Intel: Skylake: Add support for module GUIDs Harsha Priya (1): ASoC: Intel: Skylake: Adding nau88l25+ssm4567 machine driver Jean-Michel Hautbois (1): ASoC: sgtl5000: fix VAG power up timing Jeeja KP (39): ASoC: Intel: Skylake: Fix to correct check for non DSP widget ASoC: Intel: Skylake: Fix not to ignore return value in be hw_params ASoC: Intel: Skylake: Fix to add 32 bit in update FE params ASoC: Intel: Skylake: Fix to ignore codec_mask check in probe ASoC: Intel: Skylake: Fix to ignore blob check if link type is HDA ASoC: Intel: Skylake: Fix support for multiple pins in a module ASoC: Intel: Skylake: Fix bit depth when querying the NHLT blob ASoC: Intel: Skylake: use module_pin info for unbind ASoC: Intel: Skylake: Add support for virtual dsp widgets ASoC: Intel: Skylake: Fix DSP pipe underrun/overrun issue ASoC: Intel: Skylake: Fix to remove be copier widget power check ASoC: Intel: Skylake: Ignore rate check for DMIC link ASoC: Intel: Skylake: Fix to remove channel_map calculation ASoC: Intel: Skylake: Fix PM behaviour ASoC: Intel: Skylake: Fix the SSP0 Fmt fixup to 24 bit ASoC: Intel: Skylake: Add support for SSP1 BE cpu dai ASoC: Intel: Skylake: remove pm_runtime_get/put calls ASoC: Intel: Skylake: Don't enable WAKEENABLE on suspend ASoC: Intel: Skylake: Remove redundant init in resume ASoC: Intel: Skylake: Fix cleanup of dma buffer ASoC: Intel: Skylake: Reset the DSP when set D3 fails ASoC: Intel: Skylake: Fix CLDMA buffer wrap case ASoC: Intel: Skylake: Fix null ptr dereferenced in skl_tplg_bind_sinks ASoC: Intel: Skylake: Add pm ops for skl_rt286 machine ASoC: Intel: Skylake: Reconfigure HDA stream register in prepare/resume ASoC: Intel: Skylake: Fix to update bit depth for module params ASoC: Intel: Skylake: Update pcm capability ASoC: Intel: Skylake: Poll CLDMA RUN bit when set ASoC: Intel: Skylake: Update DMIC DAIs and capabilities ASoC: Intel: Skylake: Add helper routines to handle module params ASoC: Intel: Skylake: Add helper routine to handle Algo parameter ASoC: Intel: Skylake: Add support to configure module params ASoC: Intel: Skylake: Add tlv byte kcontrols ASoC: Intel: Skylake: Add memory pages to widget data. ASoC: Intel: Skylake: Fix module init data correctly ASoC: Intel: Skylake: Add dai link for DMIC capture ASoC: Intel: Skylake: Add support for active suspend ASoC: Intel: Skylake: Clear stream registers before stream setup ASoC: Intel: Skylake: Fix to set pipe state to invalid when deleting Jiada Wang (1): ASoC: wm8962: set ALC2 as non-volatile register Jie Yang (2): ASoC: Intel: sst: only select sst-firmware when DW DMAC is built-in ASoC: Intel: sst: fix sst_memcpy32 wrong with non-4x bytes issue John Keeping (4): ASoC: rockchip: i2s: separate capture and playback ASoC: rockchip: i2s: remove unused variables ASoC: es8328: Fix deemphasis values ASoC: es8328: Fix shifts for mixer switches Julia Lawall (4): ASoC: Intel: constify sst_block_ops structures ASoC: imx-pcm-dma: add NULL test ASoC: omap-hdmi-audio: add NULL test ASoC: Intel: add NULL test Koro Chen (4): ASoC: dpcm: Make BE prepare possible in suspend state ASoC: mediatek: Move 22M/24M clock control into I2S ops ASoC: mediatek: Use current HW pointer for pointer callback ASoC: mediatek: Turn AFE on/off in runtime resume/suspend Kuninori Morimoto (82): ASoC: rsnd: DMA become SSI/SRC member ASoC: rsnd: DMA related definition goes to dma.c ASoC: rsnd: rename rsnd_dma_init() to rsnd_dma_attach() ASoC: rsnd: enable to use rsnd_dai_connect() from each mod ASoC: rsnd: remove all modules when PIO fallback ASoC: rsnd: fixup rsnd_dma_of_path method for mod base common method ASoC: rsnd: move rsnd_src_ssi_irq_enable/disable() to ssi.c ASoC: rsnd: disable SRC.out only when stop timing ASoC: rsnd: rsnd_dai_stream has each mod's status insted of rsnd_mod ASoC: rsnd: Don't stop HW even if a large number of errors occur ASoC: rsnd: avoid pointless loop in rsnd_mod_interrupt() ASoC: rsnd: use common rsnd_ssi_status_xxx() ASoC: rsnd: use mod base common method on DMA phase1 ASoC: rsnd: use mod base common method on DMA phase2 ASoC: rsnd: use mod base common method on DMA phase3 ASoC: rsnd: use mod base common method on CMD ASoC: rsnd: use mod base common method on SSIU ASoC: rsnd: use mod base common method on SSI-parent ASoC: rsnd: call rsnd_src_quit() from rsnd_src_quit_gen2() ASoC: rsnd: fixup rsnd_dmapp_stop() return value ASoC: rsnd: audio_clkout0/1/2/3 are optional properties ASoC: rsnd: move CMD related operation to cmd.c ASoC: rsnd: DVC settings matches to datasheet ASoC: rsnd: MIX settings matches to datasheet ASoC: rsnd: tidyup comment position of rsnd_mod_xxx ASoC: rsnd: call clk_prepare()/clk_enable() for AUDIO_CLKx ASoC: ak4613: add single-end optional property for IN/OUT pins ASoC: ak4613: tidyup CTRL1 value selection method ASoC: rsnd: SND_SOC_RCAR doesn't depend on DMA_OF ASoC: rsnd: ssi: 24bit data needs right-aligned settings ASoC: ak4613: don't overwrite CTRL2 register ASoC: rsnd: remove Gen1 support from SRC ASoC: rsnd: cleanup RSND_REG_xxx ASoC: rsnd: SRC settings matches to datasheet ASoC: rsnd: remove platform boot support from core.c ASoC: rsnd: remove platform boot support from ssi.c ASoC: rsnd: remove platform boot support from src.c ASoC: rsnd: remove platform boot support from ctu.c ASoC: rsnd: remove platform boot support from mix.c ASoC: rsnd: remove platform boot support from dvc.c ASoC: rsnd: remove platform boot support from gen.c ASoC: rsnd: remove struct rsnd_of_data ASoC: rsnd: remove struct rcar_snd_info ASoC: rsnd: remove struct platform_device from probe/remove parameter ASoC: ak4613: tidyup vendor prefix from ak4613 to asahi-kasei ASoC: rsnd: tidyup void* cast for 64bit compiler ASoC: rsnd: indicate register name for debug ASoC: rsnd: tidyup semantics of rsnd_src_record_error() ASoC: rsnd: tidyup semantics of rsnd_ssi_record_error() ASoC: rsnd: don't use normal *mod in adg.c ASoC: rsnd: add missing ADINR::CHNUM on DVC/SRC/SSIU ASoC: rsnd: add missing SRC_O_BUSIF_MODE register ASoC: rsnd: src: rename rsnd_src_soft_reset() to rsnd_src_activation() ASoC: rsnd: mix: rename rsnd_mix_soft_reset() to rsnd_mix_activation() ASoC: rsnd: dvc: rename rsnd_dvc_soft_reset() to rsnd_dvc_activation() ASoC: rsnd: src: add rsnd_src_halt() ASoC: rsnd: mix: add rsnd_mix_halt() ASoC: rsnd: dvc: add rsnd_dvc_halt() ASoC: rsnd: add rsnd_ssi_config_init() ASoC: rsnd: set SSIWSR setting on rsnd_ssi_config_init() ASoC: rsnd: ssi enables non-stereo sound ASoC: rsnd: dvc enables non-stereo sound ASoC: rsnd: add TDM Extend Mode support ASoC: rsnd: fixup wrong snd_soc_dai_driver pointer access ASoC: rsnd: rsrc-card: check return value of snd_soc_of_get_dai_name() ASoC: rsnd: care SWSP bit for TDM/non-TDM ASoC: rsnd: tidyup data align position for capture ASoC: rsrc-card: enable to use tdm_slot on DT ASoC: rsrc-card: Remove support for setting differing DAI formats ASoC: rsrc-card: tidyup dai format for DPCM ASoC: rsnd: fixup SSIU control timing ASoC: rsnd: tidyup return value of rsnd_get_adinr_bit() ASoC: rsnd: tidyup debug print position on rsnd_dma_attach() ASoC: rsnd: rsnd_dai_connect() returns error if it connect to existing mod ASoC: rsnd: SSI/SSIU use rsnd_get_slot_extend() to check TDM ASoC: rsnd: remove rsnd_get_slot_runtime() ASoC: rsnd: tidyup rsnd_get_slot_xxx() naming ASoC: rsnd: add rsnd_set_slot() / rsnd_get_slot_num() ASoC: rsnd: add rsnd_parse_connect_common() and remove complex macro ASoC: rsnd: add missing DT example for Simple Card ASoC: rsnd: add missing DT example for Simple Card with TDM ASoC: rsnd: add Multi channel support Liam Girdwood (1): ASoC: Intel: sst: fix the IRQ locked issue Linus Walleij (2): ASoC: wm8903: Be sure to clamp return value ASoC: ac97: Be sure to clamp return value Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: use different route map for AC'97 mode Mans Rullgard (3): ASoC: wm8974: set cache type for regmap ASoC: wm8974: add devicetree support ASoC: wm8974: add binding for WM8974 codec Mark Brown (37): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into HEAD Merge branch 'fix/arizona' of git://git.kernel.org/.../broonie/sound into asoc-arizona Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/pcm-list' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/hdac' of git://git.kernel.org/.../broonie/sound into asoc-intel ASoC: hdac: Fix Makefile and Kconfig sorting Merge branch 'topic/fsl-sai' of git://git.kernel.org/.../broonie/sound into asoc-fsl Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge branch 'topic/sink' of git://git.kernel.org/.../broonie/sound into asoc-dapm Merge branch 'fix/sun4i-codec' of git://git.kernel.org/.../broonie/sound into asoc-sunxi Merge branch 'topic/sink' of git://git.kernel.org/.../broonie/sound into asoc-intel rcar: ctu: Avoid use of ret uninitialised rcar: dvc: Avoid use of ret uninitialised rcar: mux: Avoid use of ret uninitialised Merge branch 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-adsp Merge branch 'topic/cs47l24' of git://git.kernel.org/.../broonie/sound into asoc-adsp Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm-list' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branch 'asoc/topic/sunxi' into asoc-next Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/adsp', 'asoc/topic/ak4613' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel-classd', 'asoc/topic/const' and 'asoc/topic/da7218' into asoc-next Merge remote-tracking branches 'asoc/topic/da7219', 'asoc/topic/dai-link' and 'asoc/topic/doc' into asoc-next Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/dwc', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc' and 'asoc/topic/fsl-esai' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/img' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/kcontrol', 'asoc/topic/max98357a' and 'asoc/topic/mtk' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar', 'asoc/topic/rk3036' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/rt286', 'asoc/topic/rt5616' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sh', 'asoc/topic/ssm2518' and 'asoc/topic/sti' into asoc-next Merge remote-tracking branches 'asoc/topic/tegra', 'asoc/topic/topology', 'asoc/topic/wm8903' and 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8962', 'asoc/topic/wm8974' and 'asoc/topic/wm9713' into asoc-next Merge tag 'asoc-fix-v4.4-rc6' into asoc-linus Merge tag 'asoc-v4.5' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5677' into asoc-next Markus Elfring (1): ASoC: ssm2518: Use a signed return type for ssm2518_lookup_mcs() Maruthi Srinivas Bayyavarapu (3): ASoC: dwc: add runtime suspend/resume functionality ASoC: dwc: add quirk for different register offset ASoC: dwc: reconfigure dwc in 'resume' from 'suspend' Maxime Ripard (3): ASoC: sun4i: Implement MIC1 capture ASoC: sunxi: Remove useless comments and variable ASoC: sun4i-codec: pass through clk_set_rate error Mengdong Lin (9): ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest ASoC: Vendor drivers get a link's runtime by snd_soc_get_pcm_runtime() ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list & define API to add/remove a link ASoC: Define add/remove_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components Moise Gergaud (1): ASoC: sti: set iec958 channel status sampling freq Mousami Jana (1): ASoC: Intel: Skylake: add LARGE_CONFIG_GET IPC support Mythri P K (1): ASoC: core: Pass kcontrol to bytes tlv callbacks Oder Chiou (1): ASoC: rt5645: Use the mod_delayed_work instead of the queue_delayed_work and cancel_delayed_work_sync Omair M Abdullah (2): ASoC: Intel: Skylake: update mailbox uplink window offset and size ASoC: Intel: Skylake: read params from DSP if module is on PC Liao (1): ASoC: dpcm: Apply symmetry for DPCM Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx Pierre-Louis Bossart (9): ASoC: Intel: boards: align pin names between byt-rt5640 drivers ASoC: Intel: boards: start merging byt-rt5640 drivers ASoC: Intel: boards: merge DMI-based quirks in bytcr-rt5640 driver ASoC: Intel: tag byt-rt5640 machine driver as deprecated ASoc: Intel: Atom: add deep buffer definitions for atom platforms ASoC: Intel: boards: add DEEP_BUFFER support for BYT/CHT/BSW ASoC: Intel: Atom: add 24-bit support for media playback and capture ASoC: Intel: Atom: clean-up compressed DAI definition ASoC: Intel: Atom: flip logic for gain Switch Praveen Diwakar (1): ASoC: Intel: Skylake: Update ignore suspend for rt286 machine Richard Fitzgerald (3): ASoC: wm8998: Remove duplicated consts ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs ASoC: arizona: In arizona_calc_fratio make new codecs the default case Robert Jarzmik (3): ASoC: wm9713: convert to regmap ASoC: wm9713: use snd_soc_*() calls to update ac97 registers ASoC: ac97: add gpio chip Rohit Ainapure (3): ASoC: max98357a: Add ACPI ID for Maxim ASoC: Intel: Add Nuvoton+Maxim machine driver entry ASoC: Intel: Skylake: Add Nuvoton Maxim machine driver Sathyanarayana Nujella (1): ASoc: Intel: boards: Add WOV as sink for nau88l25_ssm4567 machine Sebastien Guiriec (1): ASoC: Intel: bytcr_rt5640: set SSP to I2S mode 2ch Shengjiu Wang (3): ASoC: fsl_esai: spba clock is needed by esai device ASoC: fsl_spdif: spba clk is needed by spdif device ASoC: fsl_asrc: spba clock is needed by asrc device Simon Horman (1): ASoC: rsnd: Add device tree support for r8a77{79,93,94} Sjoerd Simons (1): ASoC: rockchip: spdif: Set transmit data level to 16 samples Songjun Wu (6): ASoC: Atmel: ClassD: supports mono audio ASoC: Atmel: ClassD: add GCK's parent clock in DT binding ASoC: Atmel: ClassD: unregister codec when error occurs ASoC: atmel-pdmic: add the Pulse Density Modulation Interface Controller ASoC: atmel-classd: DT binding for PDMIC driver ASoC: atmel_wm8904: add snd_soc_pm_ops Stephen Boyd (1): ASoC: qcom: Specify LE device endianness Subhransu S. Prusty (7): ASoC: Intel: Skylake: Fix to cleanup if skl_sst_dsp_init fails ALSA: hdac: structure definition for ext_dma_params ASoC: hdac-hdmi: Add hdmi driver ASoC: hdac_hdmi: Add PM support for HDMI ASoC: hdac_hdmi: Add hdac hdmi dai ops ASoC: hdac_hdmi: Setup and start infoframe ASoC: hdac_hdmi: Use i915 component framework for PM Sudip Mukherjee (3): ASoC: hdac_hdmi: fix possible NULL dereference ASoC: tegra_alc5632: check return value ASoC: hdac_hdmi: check error return Sugar Zhang (2): ASoC: rockchip: i2s: compatible with different chips ASoC: rockchip: add playback property Takashi Iwai (1): ASoC: Fix typo in kernel doc comment for snd_soc_put_volsw_sx() Vincent Stehlé (1): ASoC: Intel: Skylake: fix typo in sizeof Vinod Koul (20): ASoC: Intel: Skylake: Fix substream dereference before check ASoC: Intel: Skylake: Fix resource cleanup on teardown ASoC: Intel: Move apci find machine routines ASoC: Intel: Atom: move atom driver to common acpi match ASoC: Intel: Skylake: Fix skl machine driver creation ASoC: Intel: add fw name to common dsp context ASoC: Intel: Skylake: Use the fw name from ACPI mach table ASoC: rt286: set combo jack for Skylake ASoC: core: refactor soc_link_dai_widgets() ASoC: Intel: Skylake: Add I2C depends for SKL machine ASoC: Intel: Skylake: Constrain the audio devices ASoC: fsl-asoc-card: Update the rtd query ASoC: Intel: Skylake: Update the rtd query ASoC: Intel: Skylake: Fix test of a field address ASoC: hdac-hdmi: make driver select CONFIG_HDMI ASoC: dapm: add a dapm sink widget ASoC: core: mark SND_SOC_BYTES_EXT as deprecated ASoC: Intel: Skylake: Move up pipe mem free ASoC: Intel: Skylake: add wov as int sink ASoC: Intel: Skylake: Fix the dapm machine map Yong Zhi (2): ASoc: Intel: boards: update ignore suspend for nau88l25_ssm4567 machine ASoc: Intel: boards: update constraints for nau88l25_ssm4567 machine ZhengShunQian (2): ASoC: rk3036: Inno codec driver for RK3036 SoC ASoC: rk3036: Add binding doc of inno-rk3036 codec driver Zidan Wang (10): ASoC: fsl_esai: ETDR and TX0~5 registers are non volatile ASoC: fsl-sai: add default register map for regmap cache ASoC: fsl_spdif: STL and STR registers are non volatile ASoC: fsl_ssi: using macro for default register map ASoC: fsl-sai: don't set bclk for Tx/Rx Synchronous with another SAI mode ASoC: fsl-sai: set xCR4/xCR5/xMR for SAI master mode ASoC: fsl_sai: add tdm slots operation support ASoC: fsl: using params_width function to simplify code ASoC: fsl_sai: fix no frame clk in master mode ASoC: fsl_asrc: add support for 8-30kHz output sample rate kbuild test robot (2): ASoC: rk3036: fix platform_no_drv_owner.cocci warnings ASoC: da7218: fix boolreturn.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 10 + .../devicetree/bindings/sound/atmel-classd.txt | 6 + .../devicetree/bindings/sound/atmel-pdmic.txt | 55 + Documentation/devicetree/bindings/sound/da7218.txt | 104 + Documentation/devicetree/bindings/sound/da7219.txt | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 5 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/img,i2s-in.txt | 47 + .../devicetree/bindings/sound/img,i2s-out.txt | 51 + .../devicetree/bindings/sound/img,parallel-out.txt | 44 + .../bindings/sound/img,pistachio-internal-dac.txt | 18 + .../devicetree/bindings/sound/img,spdif-in.txt | 41 + .../devicetree/bindings/sound/img,spdif-out.txt | 44 + .../devicetree/bindings/sound/inno-rk3036.txt | 20 + .../devicetree/bindings/sound/renesas,rsnd.txt | 82 + .../bindings/sound/renesas,rsrc-card.txt | 4 +- .../devicetree/bindings/sound/rockchip-i2s.txt | 2 + Documentation/devicetree/bindings/sound/rt5616.txt | 26 + Documentation/devicetree/bindings/sound/rt5659.txt | 75 + Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- .../devicetree/bindings/sound/sun4i-codec.txt | 3 + .../devicetree/bindings/sound/ti,pcm3168a.txt | 48 + .../devicetree/bindings/sound/wlf,wm8974.txt | 15 + Documentation/sound/alsa/img,spdif-in.txt | 49 + arch/arm/mach-s3c64xx/dev-audio.c | 47 +- arch/arm/mach-s3c64xx/include/mach/dma.h | 52 +- arch/arm/plat-samsung/devs.c | 23 +- arch/x86/include/asm/platform_sst_audio.h | 1 + drivers/dma/Kconfig | 2 +- include/linux/platform_data/asoc-s3c.h | 8 + include/sound/ac97_codec.h | 3 + include/sound/da7218.h | 109 + include/sound/da7219.h | 14 +- include/sound/designware_i2s.h | 5 + include/sound/hdaudio_ext.h | 6 + include/sound/rt5659.h | 49 + include/sound/soc-dapm.h | 4 + include/sound/soc-topology.h | 6 +- include/sound/soc.h | 42 +- include/uapi/sound/asoc.h | 2 +- include/uapi/sound/compress_params.h | 5 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 26 +- sound/soc/atmel/atmel-pdmic.c | 738 ++++ sound/soc/atmel/atmel-pdmic.h | 80 + sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/codecs/Kconfig | 56 +- sound/soc/codecs/Makefile | 18 + sound/soc/codecs/ak4613.c | 118 +- sound/soc/codecs/arizona.c | 143 +- sound/soc/codecs/arizona.h | 17 +- sound/soc/codecs/cs47l24.c | 1148 ++++++ sound/soc/codecs/cs47l24.h | 23 + sound/soc/codecs/da7218.c | 3314 +++++++++++++++ sound/soc/codecs/da7218.h | 1414 +++++++ sound/soc/codecs/da7219.c | 89 +- sound/soc/codecs/da7219.h | 9 - sound/soc/codecs/es8328.c | 25 +- sound/soc/codecs/es8328.h | 1 + sound/soc/codecs/hdac_hdmi.c | 659 +++ sound/soc/codecs/inno_rk3036.c | 490 +++ sound/soc/codecs/inno_rk3036.h | 123 + sound/soc/codecs/max98357a.c | 10 + sound/soc/codecs/pcm3168a-i2c.c | 66 + sound/soc/codecs/pcm3168a-spi.c | 65 + sound/soc/codecs/pcm3168a.c | 767 ++++ sound/soc/codecs/pcm3168a.h | 100 + sound/soc/codecs/rt286.c | 6 + sound/soc/codecs/rt298.c | 2 - sound/soc/codecs/rt5616.c | 1381 +++++++ sound/soc/codecs/rt5616.h | 1819 +++++++++ sound/soc/codecs/rt5645.c | 292 +- sound/soc/codecs/rt5659.c | 4223 ++++++++++++++++++++ sound/soc/codecs/rt5659.h | 1819 +++++++++ sound/soc/codecs/rt5677.c | 13 +- sound/soc/codecs/sgtl5000.c | 1 + sound/soc/codecs/ssm2518.c | 2 +- sound/soc/codecs/wm5110.c | 206 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8962.c | 3 +- sound/soc/codecs/wm8974.c | 8 + sound/soc/codecs/wm8998.c | 46 +- sound/soc/codecs/wm9713.c | 296 +- sound/soc/codecs/wm_adsp.c | 769 +++- sound/soc/codecs/wm_adsp.h | 23 +- sound/soc/davinci/davinci-mcasp.c | 4 +- sound/soc/dwc/designware_i2s.c | 113 +- sound/soc/fsl/fsl-asoc-card.c | 21 +- sound/soc/fsl/fsl_asrc.c | 55 +- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_esai.c | 63 +- sound/soc/fsl/fsl_sai.c | 116 +- sound/soc/fsl/fsl_sai.h | 3 + sound/soc/fsl/fsl_spdif.c | 35 +- sound/soc/fsl/fsl_ssi.c | 25 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 4 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/img/Kconfig | 52 + sound/soc/img/Makefile | 7 + sound/soc/img/img-i2s-in.c | 516 +++ sound/soc/img/img-i2s-out.c | 565 +++ sound/soc/img/img-parallel-out.c | 327 ++ sound/soc/img/img-spdif-in.c | 806 ++++ sound/soc/img/img-spdif-out.c | 441 ++ sound/soc/img/pistachio-internal-dac.c | 287 ++ sound/soc/intel/Kconfig | 46 +- sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 32 +- sound/soc/intel/atom/sst/sst_acpi.c | 67 +- sound/soc/intel/atom/sst/sst_stream.c | 2 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 2 + sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bytcr_rt5640.c | 241 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 19 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 485 +++ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 536 +++ sound/soc/intel/boards/skl_rt286.c | 128 +- sound/soc/intel/common/Makefile | 7 +- sound/soc/intel/common/sst-acpi.c | 41 +- sound/soc/intel/common/sst-acpi.h | 33 + sound/soc/intel/common/sst-dsp-priv.h | 8 +- sound/soc/intel/common/sst-dsp.c | 2 +- sound/soc/intel/common/sst-dsp.h | 2 +- sound/soc/intel/common/sst-firmware.c | 20 +- sound/soc/intel/common/sst-match-acpi.c | 43 + sound/soc/intel/haswell/sst-haswell-dsp.c | 2 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 31 +- sound/soc/intel/skylake/skl-messages.c | 280 +- sound/soc/intel/skylake/skl-nhlt.c | 19 +- sound/soc/intel/skylake/skl-pcm.c | 256 +- sound/soc/intel/skylake/skl-sst-cldma.c | 97 +- sound/soc/intel/skylake/skl-sst-dsp.h | 21 +- sound/soc/intel/skylake/skl-sst-ipc.c | 108 +- sound/soc/intel/skylake/skl-sst-ipc.h | 9 + sound/soc/intel/skylake/skl-sst.c | 217 +- sound/soc/intel/skylake/skl-topology.c | 658 ++- sound/soc/intel/skylake/skl-topology.h | 63 +- sound/soc/intel/skylake/skl-tplg-interface.h | 105 +- sound/soc/intel/skylake/skl.c | 158 +- sound/soc/intel/skylake/skl.h | 6 +- sound/soc/mediatek/mtk-afe-common.h | 1 - sound/soc/mediatek/mtk-afe-pcm.c | 59 +- sound/soc/omap/omap-hdmi-audio.c | 2 + sound/soc/pxa/brownstone.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/qcom/lpass-cpu.c | 1 + sound/soc/rockchip/rockchip_i2s.c | 130 +- sound/soc/rockchip/rockchip_max98090.c | 6 + sound/soc/rockchip/rockchip_rt5645.c | 6 + sound/soc/rockchip/rockchip_spdif.c | 6 +- sound/soc/rockchip/rockchip_spdif.h | 2 +- sound/soc/samsung/Kconfig | 2 - sound/soc/samsung/ac97.c | 29 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/dma.h | 6 +- sound/soc/samsung/dmaengine.c | 20 +- sound/soc/samsung/i2s.c | 31 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/pcm.c | 25 +- sound/soc/samsung/s3c2412-i2s.c | 16 +- sound/soc/samsung/s3c24xx-i2s.c | 16 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/spdif.c | 17 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/sh/Kconfig | 1 - sound/soc/sh/fsi.c | 11 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 118 +- sound/soc/sh/rcar/cmd.c | 171 + sound/soc/sh/rcar/core.c | 586 +-- sound/soc/sh/rcar/ctu.c | 99 +- sound/soc/sh/rcar/dma.c | 245 +- sound/soc/sh/rcar/dvc.c | 273 +- sound/soc/sh/rcar/gen.c | 133 +- sound/soc/sh/rcar/mix.c | 158 +- sound/soc/sh/rcar/rcar_snd.h | 117 - sound/soc/sh/rcar/rsnd.h | 335 +- sound/soc/sh/rcar/rsrc-card.c | 129 +- sound/soc/sh/rcar/src.c | 898 ++--- sound/soc/sh/rcar/ssi.c | 744 ++-- sound/soc/sh/rcar/ssiu.c | 225 ++ sound/soc/soc-ac97.c | 125 + sound/soc/soc-core.c | 602 ++- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-ops.c | 4 +- sound/soc/soc-pcm.c | 82 +- sound/soc/sti/uniperif_player.c | 3 +- sound/soc/sunxi/sun4i-codec.c | 279 +- sound/soc/tegra/tegra_alc5632.c | 12 +- sound/soc/tegra/tegra_wm8903.c | 3 +- 202 files changed, 30532 insertions(+), 4175 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt create mode 100644 Documentation/devicetree/bindings/sound/da7218.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,parallel-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,pistachio-internal-dac.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-out.txt create mode 100644 Documentation/devicetree/bindings/sound/inno-rk3036.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5659.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm3168a.txt create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.txt create mode 100644 Documentation/sound/alsa/img,spdif-in.txt create mode 100644 include/sound/da7218.h create mode 100644 include/sound/rt5659.h create mode 100644 sound/soc/atmel/atmel-pdmic.c create mode 100644 sound/soc/atmel/atmel-pdmic.h create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h create mode 100644 sound/soc/codecs/da7218.c create mode 100644 sound/soc/codecs/da7218.h create mode 100644 sound/soc/codecs/hdac_hdmi.c create mode 100644 sound/soc/codecs/inno_rk3036.c create mode 100644 sound/soc/codecs/inno_rk3036.h create mode 100644 sound/soc/codecs/pcm3168a-i2c.c create mode 100644 sound/soc/codecs/pcm3168a-spi.c create mode 100644 sound/soc/codecs/pcm3168a.c create mode 100644 sound/soc/codecs/pcm3168a.h create mode 100644 sound/soc/codecs/rt5616.c create mode 100644 sound/soc/codecs/rt5616.h create mode 100644 sound/soc/codecs/rt5659.c create mode 100644 sound/soc/codecs/rt5659.h create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/img-i2s-in.c create mode 100644 sound/soc/img/img-i2s-out.c create mode 100644 sound/soc/img/img-parallel-out.c create mode 100644 sound/soc/img/img-spdif-in.c create mode 100644 sound/soc/img/img-spdif-out.c create mode 100644 sound/soc/img/pistachio-internal-dac.c create mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c create mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c create mode 100644 sound/soc/intel/common/sst-acpi.h create mode 100644 sound/soc/intel/common/sst-match-acpi.c create mode 100644 sound/soc/sh/rcar/cmd.c delete mode 100644 sound/soc/sh/rcar/rcar_snd.h create mode 100644 sound/soc/sh/rcar/ssiu.c

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.4
by Mark Brown 23 Dec '15

23 Dec '15

The following changes since commit 4ef7675344d687a0ef5b0d7c0cee12da005870c0: Linux 4.4-rc6 (2015-12-20 16:06:09 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-fix-v4.4-rc6 for you to fetch changes up to 3dd5fc0eeb5d7cc07b8e3e383037261799b38897: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus (2015-12-23 00:23:27 +0000) ---------------------------------------------------------------- ASoC: Fixes for v4.4 A collection of small driver specific fixes here, nothing that'll affect users who don't have the devices concerned. At least the wm8974 bug indicates that there's not too many users of some of these devices. ---------------------------------------------------------------- Jean-Michel Hautbois (1): ASoC: sgtl5000: fix VAG power up timing John Keeping (2): ASoC: es8328: Fix deemphasis values ASoC: es8328: Fix shifts for mixer switches Mans Rullgard (1): ASoC: wm8974: set cache type for regmap Mark Brown (1): Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/es8328', 'asoc/fix/fsl-sai', 'asoc/fix/rockchip', 'asoc/fix/sgtl5000' and 'asoc/fix/wm8974' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix XDATA check in mcasp_start_tx Sjoerd Simons (1): ASoC: rockchip: spdif: Set transmit data level to 16 samples Zidan Wang (1): ASoC: fsl_sai: fix no frame clk in master mode sound/soc/codecs/es8328.c | 25 +++++++++++++++++-------- sound/soc/codecs/es8328.h | 1 + sound/soc/codecs/sgtl5000.c | 1 + sound/soc/codecs/wm8974.c | 1 + sound/soc/davinci/davinci-mcasp.c | 4 ++-- sound/soc/fsl/fsl_sai.c | 18 ++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.c | 6 ++++-- sound/soc/rockchip/rockchip_spdif.h | 2 +- 8 files changed, 45 insertions(+), 13 deletions(-)

2 2

[alsa-devel] [PATCH 0/4] add NULL test
by Julia Lawall 23 Dec '15

23 Dec '15

4 10

[alsa-devel] [PATCH] ASoC: rt5677: set PLL_CTRL2 non-volatile
by Bard Liao 23 Dec '15

23 Dec '15

There is a status bit on RT5677_PLL1_CTRL2 and RT5677_PLL2_CTRL2. That's why those registers are set volatile. However, the status bit is currently not used by codec driver. So, it should be no problem if we set them non-volatile. The purpose of setting them non-volatile is to restore the setting after a syspend/resume cycle. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, Please revert the "ASoC: rt5677: Reconfigure PLL1 after resume" patch before applying this patch. Thanks. --- sound/soc/codecs/rt5677.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f73fd12..13fef00 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -297,8 +297,6 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg) case RT5677_HAP_GENE_CTRL2: case RT5677_PWR_DSP_ST: case RT5677_PRIV_DATA: - case RT5677_PLL1_CTRL2: - case RT5677_PLL2_CTRL2: case RT5677_ASRC_22: case RT5677_ASRC_23: case RT5677_VAD_CTRL5: -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/8] Add support for voice control on Arizona ADSP
by Charles Keepax 23 Dec '15

23 Dec '15

This series add support for voice control on wm8280/5110. This is done by opening a compressed record channel, upon which data will be available once the voice control functionality has been triggered on the DSP. Changes since v1: - Merged selecting of SND_SOC_COMPRESS into patch adding basic hookup as suggested by Vinod. - Fixed a minor typo in usleep_delay, which was using a longer delay than required. - Use kcalloc to allocate an array rather than caculating size manually. Thanks, Charles Charles Keepax (8): ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together ASoC: wm_adsp: Add a handler for the compressed IRQ ASoC: wm_adsp: Pull data through compressed read include/uapi/sound/compress_params.h | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 95 +++- sound/soc/codecs/wm_adsp.c | 907 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 21 + 6 files changed, 1010 insertions(+), 21 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH 01/13] ASoC: wm_adsp: Fixup some minor formatting and checkpatch errors
by Charles Keepax 23 Dec '15

23 Dec '15

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++-------------- sound/soc/codecs/wm_adsp.h | 4 ++-- 2 files changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 905ae99..3808190 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -544,10 +544,10 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } -static int wm_coeff_info(struct snd_kcontrol *kcontrol, +static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -593,10 +593,10 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_put(struct snd_kcontrol *kcontrol, +static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; memcpy(ctl->cache, p, ctl->len); @@ -647,10 +647,10 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kcontrol, +static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; + struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; char *p = ucontrol->value.bytes.data; if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { @@ -829,8 +829,7 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, break; } - list_for_each_entry(ctl, &dsp->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) ctl->enabled = 1; @@ -1109,7 +1108,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) goto out_fw; } - header = (void*)&firmware->data[0]; + header = (void *)&firmware->data[0]; if (memcmp(&header->magic[0], "WMFW", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); @@ -1189,7 +1188,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) offset = le32_to_cpu(region->offset) & 0xffffff; type = be32_to_cpu(region->type) & 0xff; mem = wm_adsp_find_region(dsp, type); - + switch (type) { case WMFW_NAME_TEXT: region_name = "Firmware name"; @@ -1646,7 +1645,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) goto out_fw; } - hdr = (void*)&firmware->data[0]; + hdr = (void *)&firmware->data[0]; if (memcmp(hdr->magic, "WMDR", 4) != 0) { adsp_err(dsp, "%s: invalid magic\n", file); goto out_fw; @@ -1672,7 +1671,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) blocks = 0; while (pos < firmware->size && pos - firmware->size > sizeof(*blk)) { - blk = (void*)(&firmware->data[pos]); + blk = (void *)(&firmware->data[pos]); type = le16_to_cpu(blk->type); offset = le16_to_cpu(blk->offset); @@ -1816,7 +1815,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; - int val; + unsigned int val; dsp->card = codec->component.card; @@ -1829,7 +1828,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, * For simplicity set the DSP clock rate to be the * SYSCLK rate rather than making it configurable. */ - if(dsp->sysclk_reg) { + if (dsp->sysclk_reg) { ret = regmap_read(dsp->regmap, dsp->sysclk_reg, &val); if (ret != 0) { adsp_err(dsp, "Failed to read SYSCLK state: %d\n", diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 2d117cf..89b7d71 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -45,8 +45,8 @@ struct wm_adsp { struct list_head alg_regions; - int fw_id; - int fw_id_version; + unsigned int fw_id; + unsigned int fw_id_version; const struct wm_adsp_region *mem; int num_mems; -- 2.1.4

3 22

[alsa-devel] [PATCH v2] ALSA: compress: Add SND_AUDIOCODEC_BESPOKE
by Charles Keepax 23 Dec '15

23 Dec '15

When working with the compressed framework occasionally vendors will use esoteric internal audio formats. For such formats it doesn't really make sense to add an new define to the kernel as their use is not sufficiently general. This patch adds a new define SND_AUDIOCODEC_BESPOKE that vendors can use in such situations. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Added a comment to note that users of BESPOKE are free to use the reserved space in snd_enc_generic. Thanks, Charles include/uapi/sound/compress_params.h | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index d9bd9ca..9625484 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -73,7 +73,8 @@ #define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B) #define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C) #define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729 +#define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_BESPOKE /* * Profile and modes are listed with bit masks. This allows for a @@ -312,7 +313,7 @@ struct snd_enc_flac { struct snd_enc_generic { __u32 bw; /* encoder bandwidth */ - __s32 reserved[15]; + __s32 reserved[15]; /* Can be used for SND_AUDIOCODEC_BESPOKE */ } __attribute__((packed, aligned(4))); union snd_codec_options { -- 1.7.2.5

3 2

[alsa-devel] [PATCH 53/54] sound: soc: ac97: Be sure to clamp return value
by Linus Walleij 23 Dec '15

23 Dec '15

As we want gpio_chip .get() calls to be able to return negative error codes and propagate to drivers, we need to go over all drivers and make sure their return values are clamped to [0,1]. We do this by using the ret = !!(val) design pattern. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Mark: as mentioned in 00/54: either apply this directly or ACK it and I will take it into the GPIO tree. --- sound/soc/soc-ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index a2012652f212..7e0acd83b0e6 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -92,7 +92,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) dev_dbg(codec->dev, "get gpio %d : %d\n", offset, ret < 0 ? ret : ret & (1 << offset)); - return ret < 0 ? ret : ret & (1 << offset); + return ret < 0 ? ret : !!(ret & (1 << offset)); } static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, -- 2.4.3

2 1

[alsa-devel] [PATCH] ASoC: rt5616: Return error if device ID mismatch
by Axel Lin 23 Dec '15

23 Dec '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..0e9414a 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1314,7 +1314,7 @@ static int rt5616_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Device with ID register %#x is not rt5616\n", val); - ret = -ENODEV; + return -ENODEV; } regmap_write(rt5616->regmap, RT5616_RESET, 0); regmap_update_bits(rt5616->regmap, RT5616_PWR_ANLG1, -- 2.1.4

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Turn AFE on/off in runtime resume/suspend
by Koro Chen 23 Dec '15

23 Dec '15

AFE is actually allowed to be turn on before configuration of DAIs since each DAI has its own enabling control. Turn on/off AFE in runtime resume/suspend to avoid AFE being shut down when closing a DAI while other DAIs are still active. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 24 ++++++------------------ 1 file changed, 6 insertions(+), 18 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 5399a0e..08af9f5 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -382,9 +382,6 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -433,9 +430,6 @@ static void mtk_afe_hdmi_shutdown(struct snd_pcm_substream *substream, mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S3_M], afe->clocks[MTK_CLK_I2S3_B]); - - /* disable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); } static int mtk_afe_hdmi_prepare(struct snd_pcm_substream *substream, @@ -679,17 +673,6 @@ static int mtk_afe_dais_hw_free(struct snd_pcm_substream *substream, return snd_pcm_lib_free_pages(substream); } -static int mtk_afe_dais_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct mtk_afe *afe = snd_soc_platform_get_drvdata(rtd->platform); - - /* enable AFE */ - regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); - return 0; -} - static int mtk_afe_dais_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -757,7 +740,6 @@ static const struct snd_soc_dai_ops mtk_afe_dai_ops = { .shutdown = mtk_afe_dais_shutdown, .hw_params = mtk_afe_dais_hw_params, .hw_free = mtk_afe_dais_hw_free, - .prepare = mtk_afe_dais_prepare, .trigger = mtk_afe_dais_trigger, }; @@ -1118,6 +1100,9 @@ static int mtk_afe_runtime_suspend(struct device *dev) { struct mtk_afe *afe = dev_get_drvdata(dev); + /* disable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0); + /* disable AFE clk */ regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_AFE, AUD_TCON0_PDN_AFE); @@ -1164,6 +1149,9 @@ static int mtk_afe_runtime_resume(struct device *dev) /* unmask all IRQs */ regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, 0xff, 0xff); + + /* enable AFE */ + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); return 0; err_bck0: -- 1.7.9.5

2 1

[alsa-devel] Applied "ASoC: rt5616: add devicetree document for rt5616" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: rt5616: add devicetree document for rt5616 has been applied to the asoc tree at git://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 27af4e488c6a2a8c4b6102e777c29fff467a3895 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Tue, 22 Dec 2015 13:45:03 +0800 Subject: [PATCH] ASoC: rt5616: add devicetree document for rt5616 Add the description for rt5616 codec. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5616.txt | 26 ++++++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt new file mode 100644 index 000000000000..efc48c65198d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5616.txt @@ -0,0 +1,26 @@ +RT5616 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5616". + +- reg : The I2C address of the device. + +Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +Example: + +codec: rt5616@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; +}; -- 2.6.2

1 0

[alsa-devel] [PATCH 1/2] ASoC: rt5616: add an of_match table
by Caesar Wang 23 Dec '15

23 Dec '15

Add a device tree match table. This serves to make the driver's support of device tree more explicit. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/rt5616.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..e01a32d 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1287,6 +1287,14 @@ static const struct i2c_device_id rt5616_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5616_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5616_of_match[] = { + { .compatible = "realtek,rt5616", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5616_of_match); +#endif + static int rt5616_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1359,6 +1367,7 @@ static void rt5616_i2c_shutdown(struct i2c_client *client) static struct i2c_driver rt5616_i2c_driver = { .driver = { .name = "rt5616", + .of_match_table = of_match_ptr(rt5616_of_match), }, .probe = rt5616_i2c_probe, .remove = rt5616_i2c_remove, -- 1.9.1

4 6

[alsa-devel] Applied "ASoC: da7218: Enable mic level detection reporting to user-space" to the asoc tree
by Mark Brown 23 Dec '15

23 Dec '15

The patch ASoC: da7218: Enable mic level detection reporting to user-space has been applied to the asoc tree at git://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 e05c25a1af29d65260ed1458f2cc4a959030ebd2 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Thu, 3 Dec 2015 17:10:07 +0000 Subject: [PATCH] ASoC: da7218: Enable mic level detection reporting to user-space This patch adds support to the codec driver to handle mic level detect related IRQs, and report these to user-space using a uevent variable. The uevent variable string "EVENT=MIC_LEVEL_DETECT" is sent to user-space, if the mic level detect feature is enabled, and the audio captured at the chosen mic(s) is above a certain threshold. User-space can then handle the event accordingly (e.g. process audio capture stream). This method was chosen over ALSA control notification for a couple of reasons: 1) There's no requirement here for a control to read state from. The event is the only thing that's required and of interest. 2) tinyalsa support for control notifications does not exist so on platforms using this over alsa-lib there is a need to add code to support this event handling. Another possible option would be to use the standard Jack reporting framework but this really does not fit for this kind of event. Finally, use of the input device framework is not being encouraged, due to difficulties in enabling apps to access input devices, so this has also been avoided. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7218.c | 19 ++++++++++++++----- 1 file changed, 14 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index eacde128c4d6..72686517ff54 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -2202,6 +2202,16 @@ int da7218_hpldet(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(da7218_hpldet); +static void da7218_micldet_irq(struct snd_soc_codec *codec) +{ + char *envp[] = { + "EVENT=MIC_LEVEL_DETECT", + NULL, + }; + + kobject_uevent_env(&codec->dev->kobj, KOBJ_CHANGE, envp); +} + static void da7218_hpldet_irq(struct snd_soc_codec *codec) { struct da7218_priv *da7218 = snd_soc_codec_get_drvdata(codec); @@ -2232,6 +2242,10 @@ static irqreturn_t da7218_irq_thread(int irq, void *data) if (!status) return IRQ_NONE; + /* Mic level detect */ + if (status & DA7218_LVL_DET_EVENT_MASK) + da7218_micldet_irq(codec); + /* HP detect */ if (status & DA7218_HPLDET_JACK_EVENT_MASK) da7218_hpldet_irq(codec); @@ -2936,11 +2950,6 @@ static int da7218_probe(struct snd_soc_codec *codec) } if (da7218->irq) { - /* Mask off mic level events, currently not handled */ - snd_soc_update_bits(codec, DA7218_EVENT_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK, - DA7218_LVL_DET_EVENT_MSK_MASK); - ret = devm_request_threaded_irq(codec->dev, da7218->irq, NULL, da7218_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, -- 2.6.2

1 0

[alsa-devel] [PATCH 0/12] ASoC: rsnd: add Multi channel support
by Kuninori Morimoto 23 Dec '15

23 Dec '15

Hi Mark These are Multi channel support for Renesas R-Car sound. These patches are tested on Salvator-X board, but it can't use Multi channel, because supported format is different between codec chip and R-Car. Thus, it was tested on board which doesn't mount codec chip, with oscilloscope. Kuninori Morimoto (12): 1) ASoC: rsnd: fixup SSIU control timing 2) ASoC: rsnd: tidyup return value of rsnd_get_adinr_bit() 3) ASoC: rsnd: tidyup debug print position on rsnd_dma_attach() 4) ASoC: rsnd: rsnd_dai_connect() returns error if it connect to existing mod 5) ASoC: rsnd: SSI/SSIU use rsnd_get_slot_extend() to check TDM 6) ASoC: rsnd: remove rsnd_get_slot_runtime() 7) ASoC: rsnd: tidyup rsnd_get_slot_xxx() naming 8) ASoC: rsnd: add rsnd_set_slot() / rsnd_get_slot_num() 9) ASoC: rsnd: add rsnd_parse_connect_common() and remove complex macro 10) ASoC: rsnd: add missing DT example for Simple Card 11) ASoC: rsnd: add missing DT example for Simple Card with TDM 12) ASoC: rsnd: add Multi channel support Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 77 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/core.c | 111 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++---------------------------------------- sound/soc/sh/rcar/dma.c | 10 ++++----- sound/soc/sh/rcar/dvc.c | 3 +-- sound/soc/sh/rcar/gen.c | 3 +++ sound/soc/sh/rcar/rsnd.h | 45 +++++++++++++++++++++++++++++++++------ sound/soc/sh/rcar/ssi.c | 118 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++---- sound/soc/sh/rcar/ssiu.c | 57 +++++++++++++++++++++++++++++++++++++++++--------- 8 files changed, 351 insertions(+), 73 deletions(-) Best regards --- Kuninori Morimoto

3 32

[alsa-devel] [PATCH] ASoC: ssm2518: Use a signed return type for ssm2518_lookup_mcs()
by SF Markus Elfring 23 Dec '15

23 Dec '15

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Sun, 20 Dec 2015 10:34:25 +0100 The return type "unsigned int" was used by the ssm2518_lookup_mcs() function even though it will eventually return a negative error code. Improve this implementation detail by deletion of the type modifier then. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/codecs/ssm2518.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 86b81a6..e2e0bfa 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -309,7 +309,7 @@ static const struct snd_pcm_hw_constraint_list ssm2518_constraints_12288000 = { .count = ARRAY_SIZE(ssm2518_rates_12288000), }; -static unsigned int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, +static int ssm2518_lookup_mcs(struct ssm2518 *ssm2518, unsigned int rate) { const unsigned int *sysclks = NULL; -- 2.6.3

2 1

[alsa-devel] [PATCH] ASoC: rt5616: rename some alsa control names
by Bard Liao 23 Dec '15

23 Dec '15

Rename some alsa control name as what they should be. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5616.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index f4005cb..4de8c86 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -323,9 +323,9 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN1 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5616_IN1_IN2, + SOC_SINGLE_TLV("IN2 Boost Volume", RT5616_IN1_IN2, RT5616_BST_SFT2, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5616_INL1_INR1_VOL, @@ -339,7 +339,7 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { 127, 0, adc_vol_tlv), /* ADC Boost Volume Control */ - SOC_DOUBLE_TLV("ADC Boost Gain", RT5616_ADC_BST_VOL, + SOC_DOUBLE_TLV("ADC Boost Volume", RT5616_ADC_BST_VOL, RT5616_ADC_L_BST_SFT, RT5616_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: atmel_wm8904: add snd_soc_pm_ops
by Songjun Wu 23 Dec '15

23 Dec '15

Sometimes the audio play can not be resumed after it is suspended. Add snd_soc_pm_ops to execute power management operations, then this issue is fixed. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 1933bcd..fdd28ed 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -183,6 +183,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .driver = { .name = "atmel-wm8904-audio", .of_match_table = of_match_ptr(atmel_asoc_wm8904_dt_ids), + .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, .remove = atmel_asoc_wm8904_remove, -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2] ASoC: fsl_asrc: add support for 8-30kHz output sample rate
by Zidan Wang 23 Dec '15

23 Dec '15

Add 8kHz, 11.025kHz, 16kHz, 22.05kHz output sample rate support. According referance menual, "Limited support for the case when output sampling rates is between 8kHz and 30kHz. The limitation is the supported ratio (Fsin/Fsout) range as between 1/24 to 8." Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. fix indentation issue. sound/soc/fsl/fsl_asrc.c | 39 +++++++++++++++++++++++---------------- 1 file changed, 23 insertions(+), 16 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index d964016..bd32c30 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -31,21 +31,21 @@ dev_dbg(&asrc_priv->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__) /* Sample rates are aligned with that defined in pcm.h file */ -static const u8 process_option[][8][2] = { - /* 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ - {{0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ - {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ - {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ - {{0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ - {{1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ - {{1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ - {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ +static const u8 process_option[][12][2] = { + /* 8kHz 11.025kHz 16kHz 22.05kHz 32kHz 44.1kHz 48kHz 64kHz 88.2kHz 96kHz 176kHz 192kHz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 5512Hz */ + {{0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 8kHz */ + {{0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 11025Hz */ + {{1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 16kHz */ + {{1, 2}, {1, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0}, {0, 0}, {0, 0},}, /* 22050Hz */ + {{1, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0}, {0, 0},}, /* 32kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 44.1kHz */ + {{2, 2}, {2, 2}, {2, 1}, {2, 1}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0}, {0, 0},}, /* 48kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 1}, {1, 2}, {0, 2}, {0, 2}, {0, 1}, {0, 1}, {0, 1}, {0, 1}, {0, 0},}, /* 64kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 88.2kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {1, 2}, {1, 2}, {1, 2}, {1, 1}, {1, 1}, {1, 1}, {1, 1}, {1, 1},}, /* 96kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 176kHz */ + {{2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 2}, {2, 1}, {2, 1}, {2, 1}, {2, 1}, {2, 1},}, /* 192kHz */ }; /* Corresponding to process_option */ @@ -55,7 +55,7 @@ static int supported_input_rate[] = { }; static int supported_asrc_rate[] = { - 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, }; /** @@ -287,6 +287,13 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if ((outrate > 8000 && outrate < 30000) && + (outrate/inrate > 24 || inrate/outrate > 8)) { + pair_err("exceed supported ratio range [1/24, 8] for \ + inrate/outrate: %d/%d\n", inrate, outrate); + return -EINVAL; + } + /* Validate input and output clock sources */ clk_index[IN] = clk_map[IN][config->inclk]; clk_index[OUT] = clk_map[OUT][config->outclk]; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl-asoc-card: use different route map for AC'97 mode
by Maciej S. Szmigiero 23 Dec '15

23 Dec '15

fsl_ssi uses different stream names ("AC97 Playback" / "AC97 Capture") in AC'97 mode so in this case fsl-asoc-card route map should also be using them. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl-asoc-card.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c63d89da51f1..562b3bd22d9a 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -107,6 +107,13 @@ static const struct snd_soc_dapm_route audio_map[] = { {"CPU-Capture", NULL, "Capture"}, }; +static const struct snd_soc_dapm_route audio_map_ac97[] = { + {"AC97 Playback", NULL, "ASRC-Playback"}, + {"Playback", NULL, "AC97 Playback"}, + {"ASRC-Capture", NULL, "AC97 Capture"}, + {"AC97 Capture", NULL, "Capture"}, +}; + /* Add all possible widgets into here without being redundant */ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_LINE("Line Out Jack", NULL), @@ -579,7 +586,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.dev = &pdev->dev; priv->card.name = priv->name; priv->card.dai_link = priv->dai_link; - priv->card.dapm_routes = audio_map; + priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ? + audio_map_ac97 : audio_map; priv->card.late_probe = fsl_asoc_card_late_probe; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map); priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;

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[alsa-devel] [PATCH 0/5] ASoC: Allow topology to create DAI links
by mengdong.lin＠linux.intel.com 23 Dec '15

23 Dec '15

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This series allows DAI links to be created from topology info. The components with topology info can come in two ways: - As a component for cpu dai or codec dai of a DAI link predefined by machine driver. - As a auxiliary components defined by the machine driver for a soc card. For this case, the machine driver even need not define any DAI links. The ASoC core will find and probe these components when instantiating the soc card. When probing the components, topology info will be loaded and DAI links can be created by the topology core. And then ASoC core will find and bind new DAI links after probing the components. We implement a DAI link list and define API for adding/removing DAI links from topology. Mengdong Lin (5): ASoC: Implement DAI links in a list & define API to add/remove a link ASoC: Define add/remove_dai_link ops for a soc card ASoC: soc_bind_dai_link() directly returns success for a bound DAI link ASoC: Bind new DAI links after probing components ASoC: The soc card can have auxiliary components include/sound/soc.h | 29 +++++++++- sound/soc/soc-core.c | 153 +++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 180 insertions(+), 2 deletions(-) -- 2.5.0

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