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

[alsa-devel] [PATCH] ASoC: arizona: Fix bclk for sample rates that are multiple of 4kHz
by Charles Keepax 24 Dec '15

24 Dec '15

From: Nikesh Oswal <Nikesh.Oswal(a)cirrus.com> For a sample rate of 12kHz the bclk was taken from the 44.1kHz table as we test for a multiple of 8kHz. This patch fixes this issue by testing for multiples of 4kHz instead. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 88e2c74..33143fe 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1659,7 +1659,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bool reconfig; unsigned int aif_tx_state, aif_rx_state; - if (params_rate(params) % 8000) + if (params_rate(params) % 4000) rates = &arizona_44k1_bclk_rates[0]; else rates = &arizona_48k_bclk_rates[0]; -- 2.1.4

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[alsa-devel] Applied "ASoC: da7219: Add regmap patch to support old silicon" to the asoc tree
by Mark Brown 24 Dec '15

24 Dec '15

The patch ASoC: da7219: Add regmap patch to support old silicon 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 abd7c894fc41a9a674354e10ed6c55413e1db077 Mon Sep 17 00:00:00 2001 From: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Date: Wed, 23 Dec 2015 13:50:04 +0000 Subject: [PATCH] ASoC: da7219: Add regmap patch to support old silicon Initial silicon did not have master bias enabled by default, unlike later HW, so use regmap patch to align with newer defaults. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/da7219.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c6d3b32..9c7e8ec 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1592,9 +1592,14 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec) } } +static struct reg_sequence da7219_rev_aa_patch[] = { + { DA7219_REFERENCES, 0x08 }, +}; + static int da7219_probe(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + unsigned int rev; int ret; mutex_init(&da7219->lock); @@ -1604,6 +1609,26 @@ static int da7219_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regmap_read(da7219->regmap, DA7219_CHIP_REVISION, &rev); + if (ret) { + dev_err(codec->dev, "Failed to read chip revision: %d\n", ret); + goto err_disable_reg; + } + + switch (rev & DA7219_CHIP_MINOR_MASK) { + case 0: + ret = regmap_register_patch(da7219->regmap, da7219_rev_aa_patch, + ARRAY_SIZE(da7219_rev_aa_patch)); + if (ret) { + dev_err(codec->dev, "Failed to register AA patch: %d\n", + ret); + goto err_disable_reg; + } + break; + default: + break; + } + /* Handle DT/Platform data */ if (codec->dev->of_node) da7219->pdata = da7219_of_to_pdata(codec); @@ -1774,7 +1799,6 @@ static struct reg_default da7219_reg_defaults[] = { { DA7219_MIXOUT_R_CTRL, 0x10 }, { DA7219_CHIP_ID1, 0x23 }, { DA7219_CHIP_ID2, 0x93 }, - { DA7219_CHIP_REVISION, 0x00 }, { DA7219_IO_CTRL, 0x00 }, { DA7219_GAIN_RAMP_CTRL, 0x00 }, { DA7219_PC_COUNT, 0x02 }, -- 2.6.4

1 0

[alsa-devel] [PATCH v2] ASoC: fsl_asrc: old version asrc only support even numbered channels
by Zidan Wang 24 Dec '15

24 Dec '15

The latest asrc's channel bits is 4 bit, the old asrc's channel bits is 3. For the old version asrc, channel should be a even number. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. add an extra else if for the new condition. sound/soc/fsl/fsl_asrc.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index be78dc9..16ff156 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -116,6 +116,10 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) } else if (asrc_priv->channel_avail < channels) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; + } else if (asrc_priv->channel_bits < 4 && channels % 2 != 0) { + dev_err(dev, "old version asrc can't support an odd \ + channel number\n"); + ret = -EINVAL; } else { asrc_priv->channel_avail -= channels; asrc_priv->pair[index] = pair; -- 1.9.1

1 0

[alsa-devel] [PATCH 0/4] ALSA: dice: improve card registration processing
by Takashi Sakamoto 23 Dec '15

23 Dec '15

Hi, This patchset is to update a part of my previous RFCv2 (patch 01, 03 and 04. 02 is dropped according to Clemens' comment). [alsa-devel] [RFC][PATCH 00/10 v2] ALSA: dice: stabiliza packet streaming http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/101897.h… The main purpose of this patchset is to postpone card registration after successive bus resets, to ensure probe processing and related transactions. This change has effects for ALSA dice driver to get proper information about hardware, especially for Dice II based models. Changes from RFC v2: * Patch 03 is newly added to drop a workaround to ensure transactions under a situation of probe processing with successive bus reset. * Comment updates. Takashi Sakamoto (4): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-transaction.c | 122 +++++++++++++++-------- sound/firewire/dice/dice.c | 177 ++++++++++++++++----------------- sound/firewire/dice/dice.h | 3 + 3 files changed, 166 insertions(+), 136 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 1/1] ASoC: da7219: Add regmap patch to support old silicon
by Adam Thomson 23 Dec '15

23 Dec '15

Initial silicon did not have master bias enabled by default, unlike later HW, so use regmap patch to align with newer defaults. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- This patch is based on the following commit: 'ASoC: da7219: Remove support for 32KHz PLL mode' (Commit 501f72e9c5205b9d70d5d61e9b186ae7ba873f73, kernel/git/broonie/sound.git) sound/soc/codecs/da7219.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 82a7236..341a191 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1592,9 +1592,14 @@ static void da7219_handle_pdata(struct snd_soc_codec *codec) } } +static struct reg_sequence da7219_rev_aa_patch[] = { + { DA7219_REFERENCES, 0x08 }, +}; + static int da7219_probe(struct snd_soc_codec *codec) { struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); + unsigned int rev; int ret; mutex_init(&da7219->lock); @@ -1604,6 +1609,26 @@ static int da7219_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regmap_read(da7219->regmap, DA7219_CHIP_REVISION, &rev); + if (ret) { + dev_err(codec->dev, "Failed to read chip revision: %d\n", ret); + goto err_disable_reg; + } + + switch (rev & DA7219_CHIP_MINOR_MASK) { + case 0: + ret = regmap_register_patch(da7219->regmap, da7219_rev_aa_patch, + ARRAY_SIZE(da7219_rev_aa_patch)); + if (ret) { + dev_err(codec->dev, "Failed to register AA patch: %d\n", + ret); + goto err_disable_reg; + } + break; + default: + break; + } + /* Handle DT/Platform data */ if (codec->dev->of_node) da7219->pdata = da7219_of_to_pdata(codec); @@ -1774,7 +1799,6 @@ static struct reg_default da7219_reg_defaults[] = { { DA7219_MIXOUT_R_CTRL, 0x10 }, { DA7219_CHIP_ID1, 0x23 }, { DA7219_CHIP_ID2, 0x93 }, - { DA7219_CHIP_REVISION, 0x00 }, { DA7219_IO_CTRL, 0x00 }, { DA7219_GAIN_RAMP_CTRL, 0x00 }, { DA7219_PC_COUNT, 0x02 }, -- 1.9.3

1 0

[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

4 17

[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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