December 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4] ASoC: wm8960: Let wm8960 codec driver manage its own MCLK
by Zidan Wang 30 Dec '14

30 Dec '14

When we want to use wm8960 codec, we should enable its MCLK in machine driver. It's reasonable for wm8960 codec driver to manage its own MCLK to save power. Enable runtime power management, and auto enable/disable MCLK in pm_runtime resume and suspend. When wm8960 codec is being used, it will triger resume() to enable MCLK. When codec is not being used, it will triger suspend() to disable MCLK. Signed-off-by: Zidan Wang <b50113(a)freescale.com> --- sound/soc/codecs/wm8960.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 031a1ae..aff24a7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,8 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/slab.h> #include <sound/core.h> @@ -117,6 +119,7 @@ static bool wm8960_volatile(struct device *dev, unsigned int reg) } struct wm8960_priv { + struct clk *mclk; struct regmap *regmap; int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); @@ -1002,6 +1005,13 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, if (wm8960 == NULL) return -ENOMEM; + wm8960->mclk = devm_clk_get(&i2c->dev, "codec_mclk"); + + if (IS_ERR(wm8960->mclk)) { + if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap); if (IS_ERR(wm8960->regmap)) return PTR_ERR(wm8960->regmap); @@ -1041,6 +1051,9 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8960); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8960, &wm8960_dai, 1); @@ -1053,6 +1066,38 @@ static int wm8960_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8960_runtime_resume(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int wm8960_runtime_suspend(struct device *dev) +{ + struct wm8960_priv *wm8960 = dev_get_drvdata(dev); + + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8960_pm = { + SET_RUNTIME_PM_OPS(wm8960_runtime_suspend, wm8960_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -1070,6 +1115,7 @@ static struct i2c_driver wm8960_i2c_driver = { .name = "wm8960", .owner = THIS_MODULE, .of_match_table = wm8960_of_match, + .pm = &wm8960_pm, }, .probe = wm8960_i2c_probe, .remove = wm8960_i2c_remove, -- 1.9.1

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[alsa-devel] [PATCH v5 0/6] ASoC: dwc: Add device tree support to designware I2S
by Andrew Jackson 30 Dec '14

30 Dec '14

From: Andrew Jackson <Andrew.Jackson(a)arm.com> This patch set extends the DesignWare I2S driver to provide device tree support and fixes a couple of small faults. Changes v4->v5 + Remove all clk_put calls [Andrew Jackson] + Call clk_disable_unprepare [Andrew Jackson] Changes v3->v4 + Drop applied patches + Use combined function to prepare clock [Mark Brown] + Use managed clock resources to avoid clk_put [Mark Brown] + Read configuration prameters from hardware for both platform data and device tree [Mark Brown] + Re-order patch sequence [Mark Brown] + Change union name to avoid future collisions [Mark Brown] + Check return code from clk_set_rate [Mark Brown] + Check parametsrs read from hardware agaist array limits and add further comments [Mark Brown] + Add of_match_ptr [Mark Brown] Changes v2->v3 + Drop applied patch + Flush FIFOs in prepare rather than hw_params [Lars-Peter Clausen] Changes v1->v2 + Drop negative use count patch [Mark Brown] + Remove unnecessary debug print messages [Lars-Peter Clausen] + Rewrite iteration as for loop rather than do...while [Mark Brown] + Reorder patches to send fixes first [Mark Brown] + Simplify device tree code [Mark Brown] + Split device tree patch in two [Mark Brown] + Expand explanatory comment on channel configuration [Rajeev Kumar] Arnd: I've not forgotten about updating the spear entries and will submit as a separate patch. Andrew Jackson (6): ASoC: dwc: Switch to managed clock resource ASoC: dwc: Prepare clock before use ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Register components with managed interface ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S .../devicetree/bindings/sound/designware-i2s.txt | 31 +++ sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 283 ++++++++++++++++---- 3 files changed, 263 insertions(+), 52 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt

3 11

[alsa-devel] [PATCH] ASoC: mxs-saif: Do not store the irq number in the private structure
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The irq number is only used inside the probe function, so there is really no need to store it in the private structure. Use a local 'irq' variable to hold the the irq number instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 10 +++++----- sound/soc/mxs/mxs-saif.h | 1 - 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index d986508..c866ade 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -710,7 +710,7 @@ static int mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores; struct mxs_saif *saif; - int ret = 0; + int irq, ret = 0; struct device_node *master; if (!np) @@ -763,16 +763,16 @@ static int mxs_saif_probe(struct platform_device *pdev) if (IS_ERR(saif->base)) return PTR_ERR(saif->base); - saif->irq = platform_get_irq(pdev, 0); - if (saif->irq < 0) { - ret = saif->irq; + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + ret = irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); return ret; } saif->dev = &pdev->dev; - ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + ret = devm_request_irq(&pdev->dev, irq, mxs_saif_irq, 0, dev_name(&pdev->dev), saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index fbaf7ba..9a4c0b2 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -116,7 +116,6 @@ struct mxs_saif { unsigned int mclk; unsigned int mclk_in_use; void __iomem *base; - int irq; unsigned int id; unsigned int master_id; unsigned int cur_rate; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Support for combined tx/rx interrupt line
by Peter Ujfalusi 30 Dec '14

30 Dec '14

Some SoC, like da850/OMAP-L138 uses one common interrupt request for TX/RX events. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 30b94d4f9c5d..95eef582331b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -364,6 +364,20 @@ static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data) return IRQ_RETVAL(handled_mask); } +static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data) +{ + struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data; + irqreturn_t ret = IRQ_NONE; + + if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK]) + ret = davinci_mcasp_tx_irq_handler(irq, data); + + if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE]) + ret |= davinci_mcasp_rx_irq_handler(irq, data); + + return ret; +} + static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -1441,6 +1455,22 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->dev = &pdev->dev; + irq = platform_get_irq_byname(pdev, "common"); + if (irq >= 0) { + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + dev_name(&pdev->dev)); + ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, + davinci_mcasp_common_irq_handler, + IRQF_ONESHOT, irq_name, mcasp); + if (ret) { + dev_err(&pdev->dev, "common IRQ request failed\n"); + goto err; + } + + mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN; + mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN; + } + irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", -- 2.2.1

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[alsa-devel] [RFC] ASoC: wm8962: Allow codec to work when CONFIG_PM=n
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Currently the codec does not work in the CONFIG_PM=n case because the regulators and clock are kept disabled. Allow the codec to be functional by calling wm8962_runtime_resume() when CONFIG_PM=n. Tested on a imx6sl-evk in the CONFIG_PM=y and CONFIG_PM=n cases. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Probably there is a better solution, hence I am sending it as RFC. Thanks sound/soc/codecs/wm8962.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d32d554..0b041c9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3546,6 +3546,9 @@ static int wm8962_set_pdata_from_of(struct i2c_client *i2c, return 0; } +static int wm8962_runtime_resume(struct device *dev); +static int wm8962_runtime_suspend(struct device *dev); + static int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -3771,6 +3774,9 @@ static int wm8962_i2c_probe(struct i2c_client *i2c, /* The drivers should power up as needed */ regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_resume(&i2c->dev); + return 0; err_enable: @@ -3781,11 +3787,12 @@ err: static int wm8962_i2c_remove(struct i2c_client *client) { + if (!IS_ENABLED(CONFIG_PM)) + wm8962_runtime_suspend(&client->dev); snd_soc_unregister_codec(&client->dev); return 0; } -#ifdef CONFIG_PM static int wm8962_runtime_resume(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); @@ -3860,7 +3867,6 @@ static int wm8962_runtime_suspend(struct device *dev) return 0; } -#endif static struct dev_pm_ops wm8962_pm = { SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) -- 1.9.1

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[alsa-devel] [PATCH 1/9] ALSA: pcm_params: Remove unused add/sub functions
by Lars-Peter Clausen 30 Dec '14

30 Dec '14

Those two functions are not used anywhere and also their name is a bit to generic to be in a global header, so remove them. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/pcm_params.h | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 6b1c78f..15760f5 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -325,20 +325,6 @@ static inline int snd_interval_eq(const struct snd_interval *i1, const struct sn i1->max == i2->max && i1->openmax == i2->openmax; } -static inline unsigned int add(unsigned int a, unsigned int b) -{ - if (a >= UINT_MAX - b) - return UINT_MAX; - return a + b; -} - -static inline unsigned int sub(unsigned int a, unsigned int b) -{ - if (a > b) - return a - b; - return 0; -} - #define params_access(p) ((__force snd_pcm_access_t)\ snd_mask_min(hw_param_mask_c((p), SNDRV_PCM_HW_PARAM_ACCESS))) #define params_format(p) ((__force snd_pcm_format_t)\ -- 1.8.0

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[alsa-devel] [GIT PULL] ASoC updates for v3.20
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-rc2 for you to fetch changes up to 1b9f1ae8907cb13d3d2538d98385b2744c9490d5: Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next (2014-12-24 12:56:48 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.20 Nothing too exciting here yet, a small optimization for DAPM from Lars-Peter and a few small bits and pieces for drivers but nothing that really stands out. ---------------------------------------------------------------- Andrew Jackson (4): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Bard Liao (3): ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (4): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS ASoC: pcm: Fix vague codec and cpu DAI prepare error messages Jianqun Xu (3): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 ASoC: rockchip: i2s: set TDL and RDL to 16 samples Jie Yang (2): ASoC: Intel: correct the fixed free block allocation ASoC: Intel: Store the entry_point read from FW file Lars-Peter Clausen (4): ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget ASoC: wm8750: Cleanup manual bias level transitions ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected ASoC: dapm: Simplify fully route card handling Mark Brown (9): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/intel' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/rockchip', 'asoc/topic/rt5670' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/samsung' and 'asoc/topic/wm8750' into asoc-next Oder Chiou (3): ASoC: rt5677: Revise the filter powers ASoC: rt5677: Add the ASRC support ASoC: rt5677: Adjust the routing of "PLL1" Peter Rosin (1): ASoC: pcm512x: Also support PCM514x devices Takashi Iwai (1): ASoC: intel: Remove superfluous backslash in Kconfig Wei Yongjun (1): ASoC: Intel: fix sparse non static symbol warnings Wolfram Sang (4): ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers kbuild test robot (2): ASoC: Intel: fix platform_no_drv_owner.cocci warnings ASoC: Intel: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/pcm512x.txt | 3 +- include/sound/soc-dapm.h | 1 - sound/soc/codecs/pcm512x-i2c.c | 4 + sound/soc/codecs/pcm512x-spi.c | 4 + sound/soc/codecs/rt5670.c | 38 ++++ sound/soc/codecs/rt5677.c | 204 ++++++++++++++++++--- sound/soc/codecs/wm8750.c | 2 +- sound/soc/dwc/designware_i2s.c | 135 +++++++------- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 3 +- sound/soc/intel/cht_bsw_rt5672.c | 1 - sound/soc/intel/sst-firmware.c | 7 +- sound/soc/intel/sst/sst_acpi.c | 7 +- sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/pxa/spitz.c | 1 - sound/soc/rockchip/rockchip_i2s.c | 8 +- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/samsung/arndale_rt5631.c | 1 - sound/soc/soc-core.c | 17 +- sound/soc/soc-dapm.c | 105 ++--------- sound/soc/soc-pcm.c | 7 +- 21 files changed, 349 insertions(+), 210 deletions(-)

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[alsa-devel] ASoC fixes for v3.19
by Mark Brown 30 Dec '14

30 Dec '14

The following changes since commit 97bf6af1f928216fd6c5a66e8a57bfa95a659672: Linux 3.19-rc1 (2014-12-20 17:08:50 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v3.19-rc2 for you to fetch changes up to f2ec1d3b25776051a9205289dab12e9401d33f8a: Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus (2014-12-24 12:56:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 A few fixes for v3.19, a few driver specifics and one core fix which fixes a boot crash on OMAP if deferred probing kicks in due to attempting to modify static data. ---------------------------------------------------------------- Andrew Jackson (2): ASoC: dwc: Ensure FIFOs are flushed to prevent channel swap ASoC: dwc: Iterate over all channels Fang, Yang A (1): ASoC: rt5677: fixed rt5677_dsp_vad_put rt5677_dsp_vad_get panic Jarkko Nikula (3): ASoC: Intel: Add I2C dependency to two new machines ASoC: Intel: Fix BYTCR firmware name ASoC: Intel: Fix BYTCR machine driver MODULE_ALIAS Jianqun Xu (2): ASoC: rockchip: i2s: fix error defination of transmit data level ASoC: rockchip: i2s: fix maxburst of dma data to 4 Jie Yang (1): ASoC: Intel: correct the fixed free block allocation Mark Brown (3): ASoC: dapm: Remove snd_soc_of_parse_audio_routing() due to deferred probe Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/dwc', 'asoc/fix/intel', 'asoc/fix/rockchip' and 'asoc/fix/rt5677' into asoc-linus sound/soc/codecs/rt5677.c | 9 ++++--- sound/soc/dwc/designware_i2s.c | 49 +++++++++++++++++++++++-------------- sound/soc/intel/Kconfig | 4 +-- sound/soc/intel/bytcr_dpcm_rt5640.c | 2 +- sound/soc/intel/sst-firmware.c | 6 ++++- sound/soc/intel/sst/sst_acpi.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 4 +-- sound/soc/rockchip/rockchip_i2s.h | 2 +- sound/soc/soc-core.c | 14 ++++------- 9 files changed, 52 insertions(+), 40 deletions(-)

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[alsa-devel] [PATCH 0/2 v7] Digidesign Mbox 1 enhancements
by Damien Zammit 30 Dec '14

30 Dec '14

Revision 7. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 41 +++++++++++---- 3 files changed, 170 insertions(+), 9 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 1/2] [ALSA] add SNDRV_PCM_TRIGGER_DRAIN trigger
by libin.yang＠intel.com 30 Dec '14

30 Dec '14

From: Libin Yang <libin.yang(a)intel.com> Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/pcm.h | 1 + sound/core/pcm_native.c | 5 +++++ 2 files changed, 6 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 8bb00a2..bb77aa3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -109,6 +109,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_TRIGGER_PAUSE_RELEASE 4 #define SNDRV_PCM_TRIGGER_SUSPEND 5 #define SNDRV_PCM_TRIGGER_RESUME 6 +#define SNDRV_PCM_TRIGGER_DRAIN 7 #define SNDRV_PCM_POS_XRUN ((snd_pcm_uframes_t)-1) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 166d59c..15fe682 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1517,6 +1517,11 @@ static int snd_pcm_do_drain_init(struct snd_pcm_substream *substream, int state) { struct snd_pcm_runtime *runtime = substream->runtime; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + if (substream->runtime->trigger_master == substream) + substream->ops->trigger(substream, + SNDRV_PCM_TRIGGER_DRAIN); + switch (runtime->status->state) { case SNDRV_PCM_STATE_PREPARED: /* start playback stream if possible */ -- 1.9.1

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[alsa-devel] [PATCH] mfd: wm8994: set mfd id-base for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. If not, duplicate filename error was reported. - sysfs: cannot create duplicate filename '/devices/.../wm8994-ldo' Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..47cd3cc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,12 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, + .id = 0, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, + .id = 1, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +344,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, 0, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH] mfd: wm8994: use PLATFORM_DEVID_AUTO for regulator devs creation to avoid conflicts
by Inha Song 30 Dec '14

30 Dec '14

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. but, if we use PLATFORM_DEVID_AUTO flag, cell ids are automatically allocated and managed without cell id setting. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..0505e45 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,10 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +342,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, PLATFORM_DEVID_AUTO, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH] ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers
by Nicolin Chen 30 Dec '14

30 Dec '14

The xDC field should have 5 bit width according to Reference Manual. Thus this patch fixes it. Signed-off-by: Aurelien BOUIN <a_bouin(a)yahoo.fr> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- The patch was originally submitted by Aurelien BOUIN while in an informal way. And he hasn't finished any re-submitting during the past two weeks. Thus I create a new patch with his signed-off included so as to fix the probelm as soon as possible, not sure whether it's decent or not though. --Nicolin sound/soc/fsl/fsl_esai.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 91a550f..5e793bb 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -302,7 +302,7 @@ #define ESAI_xCCR_xFP_MASK (((1 << ESAI_xCCR_xFP_WIDTH) - 1) << ESAI_xCCR_xFP_SHIFT) #define ESAI_xCCR_xFP(v) ((((v) - 1) << ESAI_xCCR_xFP_SHIFT) & ESAI_xCCR_xFP_MASK) #define ESAI_xCCR_xDC_SHIFT 9 -#define ESAI_xCCR_xDC_WIDTH 4 +#define ESAI_xCCR_xDC_WIDTH 5 #define ESAI_xCCR_xDC_MASK (((1 << ESAI_xCCR_xDC_WIDTH) - 1) << ESAI_xCCR_xDC_SHIFT) #define ESAI_xCCR_xDC(v) ((((v) - 1) << ESAI_xCCR_xDC_SHIFT) & ESAI_xCCR_xDC_MASK) #define ESAI_xCCR_xPSR_SHIFT 8 -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: fsl_spdif: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_spdif_priv private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 15 +++------------ 1 file changed, 3 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index af042942..73da1f0 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -90,7 +90,6 @@ struct spdif_mixer_control { * @sysclk: system clock for rx clock rate measurement * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel - * @name: driver name */ struct fsl_spdif_priv { struct spdif_mixer_control fsl_spdif_control; @@ -109,12 +108,8 @@ struct fsl_spdif_priv { struct clk *sysclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; - - /* The name space will be allocated dynamically */ - char name[0]; }; - /* DPLL locked and lock loss interrupt handler */ static void spdif_irq_dpll_lock(struct fsl_spdif_priv *spdif_priv) { @@ -1169,19 +1164,15 @@ static int fsl_spdif_probe(struct platform_device *pdev) if (!np) return -ENODEV; - spdif_priv = devm_kzalloc(&pdev->dev, - sizeof(struct fsl_spdif_priv) + strlen(np->name) + 1, - GFP_KERNEL); + spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) return -ENOMEM; - strcpy(spdif_priv->name, np->name); - spdif_priv->pdev = pdev; /* Initialize this copy of the CPU DAI driver structure */ memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); - spdif_priv->cpu_dai_drv.name = spdif_priv->name; + spdif_priv->cpu_dai_drv.name = dev_name(&pdev->dev); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -1203,7 +1194,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, spdif_isr, 0, - spdif_priv->name, spdif_priv); + dev_name(&pdev->dev), spdif_priv); if (ret) { dev_err(&pdev->dev, "could not claim irq %u\n", irq); return ret; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl_esai: Use dev_name() for registering the irq
by Fabio Estevam 30 Dec '14

30 Dec '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> The 'name' array is currently stored inside the fsl_esai private structure only for registering the interrupt name. This can be simplified by registering it with dev_name() instead. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_esai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 1c08ab1..dbe9204 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -44,7 +44,6 @@ * @sck_div: if using PSR/PM dividers for SCKx clock * @slave_mode: if fully using DAI slave mode * @synchronous: if using tx/rx synchronous mode - * @name: driver name */ struct fsl_esai { struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -63,7 +62,6 @@ struct fsl_esai { bool sck_div[2]; bool slave_mode; bool synchronous; - char name[32]; }; static irqreturn_t esai_isr(int irq, void *devid) @@ -739,7 +737,6 @@ static int fsl_esai_probe(struct platform_device *pdev) return -ENOMEM; esai_priv->pdev = pdev; - strncpy(esai_priv->name, np->name, sizeof(esai_priv->name) - 1); /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -779,7 +776,7 @@ static int fsl_esai_probe(struct platform_device *pdev) } ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, - esai_priv->name, esai_priv); + dev_name(&pdev->dev), esai_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u\n", irq); return ret; -- 1.9.1

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[alsa-devel] [PATCH v5 0/4] ASoC: simple-card: Add multi-CODEC support
by Jean-Francois Moine 29 Dec '14

29 Dec '14

This patch set adds multi-CODEC support to the simple card. v5: accept only "sound-dai" as the dai link property (Arnd Bergmann) v4: (Mark Brown) - document the new core function - calculation of the number of codecs moved to the core function - split the patch about simple-card.c - less code changes - remove the cast of pointers - less changes in the DT binding example 2 v3: rebase on broonie git 2014/11/03 v2: accept list of CODECs in 'sound-dais' (Benoit Cousson) Jean-Francois Moine (4): ASoC: core: add multi-codec support in DT ASoC: simple-card: Remove useless function argument ASoC: simple-card: add multi-CODECs in DT ASoC: simple-card: Remove useless check .../devicetree/bindings/sound/simple-card.txt | 18 +--- include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 87 ++++++++-------- sound/soc/soc-core.c | 110 ++++++++++++++++++--- 4 files changed, 149 insertions(+), 69 deletions(-) -- 2.1.3

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[alsa-devel] [Patch V3 00/10] ASoC: QCOM: Add support for ipq806x SOC
by Kenneth Westfield 29 Dec '14

29 Dec '14

From: Kenneth Westfield <kwestfie(a)codeaurora.org> This set of patches adds support for audio on the Qualcomm Technologies ipq806x SOC. The ipq806x SOC has audio-related hardware blocks in its low-power audio subsystem (or LPASS). One of the relevant blocks in the LPASS is its low-power audio interface (or LPAIF). This encapsulates the MI2S port, which is what these drivers are configured to use. The I2S pins are connected to an external DAC/amp chip. In addition, a single GPIO is connected to the same DAC/amp, which gives the SOC enable/disable control. The specific drivers added are: - a Codec DAI driver that controls the SOC external pins - a CPU DAI driver for controlling the LPASS-LPAIF block - a PCM MI2S platform driver These drivers, together, are used by simple-audio-card to complete the audio implementation. Corresponding additions to the device tree for the ipq806x and its documentation has also been added. Also, as this is a new directory, the MAINTAINERS file has been updated as well. == Updates from my previous post: [Patch v2 00/11] ASoC: QCOM: Add support for ipq806x SOC http://thread.gmane.org/gmane.linux.ports.arm.msm/10989 - Removed the PCM platform driver from the DTS platform and tied it to the CPU DAI driver. - Changed I2S pinctrl to use generic naming convention and moved control to CPU DAI driver. It should be controlled now by soc-core's pinctrl_pm_* functionality. - Added stub DAPM support in codec driver. As the DAC GPIO needs to be enabled last when starting playback, and disabled first when stopping playback, it seems as though the trigger function may be the place for this. Suggestions are welcome for a better place to put this. - Removed machine driver and tied DAI drivers to simple-audio-card. - Packaged the build files and Maxim codec files together in one change. - Removed QCOM as vendor from Maxim code and documentation. - Separated the SOC and board definitions into the correct DTS files. - Update device tree documentation to reflect changes. - General code cleanup. == Additional patch series upon which this patch series has device-tree dependencies (lcc) [PATCH v2 0/8] qcom audio clock control drivers http://thread.gmane.org/gmane.linux.ports.arm.msm/10793 Kenneth Westfield (10): MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Document MAX98357A bindings ASoC: qcom: Document LPASS CPU bindings ASoC: codec: Add MAX98357A codec driver ASoC: ipq806x: add LPASS header files ASoC: ipq806x: Add LPASS CPU DAI driver ASoC: ipq806x: Add I2S PCM platform driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers ARM: dts: Model IPQ LPASS audio hardware .../devicetree/bindings/sound/max98357a.txt | 14 + .../bindings/sound/qcom,lpass-cpu-mi2s.txt | 42 ++ MAINTAINERS | 7 + arch/arm/boot/dts/qcom-ipq8064.dtsi | 23 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98357a.c | 136 ++++++ sound/soc/qcom/Kconfig | 27 ++ sound/soc/qcom/Makefile | 6 + sound/soc/qcom/lpass-cpu-mi2s.c | 378 ++++++++++++++++ sound/soc/qcom/lpass-lpaif-reg.h | 155 +++++++ sound/soc/qcom/lpass-mi2s.h | 34 ++ sound/soc/qcom/lpass-pcm-mi2s.c | 486 +++++++++++++++++++++ 15 files changed, 1316 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98357a.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu-mi2s.txt create mode 100644 sound/soc/codecs/max98357a.c create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu-mi2s.c create mode 100644 sound/soc/qcom/lpass-lpaif-reg.h create mode 100644 sound/soc/qcom/lpass-mi2s.h create mode 100644 sound/soc/qcom/lpass-pcm-mi2s.c -- 1.8.2.1

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[alsa-devel] minos-one live DVD with Advanced Gtk+ Sequencer
by Joël Krähemann 28 Dec '14

28 Dec '14

Hi out there, I just built a http://linuxfromscratch.org based live DVD called minos-one http://sourceforge.net/projects/ags/files/minos-one.iso Cheers Joël

1 0

[alsa-devel] [PATCH v3] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 27 Dec '14

27 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Changes in v3: - added my SoB sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..e458957c5f52 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 1

[alsa-devel] [PATCH v2] ALSA: atmel: fix building the ac97 driver for at91-multiplatform
by Alexandre Belloni 26 Dec '14

26 Dec '14

From: Arnd Bergmann <arnd(a)arndb.de> at91 will no longer export the mach/cpu.h and mach/hardware.h header files in the future, which would break building the atmel ac97c driver. Since the cpu_is_* check is only used to find out whether we are running on avr32 or arm/at91, we can hardcode that check in the ARM case. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: http://www.spinics.net/lists/arm-kernel/msg382068.html --- sound/atmel/ac97c.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b59427d5a697..83975f8d5947 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -34,10 +34,10 @@ #include <linux/platform_data/dma-dw.h> #include <linux/dma/dw.h> +#ifdef CONFIG_AVR32 #include <mach/cpu.h> - -#ifdef CONFIG_ARCH_AT91 -#include <mach/hardware.h> +#else +#define cpu_is_at32ap7000() 0 #endif #include "ac97c.h" -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: wm5102: Use put_unaligned_be16
by Vaishali Thakkar 26 Dec '14

26 Dec '14

This patch introduces the use of function put_unaligned_le16. This is done using Coccinelle and semantic patch used is as follows: @a@ typedef u16, __be16, uint16_t; {u16,__be16,uint16_t} e16; identifier tmp; expression ptr; expression y,e; type T; @@ - tmp = cpu_to_be16(y); <+... when != tmp ( - memcpy(ptr, (T)&tmp, \(2\|sizeof(u16)\|sizeof(__be16)\|sizeof(uint16_t)\|sizeof(e16)\)); + put_unaligned_be16(y,ptr); | - memcpy(ptr, (T)&tmp, ...); + put_unaligned_be16(y,ptr); ) ...+> ? tmp = e @@ type T; identifier a.tmp; @@ - T tmp; ...when != tmp Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/codecs/wm5102.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f602349..fec21f9 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -28,6 +28,7 @@ #include <linux/mfd/arizona/core.h> #include <linux/mfd/arizona/registers.h> +#include <asm/unaligned.h> #include "arizona.h" #include "wm5102.h" @@ -617,11 +618,11 @@ static int wm5102_out_comp_coeff_get(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - uint16_t data; mutex_lock(&codec->mutex); - data = cpu_to_be16(arizona->dac_comp_coeff); - memcpy(ucontrol->value.bytes.data, &data, sizeof(data)); + + put_unaligned_be16(arizona->dac_comp_coeff, + ucontrol->value.bytes.data); mutex_unlock(&codec->mutex); return 0; -- 1.9.1

1 0

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: ad198x_ch_mode_get() ad198x_ch_mode_info() ad198x_ch_mode_info() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 33 --------------------------------- 1 file changed, 33 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..0e76dcc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -697,39 +697,6 @@ static int patch_ad1981(struct hda_codec *codec) * E/F quad mic array */ -#ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->multiout.max_channels); - if (err >= 0 && spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - return err; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ - static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: hda: patch_analog.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function ad198x_ch_mode_get() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/hda/patch_analog.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 06275f8..9d5257b 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -707,15 +707,6 @@ static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, spec->num_channel_mode); } -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, spec->multiout.max_channels); -} - static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { -- 1.7.10.4

3 3

[alsa-devel] [PATCH] sound: isa: gus: gus_instr.c: Remove unused function
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Remove the function snd_gus_simple_remove_sample() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/isa/gus/gus_instr.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/isa/gus/gus_instr.c b/sound/isa/gus/gus_instr.c index 4dc9caf..2d21408 100644 --- a/sound/isa/gus/gus_instr.c +++ b/sound/isa/gus/gus_instr.c @@ -162,11 +162,3 @@ int snd_gus_simple_get_sample(void *private_data, struct simple_instrument *inst return snd_gus_dram_read(gus, data, instr->address.memory, instr->size, 0); } - -int snd_gus_simple_remove_sample(void *private_data, struct simple_instrument *instr, - int atomic) -{ - struct snd_gus_card *gus = private_data; - - return snd_gf1_mem_free(&gus->gf1.mem_alloc, instr->address.memory); -} -- 1.7.10.4

2 1

[alsa-devel] [PATCH] sound: pci: ice1712: wm8776.c: Remove some unused functions
by Rickard Strandqvist 25 Dec '14

25 Dec '14

Removes some functions that are not used anywhere: snd_wm8776_set_master_mode() snd_wm8776_set_adc_if() snd_wm8776_set_dac_if() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/ice1712/wm8776.c | 15 --------------- sound/pci/ice1712/wm8776.h | 3 --- 2 files changed, 18 deletions(-) diff --git a/sound/pci/ice1712/wm8776.c b/sound/pci/ice1712/wm8776.c index e66c0da..ebd2fe4 100644 --- a/sound/pci/ice1712/wm8776.c +++ b/sound/pci/ice1712/wm8776.c @@ -452,21 +452,6 @@ void snd_wm8776_resume(struct snd_wm8776 *wm) snd_wm8776_write(wm, i, wm->regs[i]); } -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac) -{ - snd_wm8776_write(wm, WM8776_REG_DACIFCTRL, dac); -} - -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc) -{ - snd_wm8776_write(wm, WM8776_REG_ADCIFCTRL, adc); -} - -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode) -{ - snd_wm8776_write(wm, WM8776_REG_MSTRCTRL, mode); -} - void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power) { snd_wm8776_write(wm, WM8776_REG_PWRDOWN, power); diff --git a/sound/pci/ice1712/wm8776.h b/sound/pci/ice1712/wm8776.h index 93a2d69..42acef0 100644 --- a/sound/pci/ice1712/wm8776.h +++ b/sound/pci/ice1712/wm8776.h @@ -216,9 +216,6 @@ struct snd_wm8776 { void snd_wm8776_init(struct snd_wm8776 *wm); void snd_wm8776_resume(struct snd_wm8776 *wm); -void snd_wm8776_set_dac_if(struct snd_wm8776 *wm, u16 dac); -void snd_wm8776_set_adc_if(struct snd_wm8776 *wm, u16 adc); -void snd_wm8776_set_master_mode(struct snd_wm8776 *wm, u16 mode); void snd_wm8776_set_power(struct snd_wm8776 *wm, u16 power); void snd_wm8776_volume_restore(struct snd_wm8776 *wm); int snd_wm8776_build_controls(struct snd_wm8776 *wm); -- 1.7.10.4

3 2