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[alsa-devel] [PATCH] ASoC: adau1701: add regulator consumer support
by pascal.huerst＠gmail.com 20 Apr '15

20 Apr '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> The adau1701 has two power domains, DVDD and AVDD. Enable them both as long as the codec is in use. --- .../devicetree/bindings/sound/adi,adau1701.txt | 4 + sound/soc/codecs/adau1701.c | 125 ++++++++++++++++++--- 2 files changed, 114 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau1701.txt b/Documentation/devicetree/bindings/sound/adi,adau1701.txt index 547a49b..0d1128c 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau1701.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau1701.txt @@ -20,6 +20,8 @@ Optional properties: pin configurations as described in the datasheet, table 53. Note that the value of this property has to be prefixed with '/bits/ 8'. + - avdd-supply: Power supply for AVDD, providing 3.3V + - dvdd-supply: Power supply for DVDD, providing 3.3V Examples: @@ -28,6 +30,8 @@ Examples: compatible = "adi,adau1701"; reg = <0x34>; reset-gpio = <&gpio 23 0>; + avdd-supply = <&vdd_3v3_reg>; + dvdd-supply = <&vdd_3v3_reg>; adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>; adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4 0x4 0x4 0x4 0x4 0x4 0x4>; diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d4e219b..ca94ae8 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -16,6 +16,7 @@ #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/regulator/consumer.h> #include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> @@ -101,6 +102,10 @@ #define ADAU1701_FIRMWARE "adau1701.bin" +static const char * const supply_names[] = { + "dvdd", "avdd" +}; + struct adau1701 { int gpio_nreset; int gpio_pll_mode[2]; @@ -112,6 +117,7 @@ struct adau1701 { u8 pin_config[12]; struct sigmadsp *sigmadsp; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -669,6 +675,13 @@ static int adau1701_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + /* * Let the pll_clkdiv variable default to something that won't happen * at runtime. That way, we can postpone the firmware download from @@ -680,7 +693,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) /* initalize with pre-configured pll mode settings */ ret = adau1701_reset(codec, adau1701->pll_clkdiv, 0); if (ret < 0) - return ret; + goto exit_regulators_disable; /* set up pin config */ val = 0; @@ -696,10 +709,60 @@ static int adau1701_probe(struct snd_soc_codec *codec) regmap_write(adau1701->regmap, ADAU1701_PINCONF_1, val); return 0; + +exit_regulators_disable: + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); + return ret; } +static int adau1701_remove(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(adau1701->gpio_nreset)) + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int adau1701_suspend(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + + return 0; +} + +static int adau1701_resume(struct snd_soc_codec *codec) +{ + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + return adau1701_reset(codec, adau1701->pll_clkdiv, 0); +} +#else +#define adau1701_resume NULL +#define adau1701_suspend NULL +#endif /* CONFIG_PM */ + static struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, + .remove = adau1701_remove, + .resume = adau1701_resume, + .suspend = adau1701_suspend, .set_bias_level = adau1701_set_bias_level, .idle_bias_off = true, @@ -730,32 +793,58 @@ static int adau1701_i2c_probe(struct i2c_client *client, struct device *dev = &client->dev; int gpio_nreset = -EINVAL; int gpio_pll_mode[2] = { -EINVAL, -EINVAL }; - int ret; + int ret, i; adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); if (!adau1701) return -ENOMEM; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + adau1701->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), + adau1701->supplies); + if (ret < 0) { + dev_err(dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, &adau1701_regmap); - if (IS_ERR(adau1701->regmap)) - return PTR_ERR(adau1701->regmap); + if (IS_ERR(adau1701->regmap)) { + ret = PTR_ERR(adau1701->regmap); + goto exit_regulators_disable; + } + if (dev->of_node) { gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); - if (gpio_nreset < 0 && gpio_nreset != -ENOENT) - return gpio_nreset; + if (gpio_nreset < 0 && gpio_nreset != -ENOENT) { + ret = gpio_nreset; + goto exit_regulators_disable; + } gpio_pll_mode[0] = of_get_named_gpio(dev->of_node, "adi,pll-mode-gpios", 0); - if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) - return gpio_pll_mode[0]; + if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) { + ret = gpio_pll_mode[0]; + goto exit_regulators_disable; + } gpio_pll_mode[1] = of_get_named_gpio(dev->of_node, "adi,pll-mode-gpios", 1); - if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) - return gpio_pll_mode[1]; + if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) { + ret = gpio_pll_mode[1]; + goto exit_regulators_disable; + } of_property_read_u32(dev->of_node, "adi,pll-clkdiv", &adau1701->pll_clkdiv); @@ -769,7 +858,7 @@ static int adau1701_i2c_probe(struct i2c_client *client, ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, "ADAU1701 Reset"); if (ret < 0) - return ret; + goto exit_regulators_disable; } if (gpio_is_valid(gpio_pll_mode[0]) && @@ -778,13 +867,13 @@ static int adau1701_i2c_probe(struct i2c_client *client, GPIOF_OUT_INIT_LOW, "ADAU1701 PLL mode 0"); if (ret < 0) - return ret; + goto exit_regulators_disable; ret = devm_gpio_request_one(dev, gpio_pll_mode[1], GPIOF_OUT_INIT_LOW, "ADAU1701 PLL mode 1"); if (ret < 0) - return ret; + goto exit_regulators_disable; } adau1701->gpio_nreset = gpio_nreset; @@ -795,11 +884,17 @@ static int adau1701_i2c_probe(struct i2c_client *client, adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, &adau1701_sigmadsp_ops, ADAU1701_FIRMWARE); - if (IS_ERR(adau1701->sigmadsp)) - return PTR_ERR(adau1701->sigmadsp); + if (IS_ERR(adau1701->sigmadsp)) { + ret = PTR_ERR(adau1701->sigmadsp); + goto exit_regulators_disable; + } ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); + +exit_regulators_disable: + + regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); return ret; } -- 1.9.3

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[alsa-devel] [PATCH 00/13] ADSP Firmware Controls Update
by Charles Keepax 20 Apr '15

20 Apr '15

This series of patches starts with a few small clean ups for the ADSP code, and then finally adds support for the rev 1 and 2 firmware file formats. The revision 1 format is in common usage in the field so it is long overdue to get support for that merged to mainline. The revision 2 format is fairly new but is starting to see some deployment as well. There is still more work todo here. The main next step is to add support for arbitary length controls through the TLV binary control feature that was recently added to the kernel. This patch series will truncate any long controls to 512 bytes. Thanks, Charles Charles Keepax (13): ASoC: wm_adsp: Split out adsp1 & 2 setup algorithms ASoC: wm_adsp: Improve variable naming ASoC: wm_adsp: Remove len field from wm_adsp_alg_region ASoC: wm_adsp: Limit firmware control name to ALSA control name size ASoC: wm_adsp: Move temporary control name to the stack ASoC: wm_adsp: Clean up low level control read/write functions ASoC: wm_adsp: Factor out creation of alg_regions ASoC: wm_adsp: Remove private field from wm_coeff_ctl ASoC: wm_adsp: Group all the ALSA control functions together ASoC: wm_adsp: Add basic support for rev 1 firmware file format ASoC: wm_adsp: Add support for DSP control flags ASoC: wm_adsp: Add support for rev 2 firmware file format ASoC: wm_adsp: Warn that firmware file format 0 is depreciated sound/soc/codecs/wm_adsp.c | 1129 +++++++++++++++++++++++++++----------------- sound/soc/codecs/wm_adsp.h | 6 +- sound/soc/codecs/wmfw.h | 44 ++- 3 files changed, 733 insertions(+), 446 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH 0/2] Add support for select accessory detect mode to HPDETL or HPDETR
by Inha Song 20 Apr '15

20 Apr '15

This set of patches adds support for select accessory detect mode to HPDETL or HPDETR. Inha Song (2): extcon: arizona: Add support for select accessory detect mode when headphone detection" mfd: arizona: Update DT binding to support hpdet channel Documentation/devicetree/bindings/mfd/arizona.txt | 4 +++ drivers/extcon/extcon-arizona.c | 34 ++++++++++++++++++++--- include/linux/mfd/arizona/pdata.h | 3 ++ 3 files changed, 37 insertions(+), 4 deletions(-) -- 2.0.0.390.gcb682f8

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[alsa-devel] How to correct params_format on DPCM ?
by Kuninori Morimoto 20 Apr '15

20 Apr '15

Hi Mark, Lars, all Now, I'm using DPCM based sound, but FE/BE gets different params format. How to correct it ? This sound can be used both non-DPCM card / DPCM card -- non-DPCM card -- cpu: rsnd codec: ak4642 -- DPCM card -- FE : cpu: rsnd codec: snd-soc-dummy BE : cpu: snd-soc-dummy codec: ak4642 Here, SNDRV_PCM_FMTBIT_S16_LE is always selected on non-DPCM card. This is OK. rsnd: SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE ak4642: SNDRV_PCM_FMTBIT_S16_LE On DPCM card, FE gets S24, BE gets S16 if aplay selects S24 wav. But, FE needs S16, same as non-DPCM card. I know we can fixup BE side via dai_link.be_hw_params_fixup, but this is FE side. It can't use correct sound playback in corrent DPCM card. But, how to solve this issue ?? Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: add static inline funcs to fix a compiling issue
by Jie Yang 19 Apr '15

19 Apr '15

When CONFIG_PM_SLEEP is not selected, calling funcs snd_soc_suspend and _resume will generate a compiling issue. Here add static inline stub functions to fix it. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2afdd2e..64041f5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -387,8 +387,20 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); +#ifdef CONFIG_PM_SLEEP int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); +#else +static inline int snd_soc_suspend(struct device *dev) +{ + return 0; +} + +static inline int snd_soc_resume(struct device *dev) +{ + return 0; +} +#endif int snd_soc_poweroff(struct device *dev); int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: fix a compiling issue
by Jie Yang 19 Apr '15

19 Apr '15

Fix the compiling issue that we don't have snd_soc_suspend/resume() when CONFIG_PM_SLEEP is not defined. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 23ae040..f84eb14 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1313,8 +1313,9 @@ static void hsw_pcm_complete(struct device *dev) dev_err(dev, "failed to restore context for PCM %d\n", i); } +#ifdef CONFIG_PM_SLEEP snd_soc_resume(pdata->soc_card->dev); - +#endif err = sst_hsw_dsp_runtime_resume(hsw); if (err < 0) return; @@ -1367,7 +1368,9 @@ static int hsw_pcm_prepare(struct device *dev) sst_hsw_dsp_runtime_sleep(hsw); } +#ifdef CONFIG_PM_SLEEP snd_soc_suspend(pdata->soc_card->dev); +#endif snd_soc_poweroff(pdata->soc_card->dev); pdata->pm_state = HSW_PM_STATE_D3; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: at91sam9g20ek: Automatically disconnect non-connected pins
by Lars-Peter Clausen 18 Apr '15

18 Apr '15

According to the schematics the both LHPOUT and RHPOUT are connected to the external connector. RHPOUT is missing from the DAPM routes, but otherwise they seem to be complete. This patch adds the missing route and then sets the fully_routed flag for the card. This allows to remove all the manual calls to snd_soc_dapm_nc_pin(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/sam9g20_wm8731.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 8de8361..d7469cd 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -95,8 +95,9 @@ static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { static const struct snd_soc_dapm_route intercon[] = { - /* speaker connected to LHPOUT */ + /* speaker connected to LHPOUT/RHPOUT */ {"Ext Spk", NULL, "LHPOUT"}, + {"Ext Spk", NULL, "RHPOUT"}, /* mic is connected to Mic Jack, with WM8731 Mic Bias */ {"MICIN", NULL, "Mic Bias"}, @@ -108,9 +109,7 @@ static const struct snd_soc_dapm_route intercon[] = { */ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; printk(KERN_DEBUG @@ -124,10 +123,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* not connected */ - snd_soc_dapm_nc_pin(dapm, "RLINEIN"); - snd_soc_dapm_nc_pin(dapm, "LLINEIN"); - #ifndef ENABLE_MIC_INPUT snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic"); #endif @@ -158,6 +153,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), .dapm_routes = intercon, .num_dapm_routes = ARRAY_SIZE(intercon), + .fully_routed = true, }; static int at91sam9g20ek_audio_probe(struct platform_device *pdev) -- 1.8.0

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Re: [alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Use specific variable for lo_vag
by Mark Brown 18 Apr '15

18 Apr '15

On Thu, Apr 16, 2015 at 02:51:56PM +0200, Alexander Stein wrote: > This is a preparation for calculating lo_vol which needs both vag > and lo_vag. Applied both, thanks.

1 0

[alsa-devel] Understanding Atmel SSC SND_SOC_DAIFMT_CBS_CFS mode
by Angelo Compagnucci 18 Apr '15

18 Apr '15

Hi List, I'm looking for a way to understand how to correctly configure the Atmel SSC dai on a SAMA5D3 to work in master mode with an I2S codec. >From this comment: /* * I2S format, SSC provides BCLK and LRC clocks. * * The SSC transmit and receive clocks are generated * from the MCK divider, and the BCLK signal * is output on the SSC TK line. */ My question is: should be MCK divider configured manually? If yes how? Actually I'm trying to poke around in atmel_ssc_dai and found a way to generate a stable bit clock, but frame clock is not working as expected. Have you any advice on how to set correctly SSC parameters to generate a stable bit clock and frame clock? Any advice? Thank you! -- Profile: http://it.linkedin.com/in/compagnucciangelo

1 0

[alsa-devel] Support for Synchronous Audio
by Anoop Rajan 18 Apr '15

18 Apr '15

Hi All, My self Anoop, an audio frame work developer, but a newbie to linux ALSA. We have requirements to play synchronous multi-room audio with very minimal node to node latency(<100 us).( Node to node latency: How much one speaker is out of sync with other). Right now we have logic in user space to synchronize the audio using time stamps derived from a common clock, but the node to node latency is over 1ms. Though I make sure that I start the PCM from user layer synchronously, the actual I2S out is out of sync(in consistent values ranging from 100 to 1ms). I beleive the un-deterministic path delay from user to ALSA buffer, big ALSA buffer(8k Samples), scheduling variances(high priority/interrupt preemption) all might have added to this increased latency. Rather than fine tuning the above discrepancies , I was thinking about some designs to pass the audio frames with time stamps to kernel layer and to implement some logic in kernel (before pushing to DMA) to present the audio at exact stamped time to have a better node to node latency. Before doing that I was trying to find out what ALSA supports for this requirement and from web could see various comments regarding this (proposal to snd_pcm_start_at()<http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/082091.ht…>) and further proposed discussions on Audio mini summit, but I couldn't find further information's on it. Would you please guide me. Does ALSA provides some means to achieve minimal Node to Node latency? would you please provide some pointers/documents to me to understand better. Thanks, Anoop

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[alsa-devel] [PATCH] sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND)
by Alexey Khoroshilov 18 Apr '15

18 Apr '15

A deadlock can be initiated by userspace via ioctl(SNDCTL_SEQ_OUTOFBAND) on /dev/sequencer with TMR_ECHO midi event. In this case the control flow is: sound_ioctl() -> case SND_DEV_SEQ: case SND_DEV_SEQ2: sequencer_ioctl() -> case SNDCTL_SEQ_OUTOFBAND: spin_lock_irqsave(&lock,flags); play_event(); -> case EV_TIMING: seq_timing_event() -> case TMR_ECHO: seq_copy_to_input() -> spin_lock_irqsave(&lock,flags); It seems that spin_lock_irqsave() around play_event() is not necessary, because the only other call location in seq_startplay() makes the call without acquiring spinlock. So, the patch just removes spinlocks around play_event(). By the way, it removes unreachable code in seq_timing_event(), since (seq_mode == SEQ_2) case is handled in the beginning. Compile tested only. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> --- sound/oss/sequencer.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index c0eea1dfe90f..f19da4b47c1d 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec) break; case TMR_ECHO: - if (seq_mode == SEQ_2) - seq_copy_to_input(event_rec, 8); - else - { - parm = (parm << 8 | SEQ_ECHO); - seq_copy_to_input((unsigned char *) &parm, 4); - } + parm = (parm << 8 | SEQ_ECHO); + seq_copy_to_input((unsigned char *) &parm, 4); break; default:; @@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a int mode = translate_mode(file); struct synth_info inf; struct seq_event_rec event_rec; - unsigned long flags; int __user *p = arg; orig_dev = dev = dev >> 4; @@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a case SNDCTL_SEQ_OUTOFBAND: if (copy_from_user(&event_rec, arg, sizeof(event_rec))) return -EFAULT; - spin_lock_irqsave(&lock,flags); play_event(event_rec.arr); - spin_unlock_irqrestore(&lock,flags); return 0; case SNDCTL_MIDI_INFO: -- 1.9.1

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Re: [alsa-devel] Lost Interrput on VIA VT1708S
by Raymond Yau 17 Apr '15

17 Apr '15

> > The second opint is what you have an issue with - see "Lost interrupts" in comment #28. > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/c… hwptr tracking was replaced by tracepoint This seem to be the reason that you only find those Lost Interrput in the system log https://bugs.freedesktop.org/show_bug.cgi?id=49608#c28 https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/cor…

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[alsa-devel] [PATCH -next] ASoC: Intel: sst_byt: remove kfree for memory allocated with devm_kzalloc
by weiyj_lk＠163.com 17 Apr '15

17 Apr '15

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free memory allocated with devm_kzalloc and using kfree leads to a double free. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/baytrail/sst-baytrail-ipc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/baytrail/sst-baytrail-ipc.c b/sound/soc/intel/baytrail/sst-baytrail-ipc.c index 1efb33b..a839dbf 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -759,7 +759,6 @@ fw_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(byt); return err; }

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[alsa-devel] [patch] ALSA: hda - potential (but unlikely) uninitialized variable
by Dan Carpenter 17 Apr '15

17 Apr '15

This function is a bit unusual because it accepts negative values as "conn_len". It's theoretically possible for both "cache_len" and "conn_len" to be -ENOSPC and in that case we would oops trying to run memcmp() on the uninitialized "list" pointer. My static checker complains about this. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ce5a6da..387fdfc 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -556,8 +556,8 @@ static void print_conn_list(struct snd_info_buffer *buffer, /* Get Cache connections info */ cache_len = snd_hda_get_conn_list(codec, nid, &list); - if (cache_len != conn_len - || memcmp(list, conn, conn_len)) { + if (cache_len < 0 || cache_len != conn_len || + memcmp(list, conn, conn_len) != 0) { snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len); if (cache_len > 0) { snd_iprintf(buffer, " ");

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[alsa-devel] [PATCH -next] ASoC: Intel: sst_hsw: remove kfree for memory allocated with devm_kzalloc
by weiyj_lk＠163.com 17 Apr '15

17 Apr '15

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> It's not necessary to free memory allocated with devm_kzalloc and using kfree leads to a double free. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 344a1e9..324eceb 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2201,7 +2201,6 @@ dma_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init);

4 3

[alsa-devel] [PATCH -next] ASoC: cs35l32: Remove unused including <linux/version.h>
by weiyj_lk＠163.com 17 Apr '15

17 Apr '15

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Remove including <linux/version.h> that don't need it. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/cs35l32.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 60598b2..8f40025 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -13,7 +13,6 @@ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h>

3 2

[alsa-devel] [PATCH] ASoC: samsung: s3c24xx-i2s: Fix return value check in s3c24xx_iis_dev_probe()
by weiyj_lk＠163.com 17 Apr '15

17 Apr '15

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function devm_ioremap_resource() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 326d3c3..5bf7236 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -461,8 +461,8 @@ static int s3c24xx_iis_dev_probe(struct platform_device *pdev) return -ENOENT; } s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + if (IS_ERR(s3c24xx_i2s.regs)) + return PTR_ERR(s3c24xx_i2s.regs); s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO;

3 2

[alsa-devel] [PATCH] ASoC: tfa9879: Fix return value check in tfa9879_i2c_probe()
by weiyj_lk＠163.com 17 Apr '15

17 Apr '15

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function devm_kzalloc() returns NULL not ERR_PTR(). The IS_ERR() test in the return value check should be replaced with NULL test. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/tfa9879.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tfa9879.c b/sound/soc/codecs/tfa9879.c index 16f1b71..aab0af6 100644 --- a/sound/soc/codecs/tfa9879.c +++ b/sound/soc/codecs/tfa9879.c @@ -280,8 +280,8 @@ static int tfa9879_i2c_probe(struct i2c_client *i2c, int i; tfa9879 = devm_kzalloc(&i2c->dev, sizeof(*tfa9879), GFP_KERNEL); - if (IS_ERR(tfa9879)) - return PTR_ERR(tfa9879); + if (!tfa9879) + return -ENOMEM; i2c_set_clientdata(i2c, tfa9879);

3 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix platform_get_irq() error handling
by Fabio Estevam 17 Apr '15

17 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> We should check whether platform_get_irq() returns a negative number and propagate the error in this case. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index e8bb8ee..0d48804 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1357,7 +1357,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->irq = platform_get_irq(pdev, 0); - if (!ssi_private->irq) { + if (ssi_private->irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", pdev->name); return ssi_private->irq; } -- 1.9.1

2 1

[alsa-devel] [PATCH v3 0/2] ALSA: HDA: add stream params to hda core
by Vinod Koul 17 Apr '15

17 Apr '15

This two patches are splut form the series v2 as they are specfic to HDAC. First adds ASoC device type later to be used by ASOC driver and second for the stream params. Jeeja KP (1): ALSA: hda - add generic functions to set hdac stream params Ramesh Babu (1): ALSA: hda - add ASoC device type for hda core include/sound/hdaudio.h | 3 +++ sound/hda/hdac_stream.c | 38 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 41 insertions(+) -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCH] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 17 Apr '15

17 Apr '15

在 2015年04月17日 15:14, Anish Kumar 写道: > 在 2015年04月16日 10:41, Anish Kumar 写道: >> 在 2015年04月16日 05:14, Anish Kumar 写道: >>> 在 04/10/2015 03:38 AM, Anish Kumar 写道: >>>>> To fix pop noise when shutdown,the pop noise during shutdown >>>>> is the pmic cutoff power of codec without any notice. >>>> Is the power turned off without any intimation to the codec driver? >>>> Or is it physically removed? Can you describe the probelm >>>> better? >>>> When device is shutdown with playing music, the origin driver does nothing for shutdown operation, >>>> as a result, pmic will cut off the power of codec IC, thus cause a pop noise from speaker and headphone. >>>> So the patch added a shutdown callback for driver, enable volume smoothing and disable zero cross, >>>> this is referenced from mic_input which does the samething before mic switch. >>>> I'm very happy if some suggestion from maximintegrate . :-) >>> Please let me know if the patch worked for you. >>> Yes, the patch can work for me. >> I hope this patch is what you tested. >> >> +static void max98090_i2c_shutdown(struct i2c_client *i2c) >> +{ >> + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); >> + >> + dev_info(&i2c->dev, "shut down device\n"); >> + >> + /* enable zero crossing */ >> + regmap_write(max98090->regmap, >> + M98090_REG_LEVEL_CONTROL, M98090_ZDENN_MASK-1); >> + >> + /* turn down volume levels for all the outputs */ >> + regmap_write(max98090->regmap, >> + M98090_REG_RCV_LOUTL_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_LOUTR_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_LEFT_SPK_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_RIGHT_SPK_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_LEFT_HP_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_RIGHT_HP_VOLUME, 0x00); >> + regmap_write(max98090->regmap, >> + M98090_REG_DEVICE_SHUTDOWN, 0x00); >> + msleep(40); >> +} >> + >> Please go ahead and submit to Mark. > With the about is also useful to solve it. > > @zengxing, could you also help verify it? > Verified ? zengxing, who told me is useful to solve it. >>> We had tested on chromiumos. >>>> +static void max98090_i2c_shutdown(struct i2c_client *i2c) >>>> +{ >>>> + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); >>>> + >>>> + dev_info(&i2c->dev, "shut down device\n"); >>>> + >>>> + /* Enable volume smoothing, disable zero cross. This will cause >>>> + * a quick 40ms ramp to mute on shutdown. >>>> + */ >>>> + regmap_write(max98090->regmap, >>>> + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); >>>> + regmap_write(max98090->regmap, >>>> + M98090_REG_DEVICE_SHUTDOWN, 0x00); >>>> + msleep(40); >>>> +} >>>> + >>>> >>>> >>> >>> >> -- >> ************************************************************************************** >> 王晓腾 Caesar Wang >> Product R&D Dept.III >> Fuzhou Rockchip Electronics Co.Ltd >> Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) >> 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) >> Tel：+86-591-83991906/07 - 8221 >> Mobile:+86 15059456742 >> E-mail : wxt(a)rock-chips.com >> *************************************************************************************** >> *************************************************************************************** >> IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may >> contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. >> If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the >> information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material >> in its entirety in any format. Thank you. >> *************************************************************************************** >> > -- > ************************************************************************************** > 王晓腾 Caesar Wang > Product R&D Dept.III > Fuzhou Rockchip Electronics Co.Ltd > Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) > 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) > Tel：+86-591-83991906/07 - 8221 > Mobile:+86 15059456742 > E-mail : wxt(a)rock-chips.com > *************************************************************************************** > *************************************************************************************** > IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may > contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. > If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the > information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material > in its entirety in any format. Thank you. > *************************************************************************************** > > -- ************************************************************************************** 王晓腾 Caesar Wang Product R&D Dept.III Fuzhou Rockchip Electronics Co.Ltd Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) Tel：+86-591-83991906/07 - 8221 Mobile:+86 15059456742 E-mail : wxt(a)rock-chips.com *************************************************************************************** *************************************************************************************** IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material in its entirety in any format. Thank you. ***************************************************************************************

1 0

[alsa-devel] [RFC 3/4] ASoC: topology: Add topology support to ASoC core and DAPM.
by Liam Girdwood 16 Apr '15

16 Apr '15

Add dynamic object support to ASoC kcontrols, widgets, components and cards. Remove all dynamic objects on sound card removal and DAPM free. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 11 +++++++++++ sound/soc/soc-core.c | 4 ++++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 21 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 3263e36..8d28b36 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,7 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/soc-topology.h> #include <sound/asoc.h> struct device; @@ -533,6 +534,7 @@ struct snd_soc_dapm_widget { int num_kcontrols; const struct snd_kcontrol_new *kcontrol_news; struct snd_kcontrol **kcontrols; + struct snd_soc_dobj dobj; /* widget input and outputs */ struct list_head sources; diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade1..e7bc994 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -27,6 +27,7 @@ #include <sound/compress_driver.h> #include <sound/control.h> #include <sound/ac97_codec.h> +#include <sound/soc-topology.h> /* * Convenience kcontrol builders @@ -753,6 +754,9 @@ struct snd_soc_component { struct mutex io_mutex; + /* attached dynamic objects */ + struct list_head dobj_list; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_root; #endif @@ -1088,6 +1092,9 @@ struct snd_soc_card { struct list_head dapm_list; struct list_head dapm_dirty; + /* attached dynamic objects */ + struct list_head dobj_list; + /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; struct snd_soc_dapm_stats dapm_stats; @@ -1147,6 +1154,7 @@ struct soc_mixer_control { unsigned int sign_bit; unsigned int invert:1; unsigned int autodisable:1; + struct snd_soc_dobj dobj; }; struct soc_bytes { @@ -1157,6 +1165,8 @@ struct soc_bytes { struct soc_bytes_ext { int max; + struct snd_soc_dobj dobj; + /* used for TLV byte control */ int (*get)(unsigned int __user *bytes, unsigned int size); int (*put)(const unsigned int __user *bytes, unsigned int size); @@ -1177,6 +1187,7 @@ struct soc_enum { unsigned int mask; const char * const *texts; const unsigned int *values; + struct snd_soc_dobj dobj; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9dfa2e2..3a6ae19 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -40,6 +40,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dpcm.h> +#include <sound/soc-topology.h> #include <sound/initval.h> #define CREATE_TRACE_POINTS @@ -2434,6 +2435,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->rtd_aux[i].card = card; INIT_LIST_HEAD(&card->dapm_dirty); + INIT_LIST_HEAD(&card->dobj_list); card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); @@ -2748,6 +2750,7 @@ static void snd_soc_component_add_unlocked(struct snd_soc_component *component) } list_add(&component->list, &component_list); + INIT_LIST_HEAD(&component->dobj_list); } static void snd_soc_component_add(struct snd_soc_component *component) @@ -2824,6 +2827,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: + snd_soc_tplg_component_remove(cmpnt, SND_SOC_TPLG_ID_ALL); snd_soc_component_del_unlocked(cmpnt); mutex_unlock(&client_mutex); snd_soc_component_cleanup(cmpnt); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 86d350a..c768e02 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -41,6 +41,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-topology.h> #include <sound/initval.h> #include <trace/events/asoc.h> @@ -2165,6 +2166,9 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) list_for_each_entry_safe(p, next_p, &w->sinks, list_source) dapm_free_path(p); + /* check and free and dynamic widget kcontrols */ + snd_soc_tplg_widget_remove(w); + kfree(w->kcontrols); kfree(w->name); kfree(w); -- 2.1.0

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[alsa-devel] [RFC 2/4] ASoC: topology: Add topology core
by Liam Girdwood 16 Apr '15

16 Apr '15

The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-topology.h | 167 ++++ sound/soc/soc-topology.c | 1788 ++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 1955 insertions(+) create mode 100644 include/sound/soc-topology.h create mode 100644 sound/soc/soc-topology.c diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h new file mode 100644 index 0000000..a235004 --- /dev/null +++ b/include/sound/soc-topology.h @@ -0,0 +1,167 @@ +/* + * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simple file API to load FW that includes mixers, coefficients, DAPM graphs, + * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc. + */ + +#ifndef __LINUX_SND_SOC_TPLG_H +#define __LINUX_SND_SOC_TPLG_H + +#include <sound/asoc.h> +#include <linux/list.h> + +struct firmware; +struct snd_kcontrol; +struct snd_soc_tplg_pcm_be; +struct snd_ctl_elem_value; +struct snd_ctl_elem_info; +struct snd_soc_dapm_widget; +struct snd_soc_component; +struct snd_soc_tplg_pcm_fe; +struct snd_soc_dapm_context; +struct snd_soc_card; + +/* object scan be loaded and unloaded in groups with identfying indexes */ +#define SND_SOC_TPLG_ID_ALL 0 /* ID that matches all FW objects */ + +/* dynamic object type */ +enum snd_soc_dobj_type { + SND_SOC_DOBJ_NONE = 0, /* object is not dynamic */ + SND_SOC_DOBJ_MIXER, + SND_SOC_DOBJ_ENUM, + SND_SOC_DOBJ_BYTES, + SND_SOC_DOBJ_PCM, + SND_SOC_DOBJ_DAI_LINK, + SND_SOC_DOBJ_CODEC_LINK, + SND_SOC_DOBJ_WIDGET, +}; + +/* dynamic control object */ +struct snd_soc_dobj_control { + struct snd_kcontrol *kcontrol; + char **dtexts; + unsigned long *dvalues; +}; + +/* dynamic widget object */ +struct snd_soc_dobj_widget { + unsigned int kcontrol_enum:1; /* this widget is an enum kcontrol */ +}; + +/* dynamic PCM DAI object */ +struct snd_soc_dobj_pcm_dai { + struct snd_soc_tplg_pcm_dai *pd; + unsigned int count; +}; + +/* generic dynamic object - all dynamic objects belong to this struct */ +struct snd_soc_dobj { + enum snd_soc_dobj_type type; + unsigned int id; /* objects can belong in different groups */ + struct list_head list; + struct snd_soc_tplg_ops *ops; + union { + struct snd_soc_dobj_control control; + struct snd_soc_dobj_widget widget; + struct snd_soc_dobj_pcm_dai pcm_dai; + }; + void *private; /* core does not touch this */ +}; + +/* + * Kcontrol operations - used to map handlers onto firmware based controls. + */ +struct snd_soc_tplg_kcontrol_ops { + u32 id; + int (*get)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*put)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + int (*info)(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +}; + +/* + * DAPM widget event handlers - used to map handlers onto widgets. + */ +struct snd_soc_tplg_widget_events { + u16 type; + int (*event_handler)(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event); +}; + +/* + * Public API - Used by component drivers to load and unload dynamic objects + * and their resources. + */ +struct snd_soc_tplg_ops { + + /* external kcontrol init - used to set ext funcs + private data */ + int (*control_load) (struct snd_soc_component *, + struct snd_kcontrol_new *, struct snd_soc_tplg_control_hdr *); + int (*control_unload) (struct snd_soc_component *, + struct snd_soc_dobj *); + + /* external widget init - used to set ext funcs + private data */ + int (*widget_load) (struct snd_soc_component *, + struct snd_soc_dapm_widget *, + struct snd_soc_tplg_dapm_widget *); + int (*widget_unload) (struct snd_soc_component *, + struct snd_soc_dobj *); + + /* FE - used to set ext funcs + private data */ + int (*pcm_dai_load) (struct snd_soc_component *, + struct snd_soc_tplg_pcm_dai *pcm_dai, int num_fe); + int (*pcm_dai_unload) (struct snd_soc_component *, + struct snd_soc_dobj *); + + /* callback to handle vendor bespoke data */ + int (*vendor_load) (struct snd_soc_component *, + struct snd_soc_tplg_hdr *); + int (*vendor_unload) (struct snd_soc_component *, + struct snd_soc_tplg_hdr *); + + /* completion - called at completion of firmware loading */ + void (*complete) (struct snd_soc_component *); + + /* manifest - optional to inform component of manifest */ + int (*manifest) (struct snd_soc_component *, + struct snd_soc_tplg_manifest *); + + /* bespoke kcontrol handlers available for binding */ + const struct snd_soc_tplg_kcontrol_ops *io_ops; + int io_ops_count; +}; + +/* gets a pointer to data from the firmware block header */ +static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) +{ + const void *ptr = hdr; + return ptr + sizeof(*hdr); +} + +/* Dynamic Object loading and removal for component drivers */ +int snd_soc_tplg_component_load(struct snd_soc_component *comp, + struct snd_soc_tplg_ops *ops, const struct firmware *fw, + u32 index); +int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index); + +/* Widget removal - widgets also removed wth component API */ +void snd_soc_tplg_widget_remove(struct snd_soc_dapm_widget *w); +void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, + u32 index); + +/* Binds event handlers to dynamic widgets */ +int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, + const struct snd_soc_tplg_widget_events *events, int num_events, + u16 event_type); + +#endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c new file mode 100644 index 0000000..620490c --- /dev/null +++ b/sound/soc/soc-topology.c @@ -0,0 +1,1788 @@ +/* + * soc-topology.c -- ALSA SoC Topology + * + * Copyright (C) 2012 Texas Instruments Inc. + * Copyright (C) 2015 Intel Corporation. + * + * Authors: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> + * K, Mythri P <mythri.p.k(a)intel.com> + * Prusty, Subhransu S <subhransu.s.prusty(a)intel.com> + * B, Jayachandran <jayachandran.b(a)intel.com> + * Abdullah, Omair M <omair.m.abdullah(a)intel.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Add support to read audio firmware topology alongside firmware text. The + * topology data can contain kcontrols, DAPM graphs, widgets, DAIs, DAI links, + * equalizers, firmware, coefficients etc. + * + * This file only manages the core ALSA and ASoC components, all other bespoke + * firmware topology data is passed to component drivers for bespoke handling. + */ + +#include <linux/kernel.h> +#include <linux/export.h> +#include <linux/list.h> +#include <linux/firmware.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/soc-topology.h> + +/* + * We make several passes over the data (since it wont necessarily be ordered) + * and process objects in the following order. This guarantees the component + * drivers will be ready with any vendor data before the mixers and DAPM objects + * are loaded (that may make use of the vendor data). + */ +#define SOC_TPLG_PASS_MANIFEST 0 +#define SOC_TPLG_PASS_VENDOR 1 +#define SOC_TPLG_PASS_MIXER 2 +#define SOC_TPLG_PASS_WIDGET 3 +#define SOC_TPLG_PASS_GRAPH 4 +#define SOC_TPLG_PASS_PINS 5 +#define SOC_TPLG_PASS_PCM_DAI 6 + +#define SOC_TPLG_PASS_START SOC_TPLG_PASS_MANIFEST +#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PCM_DAI + +struct soc_tplg { + const char *file; + const struct firmware *fw; + + /* runtime FW parsing */ + const u8 *pos; /* read postion */ + const u8 *hdr_pos; /* header position */ + unsigned int pass; /* pass number */ + + /* component caller */ + struct device *dev; + struct snd_soc_component *comp; + u32 id; /* current block ID */ + u32 req_id; /* required ID, only loaded/free matching blocks */ + + /* kcontrol operations */ + const struct snd_soc_tplg_kcontrol_ops *io_ops; + int io_ops_count; + + /* optional fw loading callbacks to component drivers */ + struct snd_soc_tplg_ops *ops; +}; + +static int soc_tplg_process_headers(struct soc_tplg *tplg); +static void soc_tplg_complete(struct soc_tplg *tplg); + +/* mapping of Kcontrol types and associated operations. */ +static const struct snd_soc_tplg_kcontrol_ops io_ops[] = { + {SOC_CONTROL_IO_VOLSW, snd_soc_get_volsw, + snd_soc_put_volsw, snd_soc_info_volsw}, + {SOC_CONTROL_IO_VOLSW_SX, snd_soc_get_volsw_sx, + snd_soc_put_volsw_sx, NULL}, + {SOC_CONTROL_IO_ENUM, snd_soc_get_enum_double, + snd_soc_put_enum_double, snd_soc_info_enum_double}, + {SOC_CONTROL_IO_ENUM_EXT, NULL, + NULL, snd_soc_info_enum_double}, + {SOC_CONTROL_IO_BYTES, snd_soc_bytes_get, + snd_soc_bytes_put, snd_soc_bytes_info}, + {SOC_CONTROL_IO_BOOL_EXT, NULL, + NULL, snd_ctl_boolean_mono_info}, + {SOC_CONTROL_IO_BYTES_EXT, NULL, + NULL, snd_soc_bytes_info_ext}, + {SOC_CONTROL_IO_ENUM_VALUE, snd_soc_get_enum_double, + snd_soc_put_enum_double, NULL}, + {SOC_CONTROL_IO_RANGE, snd_soc_get_volsw_range, + snd_soc_put_volsw_range, snd_soc_info_volsw_range}, + {SOC_CONTROL_IO_VOLSW_XR_SX, snd_soc_get_xr_sx, + snd_soc_put_xr_sx, snd_soc_info_xr_sx}, + {SOC_CONTROL_IO_STROBE, snd_soc_get_strobe, + snd_soc_put_strobe, NULL}, + + {SOC_CONTROL_IO_VOLSW_EXT, NULL, + NULL, snd_soc_info_volsw}, + {SOC_DAPM_IO_VOLSW, snd_soc_dapm_get_volsw, + snd_soc_dapm_put_volsw, NULL}, + {SOC_DAPM_IO_ENUM_DOUBLE, snd_soc_dapm_get_enum_double, + snd_soc_dapm_put_enum_double, snd_soc_info_enum_double}, + {SOC_DAPM_IO_ENUM_VIRT, snd_soc_dapm_get_enum_double, + snd_soc_dapm_put_enum_double, NULL}, + {SOC_DAPM_IO_ENUM_VALUE, snd_soc_dapm_get_enum_double, + snd_soc_dapm_put_enum_double, NULL}, + {SOC_DAPM_IO_PIN, snd_soc_dapm_get_pin_switch, + snd_soc_dapm_put_pin_switch, snd_soc_dapm_info_pin_switch}, +}; + +/* check we dont overflow the data for this control chunk */ +static int soc_tplg_check_elem_count(struct soc_tplg *tplg, size_t elem_size, + unsigned int count, size_t bytes, const char *elem_type) +{ + const u8 *end = tplg->pos + elem_size * count; + + if (end > tplg->fw->data + tplg->fw->size) { + dev_err(tplg->dev, "ASoC: %s overflow end of data\n", + elem_type); + return -EINVAL; + } + + /* check there is enough room in chunk for control. + extra bytes at the end of control are for vendor data here */ + if (elem_size * count > bytes) { + dev_err(tplg->dev, + "ASoC: %s count %d of size %zu is bigger than chunk %zu\n", + elem_type, count, elem_size, bytes); + return -EINVAL; + } + + return 0; +} + +static inline int soc_tplg_is_eof(struct soc_tplg *tplg) +{ + const u8 *end = tplg->hdr_pos; + + if (end >= tplg->fw->data + tplg->fw->size) + return 1; + return 0; +} + +static inline unsigned long soc_tplg_get_hdr_offset(struct soc_tplg *tplg) +{ + return (unsigned long)(tplg->hdr_pos - tplg->fw->data); +} + +static inline unsigned long soc_tplg_get_offset(struct soc_tplg *tplg) +{ + return (unsigned long)(tplg->pos - tplg->fw->data); +} + +static enum snd_soc_dobj_type get_dobj_mixer_type( + struct snd_soc_tplg_control_hdr *control_hdr) +{ + if (control_hdr == NULL) + return SND_SOC_DOBJ_NONE; + + switch (SOC_CONTROL_GET_ID_INFO(control_hdr->index)) { + case SOC_CONTROL_TYPE_VOLSW: + case SOC_CONTROL_TYPE_VOLSW_EXT: + case SOC_CONTROL_TYPE_STROBE: + case SOC_CONTROL_TYPE_VOLSW_SX: + case SOC_CONTROL_TYPE_VOLSW_S8: + case SOC_CONTROL_TYPE_VOLSW_XR_SX: + case SOC_CONTROL_TYPE_BYTES: + case SOC_CONTROL_TYPE_BOOL_EXT: + case SOC_CONTROL_TYPE_RANGE: + case SOC_DAPM_TYPE_VOLSW: + case SOC_DAPM_TYPE_PIN: + return SND_SOC_DOBJ_MIXER; + case SOC_CONTROL_TYPE_ENUM: + case SOC_CONTROL_TYPE_ENUM_EXT: + case SOC_CONTROL_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_DOUBLE: + case SOC_DAPM_TYPE_ENUM_VIRT: + case SOC_DAPM_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_EXT: + return SND_SOC_DOBJ_ENUM; + case SOC_CONTROL_TYPE_BYTES_EXT: + return SND_SOC_DOBJ_BYTES; + default: + return SND_SOC_DOBJ_NONE; + } +} + +static enum snd_soc_dobj_type get_dobj_type(struct snd_soc_tplg_hdr *hdr, + struct snd_soc_tplg_control_hdr *control_hdr) +{ + switch (hdr->type) { + case SND_SOC_TPLG_MIXER: + return get_dobj_mixer_type(control_hdr); + case SND_SOC_TPLG_DAPM_GRAPH: + case SND_SOC_TPLG_MANIFEST: + return SND_SOC_DOBJ_NONE; + case SND_SOC_TPLG_DAPM_WIDGET: + return SND_SOC_DOBJ_WIDGET; + case SND_SOC_TPLG_DAI_LINK: + return SND_SOC_DOBJ_DAI_LINK; + case SND_SOC_TPLG_PCM: + return SND_SOC_DOBJ_PCM; + case SND_SOC_TPLG_CODEC_LINK: + return SND_SOC_DOBJ_CODEC_LINK; + default: + return SND_SOC_DOBJ_NONE; + } +} + +static inline void soc_bind_err(struct soc_tplg *tplg, int idx, int count) +{ + dev_err(tplg->dev, + "ASoC: invalid control type %d:%d:%d count %d\n", + SOC_CONTROL_GET_ID_GET(idx), SOC_CONTROL_GET_ID_PUT(idx), + SOC_CONTROL_GET_ID_INFO(idx), count); +} + +static inline void soc_control_err(struct soc_tplg *tplg, const char *name, + int idx) +{ + dev_err(tplg->dev, + "ASoC: no complete mixer IO handler for %s type (g,p,i) %d:%d:%d\n", + name, SOC_CONTROL_GET_ID_GET(idx), + SOC_CONTROL_GET_ID_PUT(idx), SOC_CONTROL_GET_ID_INFO(idx)); +} + +/* pass vendor data to component driver for processing */ +static int soc_tplg_vendor_load_(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + int ret = 0; + + if (tplg->comp && tplg->ops && tplg->ops->vendor_load) + ret = tplg->ops->vendor_load(tplg->comp, hdr); + + if (ret < 0) + dev_err(tplg->dev, + "ASoC: vendor load failed at hdr offset %ld/0x%lx for type %d:%d\n", + soc_tplg_get_hdr_offset(tplg), + soc_tplg_get_hdr_offset(tplg), + hdr->type, hdr->vendor_type); + return ret; +} + +/* pass vendor data to component driver for processing */ +static int soc_tplg_vendor_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + if (tplg->pass != SOC_TPLG_PASS_VENDOR) + return 0; + + return soc_tplg_vendor_load_(tplg, hdr); +} + +/* optionally pass new dynamic widget to component driver. This is mainly for + * external widgets where we can assign private data/ops */ +static int soc_tplg_widget_load(struct soc_tplg *tplg, + struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) +{ + if (tplg->comp && tplg->ops && tplg->ops->widget_load) + return tplg->ops->widget_load(tplg->comp, w, tplg_w); + + dev_dbg(tplg->dev, "ASoC: no handler specified for ext widget %s\n", + w->name); + return 0; +} + +/* pass dynamic FEs configurations to component driver */ +static int soc_tplg_pcm_dai_load(struct soc_tplg *tplg, + struct snd_soc_tplg_pcm_dai *pcm_dai, int num_pcm_dai) +{ + if (tplg->comp && tplg->ops && tplg->ops->pcm_dai_load) + return tplg->ops->pcm_dai_load(tplg->comp, pcm_dai, num_pcm_dai); + + dev_dbg(tplg->dev, "ASoC: no handler specified for FEs\n"); + return 0; +} + +/* tell the component driver that all firmware has been loaded in this request */ +static void soc_tplg_complete(struct soc_tplg *tplg) +{ + if (tplg->comp && tplg->ops && tplg->ops->complete) + tplg->ops->complete(tplg->comp); +} + +/* add a dynamic kcontrol */ +static int soc_tplg_add_dcontrol(struct snd_card *card, struct device *dev, + const struct snd_kcontrol_new *control_new, const char *prefix, + void *data, struct snd_kcontrol **kcontrol) +{ + int err; + + *kcontrol = snd_soc_cnew(control_new, data, control_new->name, prefix); + if (*kcontrol == NULL) { + dev_err(dev, "ASoC: Failed to create new kcontrol %s\n", + control_new->name); + return -ENOMEM; + } + + err = snd_ctl_add(card, *kcontrol); + if (err < 0) { + dev_err(dev, "ASoC: Failed to add %s: %d\n", + control_new->name, err); + return err; + } + + return 0; +} + +/* add a dynamic kcontrol for component driver */ +static int soc_tplg_add_kcontrol(struct soc_tplg *tplg, + struct snd_kcontrol_new *k, struct snd_kcontrol **kcontrol) +{ + struct snd_soc_component *comp = tplg->comp; + + return soc_tplg_add_dcontrol(comp->card->snd_card, + comp->dev, k, NULL, comp, kcontrol); +} + +/* remove a mixer kcontrol */ +static void remove_mixer(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + struct snd_card *card = comp->card->snd_card; + struct soc_mixer_control *sm = + container_of(dobj, struct soc_mixer_control, dobj); + const unsigned int *p = NULL; + + if (pass != SOC_TPLG_PASS_MIXER) + return; + + if (dobj->ops && dobj->ops->control_unload) + dobj->ops->control_unload(comp, dobj); + + if (sm->dobj.control.kcontrol->tlv.p) + p = sm->dobj.control.kcontrol->tlv.p; + snd_ctl_remove(card, sm->dobj.control.kcontrol); + list_del(&sm->dobj.list); + kfree(sm); + kfree(p); +} + +/* remove an enum kcontrol */ +static void remove_enum(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + struct snd_card *card = comp->card->snd_card; + struct soc_enum *se = container_of(dobj, struct soc_enum, dobj); + int i; + + if (pass != SOC_TPLG_PASS_MIXER) + return; + + if (dobj->ops && dobj->ops->control_unload) + dobj->ops->control_unload(comp, dobj); + + snd_ctl_remove(card, se->dobj.control.kcontrol); + list_del(&se->dobj.list); + + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); + kfree(se); +} + +/* remove a byte kcontrol */ +static void remove_bytes(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + struct snd_card *card = comp->card->snd_card; + struct soc_bytes_ext *sb = + container_of(dobj, struct soc_bytes_ext, dobj); + + if (pass != SOC_TPLG_PASS_MIXER) + return; + + if (dobj->ops && dobj->ops->control_unload) + dobj->ops->control_unload(comp, dobj); + + snd_ctl_remove(card, sb->dobj.control.kcontrol); + list_del(&sb->dobj.list); + kfree(sb); +} + +/* remove a widget and it's kcontrols - routes must be removed first */ +static void remove_widget(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + struct snd_card *card = comp->card->snd_card; + struct snd_soc_dapm_widget *w = + container_of(dobj, struct snd_soc_dapm_widget, dobj); + int i; + + if (pass != SOC_TPLG_PASS_WIDGET) + return; + + if (dobj->ops && dobj->ops->widget_unload) + dobj->ops->widget_unload(comp, dobj); + + /* + * Dynamic Widgets either have 1 enum kcontrol or 1..N mixers. + * The enum may either have an array of values or strings. + */ + if (dobj->widget.kcontrol_enum) { + /* enumerated widget mixer */ + struct soc_enum *se = + (struct soc_enum *)w->kcontrols[0]->private_value; + + snd_ctl_remove(card, w->kcontrols[0]); + + kfree(se->dobj.control.dvalues); + for (i = 0; i < se->items; i++) + kfree(se->dobj.control.dtexts[i]); + + kfree(se); + kfree(w->kcontrol_news); + } else { + /* non enumerated widget mixer */ + for (i = 0; i < w->num_kcontrols; i++) { + struct snd_kcontrol *kcontrol = w->kcontrols[i]; + struct soc_mixer_control *sm = + (struct soc_mixer_control *) kcontrol->private_value; + + kfree(w->kcontrols[i]->tlv.p); + + snd_ctl_remove(card, w->kcontrols[i]); + kfree(sm); + } + kfree(w->kcontrol_news); + } + /* widget w is freed by soc-dapm.c */ +} + +/* remove PCM DAI configurations */ +static void remove_pcm_dai(struct snd_soc_component *comp, + struct snd_soc_dobj *dobj, int pass) +{ + if (pass != SOC_TPLG_PASS_PCM_DAI) + return; + + if (dobj->ops && dobj->ops->pcm_dai_unload) + dobj->ops->pcm_dai_unload(comp, dobj); + + list_del(&dobj->list); + kfree(dobj); +} + +/* bind a kcontrol to it's IO handlers */ +static int soc_tplg_kcontrol_bind_io(u32 io_type, struct snd_kcontrol_new *k, + const struct snd_soc_tplg_kcontrol_ops *ops, int num_ops, + const struct snd_soc_tplg_kcontrol_ops *bops, int num_bops) +{ + int i; + + /* try and map standard kcontrols handler first */ + for (i = 0; i < num_ops; i++) { + + if (SOC_CONTROL_GET_ID_PUT(ops[i].id) == + SOC_CONTROL_GET_ID_PUT(io_type) && ops[i].put) + k->put = ops[i].put; + if (SOC_CONTROL_GET_ID_GET(ops[i].id) == + SOC_CONTROL_GET_ID_GET(io_type) && ops[i].get) + k->get = ops[i].get; + if (SOC_CONTROL_GET_ID_INFO(ops[i].id) == + SOC_CONTROL_GET_ID_INFO(io_type) && ops[i].info) + k->info = ops[i].info; + } + + /* standard handlers found ? */ + if (k->put && k->get && k->info) + return 0; + + /* none found so try bespoke handlers */ + for (i = 0; i < num_bops; i++) { + + if (SOC_CONTROL_GET_ID_PUT(bops[i].id) == + SOC_CONTROL_GET_ID_PUT(io_type) && bops[i].put) + k->put = bops[i].put; + if (SOC_CONTROL_GET_ID_GET(bops[i].id) == + SOC_CONTROL_GET_ID_GET(io_type) && bops[i].get) + k->get = bops[i].get; + if (SOC_CONTROL_GET_ID_INFO(bops[i].id) == + SOC_CONTROL_GET_ID_INFO(io_type) && bops[i].info) + k->info = bops[i].info; + } + + /* bespoke handlers found ? */ + if (k->put && k->get && k->info) + return 0; + + /* nothing to bind */ + return -EINVAL; +} + +/* bind a widgets to it's evnt handlers */ +int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, + const struct snd_soc_tplg_widget_events *events, + int num_events, u16 event_type) +{ + int i; + + w->event = NULL; + + for (i = 0; i < num_events; i++) { + if (event_type == events[i].type) { + + /* found - so assign event */ + w->event = events[i].event_handler; + return 0; + } + } + + /* not found */ + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_bind_event); + +/* optionally pass new dynamic kcontrol to component driver. */ +static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, + struct snd_kcontrol_new *k, struct snd_soc_tplg_control_hdr *hdr) +{ + if (tplg->comp && tplg->ops && tplg->ops->control_load) + return tplg->ops->control_load(tplg->comp, k, hdr); + + dev_dbg(tplg->dev, "ASoC: no handler specified for kcontrol %s\n", + k->name); + return 0; +} + +static int soc_tplg_create_tlv(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc, u32 tlv_size) +{ + struct snd_soc_tplg_ctl_tlv *tplg_tlv; + struct snd_ctl_tlv *tlv; + + if (tlv_size == 0) + return 0; + + tplg_tlv = (struct snd_soc_tplg_ctl_tlv *) tplg->pos; + tplg->pos += tlv_size; + + tlv = kzalloc(sizeof(*tlv) + tlv_size, GFP_KERNEL); + if (tlv == NULL) + return -ENOMEM; + + dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", + tplg_tlv->numid, tplg_tlv->length); + tlv->numid = tplg_tlv->numid; + tlv->length = tplg_tlv->length; + memcpy(tlv->tlv, tplg_tlv + 1, tplg_tlv->length); + kc->tlv.p = (void *)tlv; + + return 0; +} + +static inline void soc_tplg_free_tlv(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc) +{ + kfree(kc->tlv.p); +} + +static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, + size_t size) +{ + struct snd_soc_tplg_bytes_ext *be; + struct soc_bytes_ext *sbe; + struct snd_kcontrol_new kc; + int i, err; + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_bytes_ext), count, + size, "mixer bytes")) { + dev_err(tplg->dev, "ASoC: Invalid count %d for byte control\n", + count); + return -EINVAL; + } + + for (i = 0; i < count; i++) { + be = (struct snd_soc_tplg_bytes_ext *)tplg->pos; + + /* validate kcontrol */ + if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + + sbe = kzalloc(sizeof(*sbe), GFP_KERNEL); + if (sbe == NULL) + return -ENOMEM; + + tplg->pos += (sizeof(struct snd_soc_tplg_bytes_ext) + + be->priv.length); + + dev_dbg(tplg->dev, + "ASoC: adding bytes kcontrol %s with access 0x%x\n", + be->hdr.name, be->hdr.access); + + memset(&kc, 0, sizeof(kc)); + kc.name = be->hdr.name; + kc.private_value = (long)sbe; + kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc.access = be->hdr.access; + + sbe->max = be->max; + sbe->dobj.type = SND_SOC_DOBJ_BYTES; + sbe->dobj.ops= tplg->ops; + INIT_LIST_HEAD(&sbe->dobj.list); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(be->hdr.index, &kc, io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, tplg->io_ops_count); + + if (err) { + soc_control_err(tplg, be->hdr.name, + be->hdr.index); + kfree(sbe); + continue; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc, + (struct snd_soc_tplg_control_hdr*)be); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + be->hdr.name); + kfree(sbe); + continue; + } + + /* register control here */ + err = soc_tplg_add_kcontrol(tplg, &kc, + &sbe->dobj.control.kcontrol); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to add %s\n", + be->hdr.name); + kfree(sbe); + continue; + } + + list_add(&sbe->dobj.list, &tplg->comp->dobj_list); + } + return 0; + +} +static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, + size_t size) +{ + struct snd_soc_tplg_mixer_control *mc; + struct soc_mixer_control *sm; + struct snd_kcontrol_new kc; + int i, err; + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_mixer_control), + count, size, "mixers")) { + + dev_err(tplg->dev, "ASoC: invalid count %d for controls\n", + count); + return -EINVAL; + } + + for (i = 0; i < count; i++) { + mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; + + /* validate kcontrol */ + if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + + sm = kzalloc(sizeof(*sm), GFP_KERNEL); + if (sm == NULL) + return -ENOMEM; + tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + + mc->priv.length); + + dev_dbg(tplg->dev, + "ASoC: adding mixer kcontrol %s with access 0x%x\n", + mc->hdr.name, mc->hdr.access); + + memset(&kc, 0, sizeof(kc)); + kc.name = mc->hdr.name; + kc.private_value = (long)sm; + kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc.access = mc->hdr.access; + + sm->reg = mc->reg; + sm->rreg = mc->rreg; + sm->shift = mc->shift; + sm->rshift = mc->rshift; + sm->max = mc->max; + sm->min = mc->min; + sm->invert = mc->invert; + sm->platform_max = mc->platform_max; + sm->dobj.id = tplg->id; + sm->dobj.ops= tplg->ops; + sm->dobj.type = SND_SOC_DOBJ_MIXER; + INIT_LIST_HEAD(&sm->dobj.list); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(mc->hdr.index, &kc, io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, tplg->io_ops_count); + if (err) { + soc_control_err(tplg, mc->hdr.name, + mc->hdr.index); + kfree(sm); + continue; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc, + (struct snd_soc_tplg_control_hdr*) mc); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + mc->hdr.name); + kfree(sm); + continue; + } + + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, mc->hdr.tlv_size); + + /* register control here */ + err = soc_tplg_add_kcontrol(tplg, &kc, + &sm->dobj.control.kcontrol); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to add %s\n", + mc->hdr.name); + soc_tplg_free_tlv(tplg, &kc); + kfree(sm); + continue; + } + + list_add(&sm->dobj.list, &tplg->comp->dobj_list); + } + + return 0; +} + +static int soc_tplg_denum_create_texts(struct soc_enum *se, + struct snd_soc_tplg_enum_control *ec) +{ + int i, ret; + + se->dobj.control.dtexts = + kzalloc(sizeof(char *) * ec->items, GFP_KERNEL); + if (se->dobj.control.dtexts == NULL) + return -ENOMEM; + + for (i = 0; i < ec->items; i++) { + + if (strnlen(ec->texts[i], SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) { + ret = -EINVAL; + goto err; + } + + se->dobj.control.dtexts[i] = kstrdup(ec->texts[i], GFP_KERNEL); + if (!se->dobj.control.dtexts[i]) { + ret = -ENOMEM; + goto err; + } + } + + return 0; + +err: + for (--i; i >= 0; i--) + kfree(se->dobj.control.dtexts[i]); + kfree(se->dobj.control.dtexts); + return ret; +} + +static int soc_tplg_denum_create_values(struct soc_enum *se, + struct snd_soc_tplg_enum_control *ec) +{ + if (ec->items > sizeof(*ec->values)) + return -EINVAL; + + se->dobj.control.dvalues = + kmalloc(ec->items * sizeof(u32), GFP_KERNEL); + if (!se->dobj.control.dvalues) + return -ENOMEM; + + memcpy(se->dobj.control.dvalues, ec->values, ec->items * sizeof(u32)); + return 0; +} + +static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, + size_t size) +{ + struct snd_soc_tplg_enum_control *ec; + struct soc_enum *se; + struct snd_kcontrol_new kc; + int i, ret, err; + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_enum_control), + count, size, "enums")) { + + dev_err(tplg->dev, "ASoC: invalid count %d for enum controls\n", + count); + return -EINVAL; + } + + for (i = 0; i < count; i++) { + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.length); + + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + + se = kzalloc((sizeof(*se)), GFP_KERNEL); + if (se == NULL) + return -ENOMEM; + + dev_dbg(tplg->dev, "ASoC: adding enum kcontrol %s size %d\n", + ec->hdr.name, ec->items); + + memset(&kc, 0, sizeof(kc)); + kc.name = ec->hdr.name; + kc.private_value = (long)se; + kc.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc.access = ec->hdr.access; + + se->reg = ec->reg; + se->shift_l = ec->shift_l; + se->shift_r = ec->shift_r; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.id = tplg->id; + se->dobj.type = SND_SOC_DOBJ_ENUM; + se->dobj.ops= tplg->ops; + INIT_LIST_HEAD(&se->dobj.list); + + switch (SOC_CONTROL_GET_ID_INFO(ec->hdr.index)) { + case SOC_DAPM_TYPE_ENUM_VALUE: + case SOC_CONTROL_TYPE_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, + "ASoC: could not create values for %s\n", + ec->hdr.name); + kfree(se); + continue; + } + /* fall through and create texts */ + case SOC_CONTROL_TYPE_ENUM: + case SOC_CONTROL_TYPE_ENUM_EXT: + case SOC_DAPM_TYPE_ENUM_EXT: + case SOC_DAPM_TYPE_ENUM_DOUBLE: + case SOC_DAPM_TYPE_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, + "ASoC: could not create texts for %s\n", + ec->hdr.name); + kfree(se); + continue; + } + break; + default: + dev_err(tplg->dev, + "ASoC: invalid enum control type %d for %s\n", + ec->hdr.index, ec->hdr.name); + kfree(se); + continue; + } + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(ec->hdr.index, &kc, io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, tplg->io_ops_count); + if (err) { + soc_control_err(tplg, ec->hdr.name, + ec->hdr.index); + kfree(se); + continue; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc, + (struct snd_soc_tplg_control_hdr*) ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + ec->hdr.name); + kfree(se); + continue; + } + + /* register control here */ + ret = soc_tplg_add_kcontrol(tplg, + &kc, &se->dobj.control.kcontrol); + if (ret < 0) { + dev_err(tplg->dev, "ASoC: could not add kcontrol %s\n", + ec->hdr.name); + kfree(se); + continue; + } + + list_add(&se->dobj.list, &tplg->comp->dobj_list); + } + + return 0; +} + +static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + struct snd_soc_tplg_elems *elems = + (struct snd_soc_tplg_elems *)tplg->pos; + struct snd_soc_tplg_control_hdr *control_hdr; + int i; + + if (tplg->pass != SOC_TPLG_PASS_MIXER) { + tplg->pos += sizeof(struct snd_soc_tplg_elems) + hdr->size; + return 0; + } + + tplg->pos += sizeof(struct snd_soc_tplg_elems); + + dev_dbg(tplg->dev, "ASoC: adding %d kcontrols\n", elems->count); + + for (i = 0; i < elems->count; i++) { + + control_hdr = (struct snd_soc_tplg_control_hdr *)tplg->pos; + + switch (SOC_CONTROL_GET_ID_INFO(control_hdr->index)) { + case SOC_CONTROL_TYPE_VOLSW: + case SOC_CONTROL_TYPE_VOLSW_EXT: + case SOC_CONTROL_TYPE_STROBE: + case SOC_CONTROL_TYPE_VOLSW_SX: + case SOC_CONTROL_TYPE_VOLSW_S8: + case SOC_CONTROL_TYPE_VOLSW_XR_SX: + case SOC_CONTROL_TYPE_BYTES: + case SOC_CONTROL_TYPE_BOOL_EXT: + case SOC_CONTROL_TYPE_RANGE: + case SOC_DAPM_TYPE_VOLSW: + case SOC_DAPM_TYPE_PIN: + soc_tplg_dmixer_create(tplg, 1, hdr->size); + break; + case SOC_CONTROL_TYPE_ENUM: + case SOC_CONTROL_TYPE_ENUM_EXT: + case SOC_CONTROL_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_DOUBLE: + case SOC_DAPM_TYPE_ENUM_VIRT: + case SOC_DAPM_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_EXT: + soc_tplg_denum_create(tplg, 1, hdr->size); + break; + case SOC_CONTROL_TYPE_BYTES_EXT: + soc_tplg_dbytes_create(tplg, 1, hdr->size); + break; + default: + soc_bind_err(tplg, control_hdr->index, elems->count); + } + } + + return 0; +} + +static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; + struct snd_soc_dapm_route route; + struct snd_soc_tplg_elems *elems = + (struct snd_soc_tplg_elems *)tplg->pos; + struct snd_soc_tplg_dapm_graph_elem *elem; + int count = elems->count, i; + + if (tplg->pass != SOC_TPLG_PASS_GRAPH) { + tplg->pos += sizeof(struct snd_soc_tplg_elems) + hdr->size; + return 0; + } + + tplg->pos += sizeof(struct snd_soc_tplg_elems); + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_dapm_graph_elem), + count, hdr->size, "graph")) { + + dev_err(tplg->dev, "ASoC: invalid count %d for DAPM routes\n", + count); + return -EINVAL; + } + + dev_dbg(tplg->dev, "ASoC: adding %d DAPM routes\n", count); + + for (i = 0; i < count; i++) { + elem = (struct snd_soc_tplg_dapm_graph_elem *)tplg->pos; + tplg->pos += sizeof(struct snd_soc_tplg_dapm_graph_elem); + + /* validate routes */ + if (strnlen(elem->source, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + if (strnlen(elem->sink, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + + route.source = elem->source; + route.sink = elem->sink; + route.connected = NULL; /* set to NULL atm for tplg users */ + if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0) + route.control = NULL; + else + route.control = elem->control; + + /* add route, but keep going if some fail */ + snd_soc_dapm_add_routes(dapm, &route, 1); + } + + return 0; +} + +static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( + struct soc_tplg *tplg, int num_kcontrols) +{ + struct snd_kcontrol_new *kc; + struct soc_mixer_control *sm; + struct snd_soc_tplg_mixer_control *mc; + int i, err; + + kc = kzalloc(sizeof(*kc) * num_kcontrols, GFP_KERNEL); + if (kc == NULL) + return NULL; + + for (i = 0; i < num_kcontrols; i++) { + mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; + sm = kzalloc(sizeof(*sm), GFP_KERNEL); + if (sm == NULL) + goto err; + + tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + + mc->priv.length); + + /* validate kcontrol */ + if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + goto err_str; + + dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", + mc->hdr.name, i); + + kc[i].name = mc->hdr.name; + kc[i].private_value = (long)sm; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = mc->hdr.access; + + sm->reg = mc->reg; + sm->rreg = mc->rreg; + sm->shift = mc->shift; + sm->rshift = mc->rshift; + sm->max = mc->max; + sm->min = mc->min; + sm->invert = mc->invert; + sm->platform_max = mc->platform_max; + sm->dobj.id = tplg->id; + INIT_LIST_HEAD(&sm->dobj.list); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(mc->hdr.index, &kc[i], io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, tplg->io_ops_count); + if (err) { + soc_control_err(tplg, mc->hdr.name, + mc->hdr.index); + kfree(sm); + continue; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_control_hdr*)mc); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + mc->hdr.name); + kfree(sm); + continue; + } + } + return kc; + +err_str: + kfree(sm); +err: + for (--i; i >= 0; i--) + kfree((void *)kc[i].private_value); + kfree(kc); + return NULL; +} + +static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( + struct soc_tplg *tplg) +{ + struct snd_kcontrol_new *kc; + struct snd_soc_tplg_enum_control *ec; + struct soc_enum *se; + int i, err; + + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + ec->priv.length); + + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return NULL; + + kc = kzalloc(sizeof(*kc), GFP_KERNEL); + if (kc == NULL) + return NULL; + + se = kzalloc(sizeof(*se), GFP_KERNEL); + if (se == NULL) + goto err; + + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); + + kc->name = ec->hdr.name; + kc->private_value = (long)se; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = ec->hdr.access; + + se->reg = ec->reg; + se->shift_l = ec->shift_l; + se->shift_r = ec->shift_r; + se->items = ec->items; + se->mask = ec->mask; + se->dobj.id = tplg->id; + + switch (SOC_CONTROL_GET_ID_INFO(ec->hdr.index)) { + case SOC_CONTROL_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_VALUE: + err = soc_tplg_denum_create_values(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + goto err_se; + } + /* fall through to create texts */ + case SOC_CONTROL_TYPE_ENUM: + case SOC_CONTROL_TYPE_ENUM_EXT: + case SOC_DAPM_TYPE_ENUM_EXT: + case SOC_DAPM_TYPE_ENUM_DOUBLE: + case SOC_DAPM_TYPE_ENUM_VIRT: + err = soc_tplg_denum_create_texts(se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", + ec->hdr.name); + goto err_se; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.index, ec->hdr.name); + goto err_se; + } + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(ec->hdr.index, kc, io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, tplg->io_ops_count); + if (err) { + soc_control_err(tplg, ec->hdr.name, + ec->hdr.index); + goto err_se; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_control_hdr*)ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + ec->hdr.name); + goto err_se; + } + + return kc; + +err_se: + /* free values and texts */ + kfree(se->dobj.control.dvalues); + for (i = 0; i < ec->items; i++) + kfree(se->dobj.control.dtexts[i]); + + kfree(se); +err: + kfree(kc); + + return NULL; +} + +static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( + struct soc_tplg *tplg, int count) +{ + struct snd_soc_tplg_bytes_ext *be; + struct soc_bytes_ext *sbe; + struct snd_kcontrol_new *kc; + int i, err; + + kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + if (!kc) + return NULL; + + for (i = 0; i < count; i++) { + be = (struct snd_soc_tplg_bytes_ext *)tplg->pos; + + /* validate kcontrol */ + if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + goto err; + + sbe = kzalloc(sizeof(*sbe), GFP_KERNEL); + if (sbe == NULL) + goto err; + + tplg->pos += (sizeof(struct snd_soc_tplg_bytes_ext) + + be->priv.length); + + dev_dbg(tplg->dev, + "ASoC: adding bytes kcontrol %s with access 0x%x\n", + be->hdr.name, be->hdr.access); + + memset(kc, 0, sizeof(*kc)); + kc[i].name = be->hdr.name; + kc[i].private_value = (long)sbe; + kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc[i].access = be->hdr.access; + + sbe->max = be->max; + INIT_LIST_HEAD(&sbe->dobj.list); + + /* map standard io handlers and check for external handlers */ + err = soc_tplg_kcontrol_bind_io(be->hdr.index, &kc[i], io_ops, + ARRAY_SIZE(io_ops), tplg->io_ops, + tplg->io_ops_count); + if (err) { + soc_control_err(tplg, be->hdr.name, + be->hdr.index); + kfree(sbe); + continue; + } + + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, &kc[i], + (struct snd_soc_tplg_control_hdr*)be); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + be->hdr.name); + kfree(sbe); + continue; + } + } + + return kc; + +err: + for (--i; i >= 0; i--) + kfree((void *)kc[i].private_value); + + kfree(kc); + return NULL; +} + +static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, + struct snd_soc_tplg_dapm_widget *w) +{ + struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; + struct snd_soc_dapm_widget template, *widget; + struct snd_soc_tplg_control_hdr *control_hdr; + int ret = 0; + + if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + if (strnlen(w->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; + + dev_dbg(tplg->dev, "ASoC: creating DAPM widget %s id %d\n", + w->name, w->id); + + memset(&template, 0, sizeof(template)); + template.id = w->id; + template.name = kstrdup(w->name, GFP_KERNEL); + if (!template.name) + return -ENOMEM; + template.sname = kstrdup(w->sname, GFP_KERNEL); + if (!template.sname) { + ret = -ENOMEM; + goto err; + } + template.reg = w->reg; + template.shift = w->shift; + template.mask = w->mask; + template.on_val = w->invert ? 0 : 1; + template.off_val = w->invert ? 1 : 0; + template.ignore_suspend = w->ignore_suspend; + template.event_flags = w->event_flags; + template.dobj.id = tplg->id; + + tplg->pos += + (sizeof(struct snd_soc_tplg_dapm_widget) + w->priv.length); + if (w->num_kcontrols == 0) { + template.num_kcontrols = 0; + goto widget; + } + + control_hdr = (struct snd_soc_tplg_control_hdr *)tplg->pos; + dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", + w->name, w->num_kcontrols, control_hdr->index); + + switch (SOC_CONTROL_GET_ID_INFO(control_hdr->index)) { + case SOC_CONTROL_TYPE_VOLSW: + case SOC_CONTROL_TYPE_STROBE: + case SOC_CONTROL_TYPE_VOLSW_SX: + case SOC_CONTROL_TYPE_VOLSW_S8: + case SOC_CONTROL_TYPE_VOLSW_XR_SX: + case SOC_CONTROL_TYPE_BYTES: + case SOC_CONTROL_TYPE_BOOL_EXT: + case SOC_CONTROL_TYPE_RANGE: + case SOC_DAPM_TYPE_VOLSW: + template.num_kcontrols = w->num_kcontrols; + template.kcontrol_news = soc_tplg_dapm_widget_dmixer_create(tplg, + template.num_kcontrols); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto hdr_err; + } + break; + case SOC_CONTROL_TYPE_ENUM: + case SOC_CONTROL_TYPE_ENUM_EXT: + case SOC_CONTROL_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_DOUBLE: + case SOC_DAPM_TYPE_ENUM_VIRT: + case SOC_DAPM_TYPE_ENUM_VALUE: + case SOC_DAPM_TYPE_ENUM_EXT: + template.dobj.widget.kcontrol_enum = 1; + template.num_kcontrols = 1; + template.kcontrol_news = soc_tplg_dapm_widget_denum_create(tplg); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto hdr_err; + } + break; + case SOC_CONTROL_TYPE_BYTES_EXT: + template.num_kcontrols = w->num_kcontrols; + template.kcontrol_news = + soc_tplg_dapm_widget_dbytes_create(tplg, + template.num_kcontrols); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto hdr_err; + } + break; + default: + dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", + SOC_CONTROL_GET_ID_GET(control_hdr->index), + SOC_CONTROL_GET_ID_PUT(control_hdr->index), + SOC_CONTROL_GET_ID_INFO(control_hdr->index)); + ret = -EINVAL; + goto hdr_err; + } + +widget: + ret = soc_tplg_widget_load(tplg, &template, w); + if (ret < 0) + goto hdr_err; + + widget = snd_soc_dapm_new_control(dapm, &template); + if (widget == NULL) { + dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", + w->name); + goto hdr_err; + } + + widget->dobj.type = SND_SOC_DOBJ_WIDGET; + widget->dobj.ops = tplg->ops; + widget->dobj.id = tplg->id; + list_add(&widget->dobj.list, &tplg->comp->dobj_list); + return 0; + +hdr_err: + kfree(template.sname); +err: + kfree(template.name); + return ret; +} + +static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + struct snd_soc_tplg_elems *elems = + (struct snd_soc_tplg_elems *)tplg->pos; + struct snd_soc_tplg_dapm_widget *widget; + int ret, count = elems->count, i; + + if (tplg->pass != SOC_TPLG_PASS_WIDGET) + return 0; + + tplg->pos += sizeof(struct snd_soc_tplg_elems); + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_dapm_graph_elem), count, + hdr->size, "widget")) { + + dev_err(tplg->dev, "ASoC: invalid count %d for widgets\n", + count); + return -EINVAL; + } + + dev_dbg(tplg->dev, "ASoC: adding %d DAPM widgets\n", count); + + for (i = 0; i < count; i++) { + widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + ret = soc_tplg_dapm_widget_create(tplg, widget); + if (ret < 0) + dev_err(tplg->dev, "ASoC: failed to load widget %s\n", + widget->name); + } + + return 0; +} + +static int soc_tplg_dapm_complete(struct soc_tplg *tplg) +{ + struct snd_soc_card *card = tplg->comp->card; + int ret; + + /* Card might not have been registered at this point. + * If so, just return success. + */ + if (!card) { + dev_warn(tplg->dev, "ASoC: Parent card not yet available," + "Do not add new widgets now\n"); + return 0; + } + + ret = snd_soc_dapm_new_widgets(card); + if (ret < 0) + dev_err(tplg->dev, "ASoC: failed to create new widgets %d\n", + ret); + + return ret; +} + +static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + struct snd_soc_tplg_elems *elems = + (struct snd_soc_tplg_elems *)tplg->pos; + struct snd_soc_tplg_pcm_dai *pcm_dai; + struct snd_soc_dobj *dobj; + int count = elems->count; + int ret; + + if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) + return 0; + + tplg->pos += sizeof(struct snd_soc_tplg_elems); + pcm_dai = (struct snd_soc_tplg_pcm_dai *)tplg->pos; + + if (soc_tplg_check_elem_count(tplg, + sizeof(struct snd_soc_tplg_pcm_dai), count, hdr->size, "PCM DAI")) { + dev_err(tplg->dev, "ASoC: invalid count %d for PCM DAI elems\n", + count); + return -EINVAL; + } + + dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); + tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) + count; + + dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL); + if (dobj == NULL) + return -ENOMEM; + + /* Call the platform driver call back to register the dais */ + ret = soc_tplg_pcm_dai_load(tplg, pcm_dai, count); + if (ret < 0) { + dev_err(tplg->comp->dev, "ASoC: PCM DAI loading failed\n"); + goto err; + } + + dobj->type = get_dobj_type(hdr, NULL); + dobj->pcm_dai.count = count; + dobj->pcm_dai.pd = pcm_dai; + dobj->ops = tplg->ops; + dobj->id = tplg->id; + list_add(&dobj->list, &tplg->comp->dobj_list); + return 0; + +err: + kfree(dobj); + return ret; +} + +static int soc_tplg_manifest_load(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + struct snd_soc_tplg_manifest *manifest; + + if (tplg->pass != SOC_TPLG_PASS_MANIFEST) + return 0; + + manifest = (struct snd_soc_tplg_manifest*)tplg->pos; + tplg->pos += sizeof(struct snd_soc_tplg_manifest); + + if (tplg->comp && tplg->ops && tplg->ops->manifest) + return tplg->ops->manifest(tplg->comp, manifest); + + dev_err(tplg->dev, "ASoC: Firmware manifest not supported\n"); + return 0; +} + +/* validate header magic, size and type */ +static int soc_valid_header(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) + return 0; + + /* big endian firmware objects not supported atm */ + if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { + dev_err(tplg->dev, + "ASoC: %s at pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n", + tplg->file, tplg->pass, hdr->magic, + soc_tplg_get_hdr_offset(tplg), tplg->fw->size); + return -EINVAL; + } + + if (hdr->magic != SND_SOC_TPLG_MAGIC) { + dev_err(tplg->dev, + "ASoC: %s at pass %d does not have a valid header got %x at offset 0x%lx size 0x%zx.\n", + tplg->file, tplg->pass, hdr->magic, + soc_tplg_get_hdr_offset(tplg), tplg->fw->size); + return -EINVAL; + } + + if (hdr->abi != SND_SOC_TPLG_ABI_VERSION) { + dev_err(tplg->dev, + "ASoC: %s at pass %d invalid ABI version got 0x%x need 0x%x at offset 0x%lx size 0x%zx.\n", + tplg->file, tplg->pass, hdr->abi, + SND_SOC_TPLG_ABI_VERSION, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + + if (hdr->size == 0) { + dev_err(tplg->dev, "ASoC: %s header has 0 size at offset 0x%lx.\n", + tplg->file, soc_tplg_get_hdr_offset(tplg)); + return -EINVAL; + } + + if (tplg->pass == hdr->type) + dev_dbg(tplg->dev, + "ASoC: Got 0x%x bytes of type %d version %d vendor %d at pass %d\n", + hdr->size, hdr->type, hdr->version, hdr->vendor_type, + tplg->pass); + + return 1; +} + +/* check header type and call appropriate handler */ +static int soc_tplg_load_header(struct soc_tplg *tplg, + struct snd_soc_tplg_hdr *hdr) +{ + tplg->pos = tplg->hdr_pos + sizeof(struct snd_soc_tplg_hdr); + + /* check for matching ID */ + if (hdr->id != tplg->req_id && hdr->id != SND_SOC_TPLG_ID_ALL) + return 0; + + tplg->id = hdr->id; + + switch (hdr->type) { + case SND_SOC_TPLG_MIXER: + return soc_tplg_kcontrol_elems_load(tplg, hdr); + case SND_SOC_TPLG_DAPM_GRAPH: + return soc_tplg_dapm_graph_elems_load(tplg, hdr); + case SND_SOC_TPLG_DAPM_WIDGET: + return soc_tplg_dapm_widget_elems_load(tplg, hdr); + case SND_SOC_TPLG_PCM: + case SND_SOC_TPLG_DAI_LINK: + case SND_SOC_TPLG_CODEC_LINK: + return soc_tplg_pcm_dai_elems_load(tplg, hdr); + case SND_SOC_TPLG_MANIFEST: + return soc_tplg_manifest_load(tplg, hdr); + default: + /* bespoke vendor data object */ + return soc_tplg_vendor_load(tplg, hdr); + } + + return 0; +} + +/* process the topology file headers */ +static int soc_tplg_process_headers(struct soc_tplg *tplg) +{ + struct snd_soc_tplg_hdr *hdr; + int ret; + + tplg->pass = SOC_TPLG_PASS_START; + + /* process the header types from start to end */ + while (tplg->pass <= SOC_TPLG_PASS_END) { + + tplg->hdr_pos = tplg->fw->data; + hdr = (struct snd_soc_tplg_hdr *)tplg->hdr_pos; + + while (!soc_tplg_is_eof(tplg)) { + + /* make sure header is valid before loading */ + ret = soc_valid_header(tplg, hdr); + if (ret < 0) + return ret; + else if (ret == 0) + break; + + /* load the header object */ + ret = soc_tplg_load_header(tplg, hdr); + if (ret < 0) + return ret; + + /* goto next header */ + tplg->hdr_pos += hdr->size + + sizeof(struct snd_soc_tplg_hdr); + hdr = (struct snd_soc_tplg_hdr *)tplg->hdr_pos; + } + + /* next data type pass */ + tplg->pass++; + } + + /* signal DAPM we are complete */ + ret = soc_tplg_dapm_complete(tplg); + if (ret < 0) + dev_err(tplg->dev, + "ASoC: failed to initialise DAPM from Firmware\n"); + + return ret; +} + +static int soc_tplg_load(struct soc_tplg *tplg) +{ + int ret; + + ret = soc_tplg_process_headers(tplg); + if (ret == 0) + soc_tplg_complete(tplg); + + return ret; +} + +/* load audio component topology from "firmware" file */ +int snd_soc_tplg_component_load(struct snd_soc_component *comp, + struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) +{ + struct soc_tplg tplg; + + /* setup parsing context */ + memset(&tplg, 0, sizeof(tplg)); + tplg.fw = fw; + tplg.dev = comp->dev; + tplg.comp = comp; + tplg.ops = ops; + tplg.req_id = id; + tplg.io_ops = ops->io_ops; + tplg.io_ops_count = ops->io_ops_count; + + return soc_tplg_load(&tplg); +} +EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); + +/* remove this dynamic widget */ +void snd_soc_tplg_widget_remove(struct snd_soc_dapm_widget *w) +{ + /* make sure we are a widget */ + if (w->dobj.type != SND_SOC_DOBJ_WIDGET) + return; + + remove_widget(w->dapm->component, &w->dobj, SOC_TPLG_PASS_WIDGET); +} +EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove); + +/* remove all dynamic widgets from this DAPM context */ +void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm, + u32 id) +{ + struct snd_soc_dapm_widget *w, *next_w; + struct snd_soc_dapm_path *p, *next_p; + + list_for_each_entry_safe(w, next_w, &dapm->card->widgets, list) { + + /* make sure we are a widget with correct context */ + if (w->dobj.type != SND_SOC_DOBJ_WIDGET || w->dapm != dapm) + continue; + + /* match ID */ + if (w->dobj.id != id && w->dobj.id != SND_SOC_TPLG_ID_ALL) + continue; + + list_del(&w->list); + + /* + * remove source and sink paths associated to this widget. + * While removing the path, remove reference to it from both + * source and sink widgets so that path is removed only once. + */ + list_for_each_entry_safe(p, next_p, &w->sources, list_sink) { + list_del(&p->list_sink); + list_del(&p->list_source); + list_del(&p->list); + kfree(p); + } + list_for_each_entry_safe(p, next_p, &w->sinks, list_source) { + list_del(&p->list_sink); + list_del(&p->list_source); + list_del(&p->list); + kfree(p); + } + /* check and free and dynamic widget kcontrols */ + snd_soc_tplg_widget_remove(w); + kfree(w->kcontrols); + kfree(w->name); + kfree(w); + } +} +EXPORT_SYMBOL_GPL(snd_soc_tplg_widget_remove_all); + +/* remove dynamic controls from the component driver */ +int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 id) +{ + struct snd_soc_dobj *dobj, *next_dobj; + int pass = SOC_TPLG_PASS_END; + + /* process the header types from end to start */ + while (pass >= SOC_TPLG_PASS_START) { + + /* remove mixer controls */ + list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, + list) { + + /* match ID */ + if (dobj->id != id && + dobj->id != SND_SOC_TPLG_ID_ALL) + continue; + + switch (dobj->type) { + case SND_SOC_DOBJ_MIXER: + remove_mixer(comp, dobj, pass); + break; + case SND_SOC_DOBJ_ENUM: + remove_enum(comp, dobj, pass); + break; + case SND_SOC_DOBJ_BYTES: + remove_bytes(comp, dobj, pass); + break; + case SND_SOC_DOBJ_WIDGET: + remove_widget(comp, dobj, pass); + break; + case SND_SOC_DOBJ_PCM: + case SND_SOC_DOBJ_DAI_LINK: + case SND_SOC_DOBJ_CODEC_LINK: + remove_pcm_dai(comp, dobj, pass); + break; + default: + dev_err(comp->dev, "ASoC: invalid component type %d for removal\n", + dobj->type); + break; + } + } + pass--; + } + + /* let caller know if FW can be freed when no objects are left */ + return !list_empty(&comp->dobj_list); +} +EXPORT_SYMBOL_GPL(snd_soc_tplg_component_remove); -- 2.1.0

1 0

[alsa-devel] [PATCH 00/16] HD-audio core controller split
by Takashi Iwai 16 Apr '15

16 Apr '15

Hi, this is a patch series for splitting the existing HD-audio controller code to a core library. The purose of this split is to help writing other HD-audio controller codes more easily. With these library codes, a controller driver can be written up with small skeleton codes. I have done quick tests, and no obvious regression found, so submitted here for reviews. The patches are found in topic/hda branch of sound git tree. Note that this series isn't targeted for 4.1 but for 4.2. Takashi === Jeeja KP (1): ALSA: hda - moved alloc/free stream pages function to controller library Takashi Iwai (15): ALSA: hda - Handle error from get_response bus ops directly ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - Add DSP loader to core library code ALSA: hda - Merge codec and controller helpers ALSA: hda - Move send_cmd / get_response to hdac_bus_ops ALSA: hda - Pass bus io_ops directly from the top-level driver ALSA: hda - Migrate hdac_stream into legacy driver ALSA: hda - Migrate more hdac_stream codes ALSA: hda - Embed bus into controller object ALSA: hda - Minor refactoring ALSA: hda - Move PCM format and rate handling code to core library ALSA: hda - Add missing inclusion of <linux/clocksource.h> ALSA: hda - Reenable tracepoints for controller ALSA: hda/tegra - Fix build error and warning ALSA: hda - Drop azx_sd_read*/write*() macros include/sound/hda_register.h | 152 +++++ include/sound/hdaudio.h | 282 ++++++++- sound/hda/Kconfig | 3 + sound/hda/Makefile | 2 +- sound/hda/hdac_bus.c | 20 +- sound/hda/hdac_controller.c | 507 +++++++++++++++ sound/hda/hdac_device.c | 300 +++++++++ sound/hda/hdac_stream.c | 648 ++++++++++++++++++++ sound/pci/hda/Kconfig | 3 - sound/pci/hda/Makefile | 3 +- sound/pci/hda/hda_codec.c | 409 +------------ sound/pci/hda/hda_codec.h | 31 +- sound/pci/hda/hda_controller.c | 1295 ++++++++------------------------------- sound/pci/hda/hda_controller.h | 269 +------- sound/pci/hda/hda_intel.c | 234 +++---- sound/pci/hda/hda_intel.h | 1 + sound/pci/hda/hda_intel_trace.h | 4 +- sound/pci/hda/hda_tegra.c | 76 +-- sound/pci/hda/patch_ca0132.c | 7 +- 19 files changed, 2395 insertions(+), 1851 deletions(-) create mode 100644 include/sound/hda_register.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 2.3.5

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[alsa-devel] [PATCH 0/2] i2c: rework last users of 2.4 style attach_adapter
by Wolfram Sang 16 Apr '15

16 Apr '15

The attach_adapter mechanism of the I2C framework is deprecated for years. There are two users left; drivers for old Macintosh computers. I got the idea of replacing this mechanism with a custom one with the help of deferred probing. Because I don't have the hardware, I modified the windtunnel driver to be runnable on my Renesas Lager board (ARM). This patch is not in this series but in the branch mentioned below. I first verified that the attach_adapter method was in deed used for bus scanning. When scanning, the driver rightfully failed on detecting the actual i2c clients because there is no such hardware on my board. However, the actual scanning happened. Using my custom mechanism, the same happens: all busses get scanned, then the clients cannot be detected. Since I didn't change the actual detection code, I assume that it should be working on those Macintosh devices as well. The keywest driver is only compile-tested. That being said, I'd be more than happy, if we could find someone willing to test these patches. Top-of-tree kernel would be nice, but these patches can easily be backported to probably the beginning of deferred probing. I will also notify some ppc lists of distributions of this thread for further help. If this works out, we can _finally_ get rid of this stone-age mechanism and clean up the i2c core. Changes since RFC: return -ENODEV instead of 0 when exiting the while loop (= nothing found on all busses) The branch I used for ARM compilation is here: git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/attach_adapter_removal_with_arm_compile-experimental Thanks, Wolfram Wolfram Sang (2): macintosh: therm_windtunnel: drop using attach adapter sound: ppc: keywest: drop using attach adapter drivers/macintosh/therm_windtunnel.c | 25 +++++++++++++++++++++++-- sound/ppc/keywest.c | 23 +++++++++++++++++++---- 2 files changed, 42 insertions(+), 6 deletions(-) -- 2.1.4

1 2

Re: [alsa-devel] [PATCH] ALSA: remove deprecated use of pci api
by Takashi Iwai 16 Apr '15

16 Apr '15

At Wed, 15 Apr 2015 16:10:17 +0200, Quentin Lambert wrote: > > Replace occurences of the pci api by appropriate call to the dma api. > > A simplified version of the semantic patch that finds this problem is as > follows: (http://coccinelle.lip6.fr) > > @deprecated@ > idexpression id; > position p; > @@ > > ( > pci_dma_supported@p ( id, ...) > | > pci_alloc_consistent@p ( id, ...) > ) > > @bad1@ > idexpression id; > position deprecated.p; > @@ > ...when != &id->dev > when != pci_get_drvdata ( id ) > when != pci_enable_device ( id ) > ( > pci_dma_supported@p ( id, ...) > | > pci_alloc_consistent@p ( id, ...) > ) > > @depends on !bad1@ > idexpression id; > expression direction; > position deprecated.p; > @@ > > ( > - pci_dma_supported@p ( id, > + dma_supported ( &id->dev, > ... > + , GFP_ATOMIC > ) > | > - pci_alloc_consistent@p ( id, > + dma_alloc_coherent ( &id->dev, > ... > + , GFP_ATOMIC > ) > ) > > Signed-off-by: Quentin Lambert <lambert.quentin(a)gmail.com> Thanks, applied to for-4.2 branch. Takashi

1 0

[alsa-devel] [patch 04/39] sound: Use hrtimer_resolution instead of hrtimer_get_res()
by Thomas Gleixner 16 Apr '15

16 Apr '15

No point in converting a timespec now that the value is directly accessible. Get rid of the null check while at it. Resolution is guaranteed to be > 0. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/core/hrtimer.c | 9 +-------- sound/drivers/pcsp/pcsp.c | 15 ++++++--------- 2 files changed, 7 insertions(+), 17 deletions(-) Index: tip/sound/core/hrtimer.c =================================================================== --- tip.orig/sound/core/hrtimer.c +++ tip/sound/core/hrtimer.c @@ -121,16 +121,9 @@ static struct snd_timer *mytimer; static int __init snd_hrtimer_init(void) { struct snd_timer *timer; - struct timespec tp; int err; - hrtimer_get_res(CLOCK_MONOTONIC, &tp); - if (tp.tv_sec > 0 || !tp.tv_nsec) { - pr_err("snd-hrtimer: Invalid resolution %u.%09u", - (unsigned)tp.tv_sec, (unsigned)tp.tv_nsec); - return -EINVAL; - } - resolution = tp.tv_nsec; + resolution = hrtimer_resolution; /* Create a new timer and set up the fields */ err = snd_timer_global_new("hrtimer", SNDRV_TIMER_GLOBAL_HRTIMER, Index: tip/sound/drivers/pcsp/pcsp.c =================================================================== --- tip.orig/sound/drivers/pcsp/pcsp.c +++ tip/sound/drivers/pcsp/pcsp.c @@ -42,16 +42,13 @@ struct snd_pcsp pcsp_chip; static int snd_pcsp_create(struct snd_card *card) { static struct snd_device_ops ops = { }; - struct timespec tp; - int err; - int div, min_div, order; - - hrtimer_get_res(CLOCK_MONOTONIC, &tp); + unsigned int resolution = hrtimer_resolution; + int err, div, min_div, order; if (!nopcm) { - if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + if (resolution > PCSP_MAX_PERIOD_NS) { printk(KERN_ERR "PCSP: Timer resolution is not sufficient " - "(%linS)\n", tp.tv_nsec); + "(%linS)\n", resolution); printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " "enabled.\n"); printk(KERN_ERR "PCSP: Turned into nopcm mode.\n"); @@ -59,13 +56,13 @@ static int snd_pcsp_create(struct snd_ca } } - if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) + if (loops_per_jiffy >= PCSP_MIN_LPJ && resolution <= PCSP_MIN_PERIOD_NS) min_div = MIN_DIV; else min_div = MAX_DIV; #if PCSP_DEBUG printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n", - loops_per_jiffy, min_div, tp.tv_nsec); + loops_per_jiffy, min_div, resolution); #endif div = MAX_DIV / min_div;

2 2

[alsa-devel] Separate input and output jacks for one UCM device?
by Tanu Kaskinen 16 Apr '15

16 Apr '15

Hi Liam and alsa-devel, My understanding is that a UCM device can represent a thing that has both input and output (I don't particularly like that, but it's too late to complain). How likely do you think that there are or there will be some drivers that expose separate input and output jack kcontrols for a headset jack, to differentiate between headphones/headset/microphone? My understanding is that jack kcontrols store only booleans, so there's no way to distinguish between headphones and a headset with just one kcontrol. The current UCM "spec" doesn't support specifying multiple kcontrols, since there's only one "JackControl" value. (Perhaps the "JackDev" value suffers from this problem too, but I don't know if jack input devices already support reporting the state separately for input and output.) -- Tanu

7 27

[alsa-devel] [PATCH] ALSA: intel8x0: Check pci_iomap() success for DEVICE_ALI
by Scott Wood 16 Apr '15

16 Apr '15

DEVICE_ALI previously would jump to port_inited after calling pci_iomap(), bypassing the check for bmaddr being NULL. Signed-off-by: Scott Wood <scottwood(a)freescale.com> --- sound/pci/intel8x0.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2c5484e..dfb77a5 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card, chip->bmaddr = pci_iomap(pci, 3, 0); else chip->bmaddr = pci_iomap(pci, 1, 0); + + port_inited: if (!chip->bmaddr) { dev_err(card->dev, "Controller space ioremap problem\n"); snd_intel8x0_free(chip); return -EIO; } - - port_inited: chip->bdbars_count = bdbars[device_type]; /* initialize offsets */ -- 2.1.0

2 1

[alsa-devel] [PATCH] ALSA: hda - simplify azx_has_pm_runtime
by David Henningsson 16 Apr '15

16 Apr '15

Because AZX_DCAPS_PM_RUNTIME is always defined as non-zero, the initial part of the expression can be skipped. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_controller.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index be1b7de..0efdb09 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -404,7 +404,7 @@ struct azx { ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) #define azx_has_pm_runtime(chip) \ - (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) + ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME) /* PCM setup */ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) -- 1.9.1

2 1

Re: [alsa-devel] [PATCH] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 16 Apr '15

16 Apr '15

在 2015年04月16日 10:41, Anish Kumar 写道: > 在 2015年04月16日 05:14, Anish Kumar 写道: >> 在 04/10/2015 03:38 AM, Anish Kumar 写道: >>>> To fix pop noise when shutdown,the pop noise during shutdown >>>> is the pmic cutoff power of codec without any notice. >>> Is the power turned off without any intimation to the codec driver? >>> Or is it physically removed? Can you describe the probelm >>> better? >>> When device is shutdown with playing music, the origin driver does nothing for shutdown operation, >>> as a result, pmic will cut off the power of codec IC, thus cause a pop noise from speaker and headphone. >>> So the patch added a shutdown callback for driver, enable volume smoothing and disable zero cross, >>> this is referenced from mic_input which does the samething before mic switch. >>> I'm very happy if some suggestion from maximintegrate . :-) >> Please let me know if the patch worked for you. >> Yes, the patch can work for me. > I hope this patch is what you tested. > > +static void max98090_i2c_shutdown(struct i2c_client *i2c) > +{ > + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); > + > + dev_info(&i2c->dev, "shut down device\n"); > + > + /* enable zero crossing */ > + regmap_write(max98090->regmap, > + M98090_REG_LEVEL_CONTROL, M98090_ZDENN_MASK-1); > + > + /* turn down volume levels for all the outputs */ > + regmap_write(max98090->regmap, > + M98090_REG_RCV_LOUTL_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_LOUTR_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_LEFT_SPK_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_RIGHT_SPK_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_LEFT_HP_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_RIGHT_HP_VOLUME, 0x00); > + regmap_write(max98090->regmap, > + M98090_REG_DEVICE_SHUTDOWN, 0x00); > + msleep(40); > +} > + > Please go ahead and submit to Mark. With the about is also useful to solve it. @zengxing, could you also help verify it? >> We had tested on chromiumos. >>> +static void max98090_i2c_shutdown(struct i2c_client *i2c) >>> +{ >>> + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); >>> + >>> + dev_info(&i2c->dev, "shut down device\n"); >>> + >>> + /* Enable volume smoothing, disable zero cross. This will cause >>> + * a quick 40ms ramp to mute on shutdown. >>> + */ >>> + regmap_write(max98090->regmap, >>> + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); >>> + regmap_write(max98090->regmap, >>> + M98090_REG_DEVICE_SHUTDOWN, 0x00); >>> + msleep(40); >>> +} >>> + >>> >>> >> >> >> > -- > ************************************************************************************** > 王晓腾 Caesar Wang > Product R&D Dept.III > Fuzhou Rockchip Electronics Co.Ltd > Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) > 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) > Tel：+86-591-83991906/07 - 8221 > Mobile:+86 15059456742 > E-mail : wxt(a)rock-chips.com > *************************************************************************************** > *************************************************************************************** > IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may > contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. > If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the > information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material > in its entirety in any format. Thank you. > *************************************************************************************** > -- ************************************************************************************** 王晓腾 Caesar Wang Product R&D Dept.III Fuzhou Rockchip Electronics Co.Ltd Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) Tel：+86-591-83991906/07 - 8221 Mobile:+86 15059456742 E-mail : wxt(a)rock-chips.com *************************************************************************************** *************************************************************************************** IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material in its entirety in any format. Thank you. ***************************************************************************************

1 0

Re: [alsa-devel] [PATCH] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 16 Apr '15

16 Apr '15

在 2015年04月16日 05:14, Anish Kumar 写道: > 在 04/10/2015 03:38 AM, Anish Kumar 写道: >>> To fix pop noise when shutdown,the pop noise during shutdown >>> is the pmic cutoff power of codec without any notice. >> Is the power turned off without any intimation to the codec driver? >> Or is it physically removed? Can you describe the probelm >> better? >> When device is shutdown with playing music, the origin driver does nothing for shutdown operation, >> as a result, pmic will cut off the power of codec IC, thus cause a pop noise from speaker and headphone. >> So the patch added a shutdown callback for driver, enable volume smoothing and disable zero cross, >> this is referenced from mic_input which does the samething before mic switch. >> I'm very happy if some suggestion from maximintegrate . :-) > Please let me know if the patch worked for you. Yes, the patch can work for me. We had tested on chromiumos. >> +static void max98090_i2c_shutdown(struct i2c_client *i2c) >> +{ >> + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); >> + >> + dev_info(&i2c->dev, "shut down device\n"); >> + >> + /* Enable volume smoothing, disable zero cross. This will cause >> + * a quick 40ms ramp to mute on shutdown. >> + */ >> + regmap_write(max98090->regmap, >> + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); >> + regmap_write(max98090->regmap, >> + M98090_REG_DEVICE_SHUTDOWN, 0x00); >> + msleep(40); >> +} >> + >> >> > > > > -- ************************************************************************************** 王晓腾 Caesar Wang Product R&D Dept.III Fuzhou Rockchip Electronics Co.Ltd Addr： NO.18 Building, A District, Fuzhou Software Park,Gulou District,Fuzhou, Fujian,China(Fuzhou Headquarters) 21F,Malata Building,Kejizhongyi Avenue,Nanshan District,Shenzhen (Shenzhen Office) Tel：+86-591-83991906/07 - 8221 Mobile:+86 15059456742 E-mail : wxt(a)rock-chips.com *************************************************************************************** *************************************************************************************** IMPORTANT NOTICE: This email is from Fuzhou Rockchip Electronics Co., Ltd .The contents of this email and any attachments may contain information that is privileged, confidential and/or exempt from disclosure under applicable law and relevant NDA. If you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution, or use of the information is STRICTLY PROHIBITED. Please immediately contact the sender as soon as possible and destroy the material in its entirety in any format. Thank you. ***************************************************************************************

1 0

Re: [alsa-devel] parallel, real time safe waiting?
by Johannes Lorenz 15 Apr '15

15 Apr '15

Ok, last question about this topic: Is blocked snd_pcm_reading/writing both a. real time safe and b. does not "consume 100 percent cpu power", i.e. waits silently? Thanks for the help.

2 1

[alsa-devel] [PATCH] ASoC: Intel: Remove invalid kfree of devm allocated data
by Jin Yao 15 Apr '15

15 Apr '15

kbuild robot reports following warning: "sound/soc/intel/haswell/sst-haswell-ipc.c:2204:1-6: WARNING: invalid free of devm_ allocated data" As julia explains to me, the memory allocated with devm_kalloc is freed automatically on failure of a probe function. So this kfree should be removed otherwise the double free will be got in error handler path. Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index 344a1e9..324eceb 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2201,7 +2201,6 @@ dma_err: dsp_new_err: sst_ipc_fini(ipc); ipc_init_err: - kfree(hsw); return ret; } EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: set dmaen->chan = NULL when error case
by Kuninori Morimoto 15 Apr '15

15 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_dmaen_quit() is assuming dmaen->chan is NULL if it failed to get DMAEngine channel. but, current dmaen->chan might have error value when error case (this driver is checking it by IS_ERR_OR_NULL()) This patch makes sure dmaen->chan is NULL when error case. Otherwise, it will contact to unknown address in rsnd_dmaen_quit() Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index ac3756f..144308f 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -156,6 +156,7 @@ static int rsnd_dmaen_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int id, (void *)id); } if (IS_ERR_OR_NULL(dmaen->chan)) { + dmaen->chan = NULL; dev_err(dev, "can't get dma channel\n"); goto rsnd_dma_channel_err; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: qcom: Return an error for invalid PCM trigger command
by Takashi Iwai 15 Apr '15

15 Apr '15

Fix a compile warning sound/soc/qcom/lpass-cpu.c: In function ‘lpass_cpu_daiops_trigger’: sound/soc/qcom/lpass-cpu.c:224:2: warning: ‘ret’ may be used uninitialized in this function [-Wmaybe-uninitialized] return ret; ^ Although switch () lists the most of existing cases, it's still better to cover the rest as an error properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 6698d058de29..dc790abaa331 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -194,7 +194,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; + int ret = -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: -- 2.3.5

3 2

[alsa-devel] [PATCH] ASoC: Add support for NAU8825 codec to ASoC
by Chih-Chiang Chang 15 Apr '15

15 Apr '15

The NAU88L25 is an ultra-low power high performance audio codec designed for smartphone, tablet PC, and other portable devices by Nuvoton, now add linux driver support for it. Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- include/sound/nau8825_plat.h | 22 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 593 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 150 +++++++++++ 5 files changed, 772 insertions(+) create mode 100644 include/sound/nau8825_plat.h create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/include/sound/nau8825_plat.h b/include/sound/nau8825_plat.h new file mode 100644 index 0000000..87afcd3 --- /dev/null +++ b/include/sound/nau8825_plat.h @@ -0,0 +1,22 @@ +/* + * linux/sound/snd_nau8825.h -- Platform data for NAU8825 + * + * Copyright 2015 Nuvoton Technology Corp. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_NAU8825_H +#define __LINUX_SND_NAU8825_H + +struct nau8825_platform_data { + unsigned int audio_mclk1; + unsigned int gpio_irq; + int naudint_irq; + int headset_detect; + int button_press_detect; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0bddd92..8807c57 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -464,6 +465,10 @@ config SND_SOC_MAX98357A config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8825 + tristate "Nuvoton NAU8825 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7acb6c1..acf594e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -252,6 +253,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..a8c8f59 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,593 @@ +/* + * linux/sound/soc/codecs/nau8825.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> +#include "nau8825.h" + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt); +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); +static int nau8825_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir); +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -5400, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, 2400, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -12800, 50, 0); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + + SOC_SINGLE_TLV("MIC Volume", NAU8825_ADC_DGAIN_CTRL, + NAU8825_ADC_DGAIN_SFT, + NAU8825_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_HSVOL_CTRL, + NAU8825_L_HSVOL_SFT, NAU8825_R_HSVOL_SFT, + NAU8825_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + SOC_DOUBLE("HP Playback Switch", NAU8825_HSVOL_CTRL, NAU8825_L_MUTE, + NAU8825_R_MUTE, 1, 1), + SOC_SINGLE("HP Class OP", NAU8825_CLASS_G_CTRL, NAU8825_CLASS_G_SHIFT, + 1, 0), + SOC_SINGLE("DAC Right", NAU8825_DAC_CTRL, NAU8825_DAC_R_SFT, 1, 0), + SOC_SINGLE("DAC Left", NAU8825_DAC_CTRL, NAU8825_DAC_L_SFT, 1, 0), + SOC_SINGLE("DAC Right Clock", NAU8825_DAC_CTRL, NAU8825_DAC_CLK_R_SFT, + 1, 0), + SOC_SINGLE("DAC Left Clock", NAU8825_DAC_CTRL, NAU8825_DAC_CLK_L_SFT, + 1, 0), +}; + +static const struct snd_kcontrol_new nau8825_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", NAU8825_HSVOL_CTRL, + NAU8825_L_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("HP R Switch", NAU8825_HSVOL_CTRL, + NAU8825_R_MUTE_SFT, 1, 1), +}; + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("Mic Jack"), + SND_SOC_DAPM_MICBIAS("MIC BIAS", NAU8825_BOOST, NAU8825_G_BIAS_SFT, 0), + SND_SOC_DAPM_SUPPLY("micbias", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU + | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("vmid", SND_SOC_NOPM, 0, 0, NULL, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + /* ADC IF1 */ + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC L1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_L_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC R1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_R_SFT, 0, NULL, + SND_SOC_DAPM_PRE_PMD), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8825_hpo_mix, + ARRAY_SIZE(nau8825_hpo_mix)), + SND_SOC_DAPM_PGA_S("HP amp", 1, NAU8825_CLASS_G_CTRL, + NAU8825_CLASS_G_SHIFT, 0, NULL, 0), + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + + {"ADC L", NULL, "Mic Jack"}, + {"ADC R", NULL, "Mic Jack"}, + {"AIF1TX", NULL, "ADC L"}, + {"AIF1TX", NULL, "ADC R"}, + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HP amp", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK, val_len); + return 0; +} + +static int nau8825_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) + snd_soc_update_bits(codec, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_EN); + else + snd_soc_update_bits(codec, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_DIS); + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg_val = 0; + + dev_dbg(codec->dev, " %s ###nau8825_set_dai_fmt %x\n", __func__, fmt); + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8825_I2S_BP_INV; + break; + default: + dev_alert(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8825_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8825_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8825_I2S_DF_PCM_B; + reg_val |= NAU8825_I2S_PCMB_EN; + break; + default: + dev_alert(codec->dev, "Invalid DAI I2S/PCM format\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK + | NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + reg_val); + return 0; +} + +static void config_fll_clk_12m(struct snd_soc_codec *codec) +{ + snd_soc_update_bits(codec, NAU8825_CLK_DIVIDER, 0x000F, 0x0003); + snd_soc_update_bits(codec, NAU8825_FL_1, 0x007F, 0x0001); + snd_soc_write(codec, NAU8825_FL_2, 0xC49B); + snd_soc_update_bits(codec, NAU8825_FL_3, 0x03FF, 0x0020); + snd_soc_update_bits(codec, NAU8825_FL_4, 0x0C00, 0x0800); + snd_soc_update_bits(codec, NAU8825_FL_5, 0x2000, 0x0000); + snd_soc_update_bits(codec, NAU8825_FL_6, 0x4000, 0x4000); +} + +void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + pr_debug("%s :: sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8825_INTERNALCLOCK: + snd_soc_write(codec, NAU8825_CLK_DIVIDER, 0x0853); + snd_soc_write(codec, NAU8825_FL_6, 0xE000); + snd_soc_write(codec, NAU8825_CLK_DIVIDER, 0x8853); + break; + case NAU8825_MCLK: + default: + snd_soc_write(codec, NAU8825_CLK_DIVIDER, 0x0853); + snd_soc_write(codec, NAU8825_FL_6, 0x6000); + snd_soc_write(codec, NAU8825_CLK_DIVIDER, 0x8853); + break; + } +} + +static int nau8825_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (clk_id) { + case NAU8825_MCLK: + config_fll_clk_12m(codec); + set_sys_clk(codec, clk_id); + break; + case NAU8825_INTERNALCLOCK: + set_sys_clk(codec, clk_id); + break; + default: + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8825_set_bias_level BIAS_ON\n"); + snd_soc_update_bits(codec, NAU8825_BIAS_ADJ, + NAU8825_VMID_MASK, NAU8825_VMID_EN); + snd_soc_update_bits(codec, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_EN); + break; + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8825_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8825_INTERNALCLOCK); + snd_soc_update_bits(codec, NAU8825_BIAS_ADJ, NAU8825_VMID_MASK, + NAU8825_VMID_DIS); + snd_soc_update_bits(codec, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8825_set_bias_level %d\n", level); + return 0; +} + +static int nau8825_codec_suspend(struct snd_soc_codec *codec) +{ + nau8825_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int nau8825_resume(struct snd_soc_codec *codec) +{ + nau8825_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static const struct reg_default nau8825_reg[] = { + /* software reset */ + {0x0000, 0x0001}, + {0x0000, 0x0002}, + /* enable clock source */ + {0x0001, 0x07FF}, + /* enable VMID and Bias */ + {0x0076, 0x3140}, + /* setup clock divider */ + {0x0003, 0x0050}, + /* jack detection configuration */ + {0x000D, 0x00E0}, + {0x000F, 0x0801}, + {0x0012, 0x0010}, + /* keypad detection configuration */ + {0x0013, 0x0280}, + {0x0014, 0x7310}, + {0x0015, 0x050E}, + {0x0016, 0x1B2A}, + /* audio format configuration */ + {0x001A, 0x0800}, + {0x001C, 0x000E}, + {0x001D, 0x0010}, + /* sampling rate control */ + {0x002B, 0x0012}, + /* DAC sampling rate control */ + {0x002C, 0x0082}, + /* ADC and DAC mixer control */ + {0x0030, 0x00D2}, + {0x0033, 0x00CF}, + {0x0034, 0x02CF}, + /* DAC class-G control */ + {0x006A, 0x1003}, + {0x0050, 0x2007}, + /* ADC PGA control */ + {0x0072, 0x0260}, + /* DAC power down enabled */ + {0x0080, 0x03A0}, + /* enable DAC clock */ + {0x0073, 0x336C}, + /* enable MIC Bias */ + {0x0074, 0x5502}, + /* powerup output driver */ + {0x007F, 0x473C}, + {0x007F, 0x473F}, + /* DAC power down disabled */ + {0x0080, 0x00A0}, + /* adjust MIC Bias */ + {0x0066, 0x2060}, + {0x0066, 0x2060}, + {0x0066, 0x0060}, +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + int i; + struct nau8825_priv *nau8825; + + nau8825 = snd_soc_codec_get_drvdata(codec); + nau8825->codec = codec; + /*writing initial register values to the codec*/ + for (i = 0; i < ARRAY_SIZE(nau8825_reg); i++) + snd_soc_write(codec, nau8825_reg[i].reg, nau8825_reg[i].def); + return 0; +} + +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + regmap_write(nau8825->regmap, NAU8825_RESET, 0); + regmap_write(nau8825->regmap, NAU8825_RESET, 1); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_nau8825 = { + .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, + .suspend = nau8825_codec_suspend, + .resume = nau8825_resume, + .set_bias_level = nau8825_set_bias_level, + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static bool nau8825_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_ENA_CTRL: + case NAU8825_CLK_EN: + case NAU8825_CLK_DIVIDER: + case NAU8825_FL_1: + case NAU8825_FL_2: + case NAU8825_FL_3: + case NAU8825_FL_4: + case NAU8825_FL_5: + case NAU8825_FL_6: + case NAU8825_HEADSET_CTRL: + case NAU8825_JACK_DET_CTRL: + case NAU8825_IRQ_SETTING_1: + case NAU8825_IRQ_STATUS: + case NAU8825_IRQ_CLEAR: + case NAU8825_IRQ_SETTING: + case NAU8825_SAR_ADC: + case NAU8825_VDET_COEFFICIENT: + case NAU8825_VDET_THRESHOLD_1: + case NAU8825_VDET_THRESHOLD_2: + case NAU8825_GPIO2_CTRL: + case NAU8825_I2S_PCM_CTRL_1: + case NAU8825_I2S_PCM_CTRL_2: + case NAU8825_ADC_RATE: + case NAU8825_DAC_CTRL1: + case NAU8825_DAC_CTRL2: + case NAU8825_ADC_DGAIN_CTRL: + case NAU8825_DAC_MUTE_CTRL: + case NAU8825_HSVOL_CTRL: + case NAU8825_DACL_CTRL: + case NAU8825_DACR_CTRL: + case NAU8825_SAR_ADC_OUTPUT: + case NAU8825_ANALOG_CTRL2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASS_G_CTRL: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_ENA_CTRL: + case NAU8825_CLK_EN: + case NAU8825_CLK_DIVIDER: + case NAU8825_FL_1: + case NAU8825_FL_2: + case NAU8825_FL_3: + case NAU8825_FL_4: + case NAU8825_FL_5: + case NAU8825_FL_6: + case NAU8825_HEADSET_CTRL: + case NAU8825_JACK_DET_CTRL: + case NAU8825_IRQ_SETTING_1: + case NAU8825_IRQ_STATUS: + case NAU8825_IRQ_CLEAR: + case NAU8825_IRQ_SETTING: + case NAU8825_SAR_ADC: + case NAU8825_VDET_COEFFICIENT: + case NAU8825_VDET_THRESHOLD_1: + case NAU8825_VDET_THRESHOLD_2: + case NAU8825_GPIO2_CTRL: + case NAU8825_I2S_PCM_CTRL_1: + case NAU8825_I2S_PCM_CTRL_2: + case NAU8825_ADC_RATE: + case NAU8825_DAC_CTRL1: + case NAU8825_DAC_CTRL2: + case NAU8825_ADC_DGAIN_CTRL: + case NAU8825_DAC_MUTE_CTRL: + case NAU8825_HSVOL_CTRL: + case NAU8825_DACL_CTRL: + case NAU8825_DACR_CTRL: + case NAU8825_SAR_ADC_OUTPUT: + case NAU8825_ANALOG_CTRL2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASS_G_CTRL: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP: + return true; + default: + return false; + } +} + +static const struct regmap_config nau8825_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = NAU8825_MAX_REGISTER, + .volatile_reg = nau8825_volatile_register, + .readable_reg = nau8825_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg), +}; + +static struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_sysclk = nau8825_dai_set_sysclk, + .set_fmt = nau8825_set_dai_fmt, + .digital_mute = nau8825_dac_mute, +}; + +static struct snd_soc_dai_driver nau8825_dai_driver[] = { + { + .name = "nau8825-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, + } +}; + + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct nau8825_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct nau8825_priv *nau8825; + int ret; + + nau8825 = devm_kzalloc(&i2c->dev, sizeof(struct nau8825_priv), + GFP_KERNEL); + if (nau8825 == NULL) + return -ENOMEM; + nau8825->i2c = i2c; + i2c_set_clientdata(i2c, nau8825); + if (pdata) + nau8825->pdata = *pdata; + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap); + if (IS_ERR(nau8825->regmap)) { + ret = PTR_ERR(nau8825->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + goto err_enable; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8825, + nau8825_dai_driver, + ARRAY_SIZE(nau8825_dai_driver)); +err_enable: + return ret; +} + +static int nau8825_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_id[] = { + {"nau8825", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_id); + +static const struct acpi_device_id nau8825_acpi_match[] = { + { "10508825", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match); + +static struct i2c_driver nau8825_i2c_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + }, + .probe = nau8825_i2c_probe, + .remove = (nau8825_i2c_remove), + .id_table = nau8825_i2c_id, +}; +module_i2c_driver(nau8825_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8825 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL v2"); + + diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..b16d4d1 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,150 @@ +/* + * linux/sound/soc/codecs/nau8825.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8825_H +#define _NAU8825_H + +#include <sound/nau8825_plat.h> + +#define NAU8825_RESET 0x00 +#define NAU8825_ENA_CTRL 0x01 +#define NAU8825_CLK_EN 0x02 +#define NAU8825_CLK_DIVIDER 0x03 +#define NAU8825_FL_1 0x04 +#define NAU8825_FL_2 0x05 +#define NAU8825_FL_3 0x06 +#define NAU8825_FL_4 0x07 +#define NAU8825_FL_5 0x08 +#define NAU8825_FL_6 0x09 +#define NAU8825_HEADSET_CTRL 0x0C +#define NAU8825_JACK_DET_CTRL 0x0D +#define NAU8825_IRQ_SETTING_1 0x0F +#define NAU8825_IRQ_STATUS 0x10 +#define NAU8825_IRQ_CLEAR 0x11 +#define NAU8825_IRQ_SETTING 0x12 +#define NAU8825_SAR_ADC 0x13 +#define NAU8825_VDET_COEFFICIENT 0x14 +#define NAU8825_VDET_THRESHOLD_1 0x15 +#define NAU8825_VDET_THRESHOLD_2 0x16 +#define NAU8825_GPIO2_CTRL 0x1A +#define NAU8825_I2S_PCM_CTRL_1 0x1C +#define NAU8825_I2S_PCM_CTRL_2 0x1D +#define NAU8825_ADC_RATE 0x2B +#define NAU8825_DAC_CTRL1 0x2C +#define NAU8825_DAC_CTRL2 0x2D +#define NAU8825_ADC_DGAIN_CTRL 0x30 +#define NAU8825_DAC_MUTE_CTRL 0x31 +#define NAU8825_HSVOL_CTRL 0x32 +#define NAU8825_DACL_CTRL 0x33 +#define NAU8825_DACR_CTRL 0x34 +#define NAU8825_CLASS_G_CTRL 0x50 +#define NAU8825_SAR_ADC_OUTPUT 0x59 +#define NAU8825_BIAS_ADJ 0x66 +#define NAU8825_ANALOG_CTRL2 0x6A +#define NAU8825_ANALOG_ADC_1 0x71 +#define NAU8825_ANALOG_ADC_2 0x72 +#define NAU8825_DAC_CTRL 0x73 +#define NAU8825_MIC_BIAS 0x74 +#define NAU8825_BOOST 0x76 +#define NAU8825_POWER_UP_CTRL 0x7F +#define NAU8825_CHARGE_BUMP 0x80 +#define NAU8825_MAX_REGISTER 0xFF + +/* reg. NAU8825_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8825_I2S_BP_MASK (0x1 << 7) +#define NAU8825_I2S_BP_INV (0x1 << 7) +#define NAU8825_I2S_PCMB_MASK (0x1 << 6) +#define NAU8825_I2S_PCMB_EN (0x1 << 6) +#define NAU8825_I2S_DL_MASK (0x3 << 2) +#define NAU8825_I2S_DF_MASK 0x3 +#define NAU8825_I2S_DF_RIGHT 0x0 +#define NAU8825_I2S_DF_LEFT 0x1 +#define NAU8825_I2S_DF_I2S 0x2 +#define NAU8825_I2S_DF_PCM_A 0x3 +#define NAU8825_I2S_DF_PCM_B 0x3 + +/* reg. NAU8825_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8825_I2S_MS_MASK (0x1 << 3) + +/* reg. NAU8825_ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DGAIN_SFT 0 + +/* reg. NAU8825_DAC_MUTE_CTRL (0x31) */ +#define NAU8825_SOFT_MUTE_MASK (0x1 << 13) +#define NAU8825_SOFT_MUTE_EN (0x1 << 13) +#define NAU8825_SOFT_MUTE_DIS (0x0 << 13) + +/* reg. NAU8825_HSVOL_CTRL (0x32) */ +#define NAU8825_R_MUTE (0x1 << 15) +#define NAU8825_R_MUTE_SFT 15 +#define NAU8825_L_MUTE (0x1 << 14) +#define NAU8825_L_MUTE_SFT 14 +#define NAU8825_L_HSVOL_SFT 6 +#define NAU8825_R_HSVOL_SFT 0 + +/* reg. NAU8825_CLASS_G_CTRL (0x50) */ +#define NAU8825_CLASS_G_RIGHT_SHIFT 2 +#define NAU8825_CLASS_G_LEFT_SHIFT 1 +#define NAU8825_CLASS_G_SHIFT 0 + +/* reg. NAU8825_BIAS_ADJ_2 (0x66) */ +#define NAU8825_VMID_MASK (0x1 << 6) +#define NAU8825_VMID_EN (0x1 << 6) +#define NAU8825_VMID_DIS (0x0 << 6) + +/* reg. NAU8825_DAC_CTRL (0x73) */ +#define NAU8825_DAC_R_SFT 13 +#define NAU8825_DAC_L_SFT 12 +#define NAU8825_DAC_CLK_R_SFT 9 +#define NAU8825_DAC_CLK_L_SFT 8 + +/* reg. NAU8825_BOOST (0x76) */ +#define NAU8825_G_BIAS_MASK (0x1 << 12) +#define NAU8825_G_BIAS_SFT 12 +#define NAU8825_G_BIAS_EN (0x1 << 12) +#define NAU8825_G_BIAS_DIS (0x0 << 12) + +/* Volume Rescale */ +#define NAU8825_VOL_RSCL_RANGE 0x36 +#define NAU8825_ADC_VOL_RSCL_RANGE 0x256 + +/* Data format */ +#define NAU8825_I2S_DL_16 (0x0 << 2) +#define NAU8825_I2S_DL_20 (0x1 << 2) +#define NAU8825_I2S_DL_24 (0x2 << 2) +#define NAU8825_I2S_DL_32 (0x3 << 2) + +enum { + NAU8825_INTERNALCLOCK = 0, + NAU8825_MCLK, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct nau8825_priv { + struct snd_soc_codec *codec; + struct nau8825_platform_data pdata; + struct regmap *regmap; + struct i2c_client *i2c; + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; + struct delayed_work delayed_work; + + struct workqueue_struct *workqueue; + struct mutex mutex; + unsigned int irq; + bool jd_status; + bool bp_status; + int jack_type; +}; +#endif /* _NAU8825_H */ -- 1.9.3 --- Avast 防毒軟體已檢查此封電子郵件的病毒。 http://www.avast.com

2 2

Re: [alsa-devel] parallel, real time safe waiting?
by Johannes Lorenz 15 Apr '15

15 Apr '15

> Of course. Not only does this page say so: > http://www.alsa-project.org/main/index.php/SMP_Design Ah, I read this, but that's probably just if two threads would share one handle: "The use of returned handles must be serialized in the application using own locking scheme." One more question: You said snd_pcm_wait() was real time safe. I guess that's also true for similar functions, like snd_pcm_htimestamp(), snd_pcm_readi(), ..., correct?

2 1

[alsa-devel] recording problem in beagleboard-mcbsp
by noman pouigt 15 Apr '15

15 Apr '15

Problem: not able to record with any of the devices in beagleboard-xm. Analysis: userspace is stuck in snd_pcm_readi function and kernel space i don't see snd_pcm_update_hw_ptr function being called. Checked the I2S clocks and they are perfect and recording data line is moving based on the data. I am able to do playback though. I am using below command to do recording. Do i need to add additional switches? Linux kernel: 3.19 root@arm:~# arecord -t wav -c 2 -r 44100 -f S32_LE -v test.wav Recording WAVE 'test.wav' : Signed 32 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Linear conversion PCM (S16_LE) Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 32 buffer_size : 27560 period_size : 5512 period_time : 124988 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 1 stop_threshold : 27560 silence_threshold: 0 silence_size : 0 boundary : 1806172160 Slave: Hardware PCM card 0 'omap3beagle' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 27560 period_size : 5512 period_time : 124988 tstamp_mode : NONE period_step : 1 avail_min : 5512 period_event : 0 start_threshold : 1 stop_threshold : 27560 silence_threshold: 0 silence_size : 0 boundary : 1806172160 appl_ptr : 0 hw_ptr : 0 [ 74.696319] omap-mcbsp 48074000.mcbsp: **** McBSP255 regs **** [ 74.696380] omap-mcbsp 48074000.mcbsp: DRR2: 0xedd0abce [ 74.696411] omap-mcbsp 48074000.mcbsp: DRR1: 0x0000 [ 74.696441] omap-mcbsp 48074000.mcbsp: DXR2: 0x0000 [ 74.696472] omap-mcbsp 48074000.mcbsp: DXR1: 0x0000 [ 74.696502] omap-mcbsp 48074000.mcbsp: SPCR2: 0x0230 [ 74.696533] omap-mcbsp 48074000.mcbsp: SPCR1: 0x0031 [ 74.696563] omap-mcbsp 48074000.mcbsp: RCR2: 0x8041 [ 74.696594] omap-mcbsp 48074000.mcbsp: RCR1: 0x0040 [ 74.696594] omap-mcbsp 48074000.mcbsp: XCR2: 0x8041 [ 74.696624] omap-mcbsp 48074000.mcbsp: XCR1: 0x0040 [ 74.696655] omap-mcbsp 48074000.mcbsp: SRGR2: 0x001f [ 74.696685] omap-mcbsp 48074000.mcbsp: SRGR1: 0x0f00 [ 74.696716] omap-mcbsp 48074000.mcbsp: PCR0: 0x000f setup: beagleoboard-xm ubuntu distribution arecord used jaroslav recording application also tried

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Re: [alsa-devel] Playback - Overwrite buffer with silence
by Nilhcraiv 15 Apr '15

15 Apr '15

Clemens wrote: It is impossible for the boundar to be zero. In my code snd_pcm_sw_params_get_boundary(swparams, &boundary) function does not return a negative value and the boundary variable is set to zero then, the boundar is zero. In any case, How I can put silence in buffer to avoid underrun? Do you know the answer? Thank you in advance.

3 4

Re: [alsa-devel] [PATCH 1/1] ASoC: fsl_sai: Add missing MODULE_DEVICE_TABLE
by Nicolin Chen 15 Apr '15

15 Apr '15

On Tue, Apr 14, 2015 at 11:33:27AM +0200, Alexander Stein wrote: > This enables module auto-loading when using device tree. > > Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > --- > sound/soc/fsl/fsl_sai.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > index ec79c3d..1fe9f5d 100644 > --- a/sound/soc/fsl/fsl_sai.c > +++ b/sound/soc/fsl/fsl_sai.c > @@ -673,6 +673,7 @@ static const struct of_device_id fsl_sai_ids[] = { > { .compatible = "fsl,imx6sx-sai", }, > { /* sentinel */ } > }; > +MODULE_DEVICE_TABLE(of, fsl_sai_ids); > > static struct platform_driver fsl_sai_driver = { > .probe = fsl_sai_probe, > -- > 2.0.5 >

1 0

[alsa-devel] alsa-announce list not used anymore?
by Tanu Kaskinen 14 Apr '15

14 Apr '15

Hi, I was thinking of subscribing to alsa-announce, but it seems that there have been no messages sent there in years. Is it ok if I remove the list from the wiki page at http://alsa-project.org/main/index.php/Mailing-lists ? Hmm... it seems that the wiki page has many other obsolete lists too... Is it so that only alsa-user and alsa-devel are relevant nowadays? -- Tanu

1 0

Re: [alsa-devel] parallel, real time safe waiting?
by Johannes Lorenz 14 Apr '15

14 Apr '15

> > 1.) Let each thread open a handle using snd_seq_open() > > 2.) Call snd_pcm_wait() in each thread > > > > Are these functions > > > > a) real time safe? > > snd_seq_open() is not, but why would that matter? > Don't you call it from the GUI thread? Ah, indeed, seq_open() is part of the initialization, so for this one, it does not matter. > > b) race-condition free, assuming I'll use pairwise different handles > > each thread > Yes. You're really 100 percent sure that this is guaranteed to work? Sorry for asking this :) I was expecting a "No, of course not.". Also, this will probably work for all handle-based-functions, as long as you keep using pairwise different handles? Regards, Johannes

2 1

[alsa-devel] parallel, real time safe waiting?
by Johannes Lorenz 14 Apr '15

14 Apr '15

Hello, I'd like to write a sequencer that can handle many synthesizers. Thus, I want them to run these synths parallel. Still, I'd like to keep my code real time safe, i.e. no malloc(), usleep(), ofstream, pthread mutexes etc. However, I wondered how to let all my threads sleep for x useconds if there's nothing to do for x useconds. My idea was: 1.) Let each thread open a handle using snd_seq_open() 2.) Call snd_pcm_wait() in each thread Are these functions a) real time safe? b) race-condition free, assuming I'll use pairwise different handles each thread Many thanks for any help. Kind regards, Johannes

2 1

[alsa-devel] [PATCH] ALSA: hda - not use regmap for vendor verb 0x781 of Intel HDMI codec
by mengdong.lin＠intel.com 14 Apr '15

14 Apr '15

From: Mengdong Lin <mengdong.lin(a)intel.com> For HDMI codec on Haswell/Broadwell/Skylake platforms, the vendor verb 0x781 is used enable DP 1.2 mode as a fixup if BIOS has not done this, in function intel_haswell_fixup_enable_dp12(). Otherwise, the display audio playback will be silent. Although the verb 0x781 is added to vendor verbs array, but snd_hdac_regmap_ encode_verb() will translate it to verb 0xf81 and cause regmap_write IO failure because 0xf81 is not in the vendor verb array and so will not be taken as a writable register. So this patch no longer uses regmap for verb 0x781 but directly send this command to enable DP1.2 mode. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ca0c05e..1849c94 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2298,8 +2298,7 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec) /* enable DP1.2 mode */ vendor_param |= INTEL_EN_DP12; - snd_hdac_regmap_add_vendor_verb(&codec->core, INTEL_SET_VENDOR_VERB); - snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0, + snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0, INTEL_SET_VENDOR_VERB, vendor_param); } -- 1.9.1

3 6

[alsa-devel] [RFC 00/11] ASoC: hda - Add ASoC Skylake HD audio driver
by Subhransu S. Prusty 14 Apr '15

14 Apr '15

This series of 11 patches adds support for ASoC Skylake HD Audio driver. Modified the hda core bus framework to support id_table match for ASoC DEV type instead of adding new wrapper. So dropped below RFC [RFC 3/3] ALSA: hda - add soc hda bus wrapper from the series [RFC 0/3] ALSA: Add soc hda bus support Jeeja KP (11): ALSA: hda - add id table support for hdac device/driver ALSA: hda - Add generic helper function to register/unregister driver ALSA: hda - add generic function to unregister all devices on bus ALSA: hda: corrected snd_hdac_stream_init() declaration ALSA: hda - exported link_reset enter/exit ALSA: hda - moved alloc/free stream pages function to controller library ALSA: hda - add generic functions to set hdac stream params ASoC: hda - add Skylake platform driver ALSA: hda - moved interrupt handler to controller library ASoC: hda - add Skylake HD audio driver ASoC: hda - enable ASoC Skylake HD audio driver include/sound/hdaudio.h | 43 ++- sound/hda/hda_bus_type.c | 2 + sound/hda/hdac_bus.c | 55 +++ sound/hda/hdac_controller.c | 78 ++++- sound/hda/hdac_stream.c | 29 ++ sound/soc/Kconfig | 3 +- sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 29 ++ sound/soc/hda/Makefile | 6 + sound/soc/hda/hda_skl.c | 829 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/hda/hda_skl.h | 49 +++ sound/soc/hda/hda_skl_pcm.c | 501 ++++++++++++++++++++++++++ 12 files changed, 1619 insertions(+), 6 deletions(-) create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/hda_skl.c create mode 100644 sound/soc/hda/hda_skl.h create mode 100644 sound/soc/hda/hda_skl_pcm.c -- 1.9.0

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[alsa-devel] [RFC 0/3] ALSA: Add soc hda bus support
by Vinod Koul 14 Apr '15

14 Apr '15

This series of 3 patches adds support for ASoC HDA bus. ASoC HDA bus is essentially a wrapper for HDA core bus and it manages ASoC hda drivers Ramesh Babu (3): ALSA: hda - remove assigning dev type inside core ALSA: hda - add ASoC device type for hda core ALSA: hda - add soc hda bus wrapper include/sound/hdaudio.h | 1 + include/sound/soc-hdac-bus.h | 103 +++++++++++++++++++ sound/hda/hdac_device.c | 1 - sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 4 + sound/soc/hda/Makefile | 3 + sound/soc/hda/soc-hdac-bus.c | 233 ++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 346 insertions(+), 1 deletion(-) create mode 100644 include/sound/soc-hdac-bus.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/soc-hdac-bus.c -- 1.7.9.5

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[alsa-devel] [GIT PULL] ASoC updates for v4.1
by Mark Brown 13 Apr '15

13 Apr '15

The following changes since commit f22e6e847115abc3a0e2ad7bb18d243d42275af1: Linux 4.0-rc7 (2015-04-06 15:39:45 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.1-2 for you to fetch changes up to 7667428f80526d908124e0647ac5dbe3dad88900: Merge remote-tracking branches 'asoc/topic/wm2200', 'asoc/topic/wm5100', 'asoc/topic/wm8731', 'asoc/topic/wm8804' and 'asoc/topic/wm8996' into asoc-next (2015-04-12 19:49:22 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.1 More updates for v4.1, pretty much all drivers: - Lots of cleanups from Lars, mainly moving things from the CODEC level to the card level. - Continuing improvements to rcar from Morimoto-san, pcm512x from Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han, and to rt5670 from Bard. - Support for some non-DSP Qualcomm platforms, Google's Storm platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC. ---------------------------------------------------------------- Alexandre Belloni (2): ASoC: atmel-pcm-pdc: merge atmel-pcm back in ASoC: atmel-pcm-dma: increase buffer_bytes_max Anish Kumar (2): ASoC: Add max98925 codec driver ASoC: max98925: trivial duplicate typo fix in set_fmt Arnd Bergmann (2): ASoC: omap: fix up SND_OMAP_SOC_OMAP_ABE_TWL6040 dependency ASoC: rnsd: fix build regression without CONFIG_OF Axel Lin (1): ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params Bard Liao (18): ASoC: rt5670: Add disabled item in dmic pin enum ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 ASoC: rt5670: Fix the speaker mono output issue ASoC: rt5670: Revert Keep sysclk on patch ASoC: rt5670: Check sysclk source by private data ASoC: rt5670: Add IRQ function ASoC: rt5670: export jack suspend/resume APIs ASoC: rt5645: Add TDM support for rt5650 ASoC: rt5645: Remove adc stereo2 filter ASoC: rt286: check regmap_read result for ID check ASoC: rt5645: Redefine format config for rt5650 ASoC: rt5645: Set use_single_rw flag for regmap ASoC: rt5645: Use update_bits for bit control ASoC: rt5645: Restore HP depop setting in HP off ASoC: rt286: Restore default in probe Caesar Wang (1): ASoC: max98090: add shutdown callback for max98090 Charles Keepax (12): ASoC: wm_adsp: Improve round to next 4-byte boundary ASoC: wm8804: Split out bus drivers ASoC: wm8804: Merge CODEC probe and bus probe regulator: core: Add devres versions of notifier registration ASoC: wm8804: Use new devres regulator_register_notifier ASoC: wm8804: Fix small issues in probe error paths ASoC: wm8804: Update DT binding document to cover regulator supplies ASoC: dapm: Fix build warning ASoC: wm5102: Remove set of volume update bits for output 3R ASoC: wm8804: Add support for hardware reset line ASoC: wm8804: Update binding documentation to include reset GPIO ASoC: wm8804: Enable runtime PM Dan Carpenter (1): ASoC: Intel: do cast earlier in sst_cdev_tstamp() Dmitry Torokhov (2): ASoC: cx20442: remove incorerct __exit markups ASoC: tlv320aic23: remove incorrect __exit markups Fabian Frederick (6): ASoC: fsl: constify of_device_id array ASoC: kirkwood: constify of_device_id array ASoC: rt5631: constify of_device_id array ASoC: ak4554: constify of_device_id array ASoC: fsi: constify of_device_id array ASoC: rsnd: constify of_device_id array Fabio Estevam (3): ASoC: fsl_ssi: Use devm_snd_soc_register_component() ASoC: fsl_ssi: Use devm_ioremap_resource() ASoC: fsl: Add the audio interface acronyms in Kconfig text Fang, Yang A (2): ALSA: Add params_set_format helper ASoC: Intel: fix machine driver warnings Geert Uytterhoeven (1): ASoC: rsnd: Use %pad to print dma_addr_t in rsnd_dmapp_init() Howard Mitchell (5): ALSA: ControlName.txt: Add 'Analogue' as a prefix for volume controls ASoC: pcm512x: Add 'Analogue' prefix to analogue volume controls ASoC: pcm512x: Fix divide by zero issue ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 Jarkko Nikula (2): ASoC: Intel: Remove support for Intel MID DMA from firmware loader ASoC: Intel: Remove vague commit about slave DMA config from firmware loader Jie Yang (9): ASoC: rt286: add jack detection disable with NULL jack passed ASoC: rt286: correct the OR to AND ASoC: Intel: Add suspend_pre and resume_post for Broadwell snd_soc_card ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in ASoC: Intel: Fix a buffer overflow issue Jin Yao (6): ASoC: Intel: move sysclk source setting to platform_clock_control for balance. ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card ASoC: Intel: move the jack creation to Braswell machine driver ASoC: Intel: Refactor common IPC/mailbox code into generic APIs ASoC: Intel: Use the generic IPC/mailbox APIs in Baytrail ASoC: Intel: Use the generic IPC/mailbox APIs in Broadwell Julia Lawall (1): ASoC: ab8500-codec: don't export static symbol Jyri Sarha (4): ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream Kenneth Westfield (13): ASoC: max98357a: Remove use of DRV_NAME ASoC: max98357a: Use standard DAI names ASoC: qcom: add LPASS header files ASoC: qcom: Add LPASS CPU DAI driver ASoC: qcom: Add LPASS platform driver ASoC: qcom: Document LPASS CPU bindings MAINTAINERS: Add QCOM audio ASoC maintainer ASoC: qcom: Change qcom,adsp in LPASS CPU bindings ASoC: qcom: Modify test for DSP in LPASS driver ASoC: qcom: Document Storm bindings ASoC: qcom: Add Storm machine driver ASoC: qcom: Add ability to build QCOM drivers ASoC: Allow for building QCOM drivers Krzysztof Kozlowski (2): ASoC: max98925: Constify regmap config and other codec data ASoC: rsnd: Fix duplicate const for DVC ramp rates Kuninori Morimoto (41): dmaengine: export symbol of of_dma_request_slave_channel() ASoC: rsnd: remove SH-DMA-BASE specific implementation ASoC: rsnd: remove un-needed parameter from rsnd_dma_init() ASoC: rsnd: remove unused rsnd_dma_available() ASoC: rsnd: remove un-needed parameter from rsnd_dma_quit() ASoC: rsnd: tidyup rsnd_dma_to_mod() macro declaration position ASoC: rsnd: enable to get resource by name ASoC: rsnd: add rsnd_gen_get_phy_addr() to get physical address ASoC: rsnd: add dma.c for Audio DMAC / Audio DMAC peri peri ASoC: rsnd: move rsnd_gen_dma_addr() from gen.c to dma.c ASoC: rsnd: enable to care 1st / 2nd DMAC on rsnd_dma_xxx() ASoC: rsnd: add Audio DMAC peri peri support rework ASoC: rsnd: remove rsnd_dma::addr ASoC: rsnd: remove rsnd_dma::dir ASoC: rsnd: use union with rsnd_dmaen / rsnd_dmapp ASoC: rsnd: enable rsnd_ssi_use_busif() for DMA ASoC: rsnd: ssi: add rsnd_ssi_of_node() ASoC: rsnd: src: add rsnd_src_of_node() ASoC: rsnd: dvc: add rsnd_dvc_of_node() ASoC: rsnd: 1st DMAC dma-names cares subnode ASoC: rsnd: add sample code of rcar_sound,src irq ASoC: rsnd: add sample code of missing clocks ASoC: rsnd: add sample code of dma entry ASoC: rsnd: add sample code of reg-names ASoC: rsnd: recover PIO mode for new dma interface ASoC: rsnd: add regmap_config::name for debugfs ASoC: ak4642: tidyup DAPM route for playback ASoC: rsnd: no more SSI restart when unusual situation ASoC: rsnd: no more SRC restart when unusual situation ASoC: rsnd: tidyup error message format ASoC: rsnd: show debug info for sampling rate convert ASoC: rsnd: add dai_link stream name ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card() ASoC: rsrc-card: add Renesas sampling rate convert sound card support ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate ASoC: rsnd: call clk_prepare/unprepare() in probe/remove ASoC: rsnd: remove useless debug message ASoC: rsnd: add DPCM based sampling rate convert ASoC: rsnd: add Synchronous SRC mode ASoC: ak4642: fixup channels_min ASoC: core: call snd_soc_runtime_set_dai_fmt() before soc_new_pcm() Lars-Peter Clausen (61): ASoC: sn95031: Pass CODEC to sn95031_jack_detection() ASoC: Allow to register jacks at the card level ASoC: simple-card: Register jacks at the card level ASoC: imx-es8328: Register jacks at the card level ASoC: wm1133-ev: Register jacks at the card level ASoC: broadwell: Register jacks at the card level ASoC: byt-max98090: Register jacks at the card level ASoC: cht_bsw_rt5645: Register jacks at the card level ASoC: mfld_machine: Register jacks at the card level ASoC: ams-deltea: Register jacks at the card level ASoC: omap-abe-twl6040: Register jacks at the card level ASoC: omap-twl4030: Register jacks at the card level ASoC: rx51: Register jacks at the card level ASoC: hx4700: Register jacks at the card level ASoC: palm27x: Register jacks at the card level ASoC: ttc-dkb: Register jacks at the card level ASoC: z2: Register jacks at the card level ASoC: h1980_uda1380: Register jacks at the card level ASoC: littlemill: Register jacks at the card level ASoC: lowland: Register jacks at the card level ASoC: rx1950_uda1380: Register jacks at the card level ASoC: smartq: Register jacks at the card level ASoC: speyside: Register jacks at the card level ASoC: tobermory: Register jacks at the card level ASoC: tegra_alc5632: Register jacks at the card level ASoC: tegra_max98090: Register jacks at the card level ASoC: tegra_rt5640: Register jacks at the card level ASoC: tegra_rt5677: Register jacks at the card level ASoC: tegra_wm8903: Register jacks at the card level ASoC: Remove snd_soc_jack_new() ASoC: Remove unnecessary device_remove_file() ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: Make snd_soc_dapm_kcontrol_codec() inline ASoC: nuc900: No need to track the dma buffer in the driver state struct ASoC: rt286: Drop unnecessary dapm bias_level initialization ASoC: wm2200: Drop unnecessary dapm bias_level initialization ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format ARM: shmobile: armadillo800eva: fix clock inversion ASoC: simple-card: Remove support for setting differing DAI formats ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue ASoC: wm8971: Integrate capacitor charging into the DAPM sequence ASoC: wm8971: Cleanup manual bias level transitions ASoC: wm8753: Integrate capacitor charging into the DAPM sequence ASoC: wm8753: Cleanup manual bias level transitions ASoC: Remove suspend_bias_level from DAPM context struct ASoC: wm8350: Move delayed work struct from DAPM context to driver state ASoC: dapm: Remove delayed_work from dapm context struct ASoC: tegra_alc5632: Use card DAPM context to access widgets ASoC: tegra_rt5677: Use card DAPM context to access widgets ASoC: tegra_wm8903: Use card DAPM context to access widgets ASoC: tegra_wm9712: Use card DAPM context to access widgets ASoC: Create card debugfs directory earlier ASoC: Make soc_dpcm_debugfs_add() non-fatal ASoC: Don't try to register debugfs entries if the parent does not exist ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary ASoC: atmel: Improve machine driver compile test coverage ASoC: wm1133-ev1: Use card DAPM context to access widgets ASoC: mop500_ab8500: Use card DAPM context to access widgets ASoC: davinci-evm: Use card DAPM context to access widgets ASoC: n810: Consistently pass the card DAPM context to n810_ext_control() ASoC: n810: Automatically disconnect non-connected pins Lu, Han (8): ASoC: Intel: add function to load firmware image ASoC: Intel: add function to load sound effect module waves ASoC: Intel: add function to enable/disable sound effect module waves ASoC: Intel: add kcontrol to enable/disable sound effect module waves ASoC: Intel: add function to set parameter to sound effect module waves ASoC: Intel: add kcontrol to set parameter to sound effect module waves Intel: ASoC: Add condition check before set param to waves ASoC: Intel: fix warning reported by static check tool smatch Manish Badarkhe (1): ASoC: davinci-evm: drop un-necessary remove function Mark Brown (38): Merge branch 'fix/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-rt5677 Merge tag 'asoc-fix-v4.0-rc2' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm-dt' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/davinci', 'asoc/topic/gpiod' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/jack', 'asoc/topic/max98357a', 'asoc/topic/omap' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5670', 'asoc/topic/rt5677', 'asoc/topic/simple', 'asoc/topic/tegra' and 'asoc/topic/wm-adsp' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8804' into asoc-next Merge branch 'topic/notifier' of git://git.kernel.org/.../broonie/regulator into asoc-wm8804 Merge tag 'v4.0-rc3' into asoc-rt5670 Merge tag 'v4.0-rc3' into asoc-intel Merge branch 'fix/intel' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/rt5670' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/jack' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'asoc-fix-rcar' into HEAD Merge branch 'topic/jack' of git://git.kernel.org/.../broonie/sound into asoc-tegra Merge tag 'v4.0-rc7' into asoc-core Merge branch 'topic/delay' of git://git.kernel.org/.../broonie/sound into asoc-core Merge branch 'topic/dapm-dt' of git://git.kernel.org/.../broonie/sound into asoc-core Merge tag 'asoc-fix-v4.0-rc7' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm512x' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4642', 'asoc/fix/atmel', 'asoc/fix/cs4271', 'asoc/fix/davinci' and 'asoc/fix/rcar' into asoc-linus Merge remote-tracking branch 'asoc/fix/wm8741' into asoc-linus Merge tag 'asoc-v4.1' into asoc-next Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/4554', 'asoc/topic/ab8500', 'asoc/topic/ak4642', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/cx20442' and 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/dmaengine', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branches 'asoc/topic/jack', 'asoc/topic/jz4740' and 'asoc/topic/kirkwood' into asoc-next Merge remote-tracking branches 'asoc/topic/link-param', 'asoc/topic/max98090', 'asoc/topic/max98925' and 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5641', 'asoc/topic/rt5677' and 'asoc/topic/sh-cleanup' into asoc-next Merge remote-tracking branches 'asoc/topic/tegra', 'asoc/topic/tlv320aic23' and 'asoc/topic/ux500' into asoc-next Merge remote-tracking branches 'asoc/topic/wm2200', 'asoc/topic/wm5100', 'asoc/topic/wm8731', 'asoc/topic/wm8804' and 'asoc/topic/wm8996' into asoc-next Mengdong Lin (1): ASoC: rt5670: add API to select ASRC clock source Nicholas Mc Guire (5): ASoC: wm8996: match wait_for_completion_timeout return type ASoC: wm8996: ensure lower bounds of 1 for timeout ASoC: arizona: match wait_for_completion_timeout return type ASoC: wm5100: match wait_for_completion_timeout return type ASoC: wm2200: match wait_for_completion_timeout return type Nicolas Ferre (1): MAINTAINERS: change the Atmel audio alsa driver entry Nicolin Chen (2): ASoC: core: Add extra dapm properties for Device Tree ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing() Nikesh Oswal (2): ASoC: wm_adsp: Ensure DSP controls are always persistent ASoC: dapm: add code to configure dai link parameters Oder Chiou (3): ASoC: rt5677: Add the chip type to distinguish the setting of the clock source ASoC: rt5677: Keep the LDO2 powered while used in the suspend mode ASoC: rt5677: add API to select ASRC clock source Pascal Huerst (1): ASoC: cs4271: Increase delay time after reset Peter Rosin (3): ASoC: atmel_ssc_dai: Allow more rates ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP Peter Ujfalusi (10): ASoC: omap: Kconfig: Support for omap5-uevm analog audio ASoC: davinci-mcasp: Add support for CBS_CFM mode ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled ASoC: davinci-i2s: Convert to use edma-pcm ASoC: davinci-vcif: Convert to use edma-pcm ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm ASoC: davinci: Remove unused davinci-pcm platform driver ASoC: davinci-mcasp: Allow complete shutdown of McASP when not in use ASoC: davinci-mcasp: Fix compilation error ASoC: davinci-mcasp: Fix ruledata setup in davinci_mcasp_startup Pierre-Louis Bossart (1): ASoC: Intel: remove misleading DMA error messages on Baytrail platforms Sapthagiri Baratam (1): ASoC: wm8804: Add DAPM widgets for SPDIF/AIF Sergej Sawazki (1): ASoC: wm8741: Fix rates constraints values Songjun Wu (1): ASoC: wm8731: let codec to manage clock by itself Subhransu S. Prusty (3): ASoC: Intel: Add memcpy32_fromio as well ASoC: Intel: Remove ignore suspend support ASoC: Intel: Remove soc pm handling to allow platform driver handle it Sudip Mukherjee (1): ASoC: Intel: remove unused functions Takeshi Kihara (1): ASoC: ak4642: enable stereo line output power-save mode Tomeu Vizoso (3): ASoC: tegra: Expose Headphones pin to userspace ASoC: tegra: Add sink for the internal mic to tegra_max98090 ASoC: tegra: Add control for the Mic Jack pin Uwe Kleine-König (5): ASoC: improve usage of gpiod API ASoC: fsi: fix license specification ASoC: fsi: reorder code to make a forward declaration superfluous ASoC: fsi: mark several data structures as const ASoC: rcar: mark device data as constant Vinod Koul (10): ASoC: core: allow pcms to be registered as nonatomic ASoC: Intel: mark cht machine driver with nonatomic trigger ASoC: Intel: update MMX ID to 3 ASoC: Intel: add support for pause and resume in sst ASoC: Intel: add support for pcm stream suspend/resume ASoC: Intel: add support for platform suspend ASoC: Intel: add pm support in sst ipc driver ASoC: Intel: Move the fw download to power_control ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM ASoC: Intel: acpi_probe: fix error return path Vishal Thanki (1): ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of Zubair Lutfullah Kakakhel (1): ASoC: jz4740: Add jz4780 support kbuild test robot (4): ASoC: tegra: fix platform_no_drv_owner.cocci warnings ASoC: max98925_spk_tlv can be static ASoC: Intel: read_shim_data() can be static ASoC: Intel: fix array_size.cocci warnings .../bindings/sound/ingenic,jz4740-i2s.txt | 2 +- .../devicetree/bindings/sound/max98925.txt | 22 + .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 43 + .../devicetree/bindings/sound/renesas,rsnd.txt | 125 ++- .../bindings/sound/renesas,rsrc-card.txt | 67 ++ Documentation/devicetree/bindings/sound/storm.txt | 23 + Documentation/devicetree/bindings/sound/wm8804.txt | 7 + Documentation/sound/alsa/ControlNames.txt | 10 +- MAINTAINERS | 9 +- arch/arm/mach-shmobile/board-armadillo800eva.c | 3 +- drivers/dma/of-dma.c | 1 + drivers/regulator/devres.c | 85 ++ include/linux/regulator/consumer.h | 16 + include/sound/pcm_params.h | 7 + include/sound/rt5670.h | 1 + include/sound/simple_card.h | 1 - include/sound/soc-dapm.h | 6 +- include/sound/soc-dpcm.h | 2 +- include/sound/soc.h | 30 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 9 +- sound/soc/atmel/Makefile | 2 - sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 79 ++ sound/soc/atmel/atmel-pcm.c | 121 --- sound/soc/atmel/atmel-pcm.h | 5 - sound/soc/atmel/atmel_ssc_dai.c | 111 ++- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/codecs/Kconfig | 22 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 2 - sound/soc/codecs/adau1977.c | 17 +- sound/soc/codecs/ak4554.c | 2 +- sound/soc/codecs/ak4642.c | 41 +- sound/soc/codecs/arizona.c | 6 +- sound/soc/codecs/cs35l32.c | 19 +- sound/soc/codecs/cs4265.c | 19 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cx20442.c | 4 +- sound/soc/codecs/max98090.c | 17 + sound/soc/codecs/max98357a.c | 11 +- sound/soc/codecs/max98925.c | 655 ++++++++++++++++ sound/soc/codecs/max98925.h | 832 ++++++++++++++++++++ sound/soc/codecs/pcm512x.c | 205 ++++- sound/soc/codecs/rt286.c | 40 +- sound/soc/codecs/rt5631.c | 2 +- sound/soc/codecs/rt5645.c | 90 ++- sound/soc/codecs/rt5645.h | 2 - sound/soc/codecs/rt5670.c | 213 ++++- sound/soc/codecs/rt5670.h | 10 + sound/soc/codecs/rt5677.c | 207 ++++- sound/soc/codecs/rt5677.h | 85 ++ sound/soc/codecs/sn95031.c | 14 +- sound/soc/codecs/sn95031.h | 3 +- sound/soc/codecs/sta350.c | 30 +- sound/soc/codecs/tas2552.c | 13 +- sound/soc/codecs/tlv320aic23-i2c.c | 4 +- sound/soc/codecs/wm2200.c | 9 +- sound/soc/codecs/wm5100.c | 7 +- sound/soc/codecs/wm5102.c | 1 - sound/soc/codecs/wm8350.c | 25 +- sound/soc/codecs/wm8731.c | 34 + sound/soc/codecs/wm8741.c | 8 +- sound/soc/codecs/wm8753.c | 73 +- sound/soc/codecs/wm8804-i2c.c | 65 ++ sound/soc/codecs/wm8804-spi.c | 57 ++ sound/soc/codecs/wm8804.c | 534 ++++++------- sound/soc/codecs/wm8804.h | 8 + sound/soc/codecs/wm8971.c | 99 +-- sound/soc/codecs/wm8996.c | 12 +- sound/soc/codecs/wm_adsp.c | 13 +- sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-evm.c | 17 +- sound/soc/davinci/davinci-i2s.c | 67 +- sound/soc/davinci/davinci-mcasp.c | 335 +++++--- sound/soc/davinci/davinci-pcm.c | 861 --------------------- sound/soc/davinci/davinci-pcm.h | 41 - sound/soc/davinci/davinci-vcif.c | 55 +- sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl-asoc-card.c | 6 + sound/soc/fsl/fsl_ssi.c | 32 +- sound/soc/fsl/imx-es8328.c | 6 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 15 +- sound/soc/generic/simple-card.c | 50 +- sound/soc/intel/Makefile | 42 +- sound/soc/intel/atom/Makefile | 7 + sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 60 ++ sound/soc/intel/{ => atom}/sst-mfld-platform.h | 1 + sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 130 +++- sound/soc/intel/{ => atom}/sst/sst.h | 12 + sound/soc/intel/{ => atom}/sst/sst_acpi.c | 4 +- sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 69 +- sound/soc/intel/{ => atom}/sst/sst_ipc.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_loader.c | 12 +- sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 26 +- sound/soc/intel/{ => atom}/sst/sst_stream.c | 2 +- sound/soc/intel/baytrail/Makefile | 4 + sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 364 ++------- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +- sound/soc/intel/boards/Makefile | 15 + sound/soc/intel/{ => boards}/broadwell.c | 50 +- sound/soc/intel/{ => boards}/byt-max98090.c | 13 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 8 +- sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 20 +- sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 108 ++- sound/soc/intel/{ => boards}/haswell.c | 10 +- sound/soc/intel/{ => boards}/mfld_machine.c | 24 +- sound/soc/intel/common/Makefile | 7 + sound/soc/intel/{ => common}/sst-acpi.c | 1 + sound/soc/intel/{ => common}/sst-dsp-priv.h | 13 + sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 2 +- sound/soc/intel/{ => common}/sst-firmware.c | 10 +- sound/soc/intel/common/sst-ipc.c | 294 +++++++ sound/soc/intel/common/sst-ipc.h | 91 +++ sound/soc/intel/haswell/Makefile | 4 + sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 9 +- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 794 ++++++++++++------- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 53 ++ sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 140 +++- sound/soc/jz4740/jz4740-i2s.c | 84 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/nuc900/nuc900-audio.h | 3 - sound/soc/nuc900/nuc900-pcm.c | 31 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 4 +- sound/soc/omap/n810.c | 23 +- sound/soc/omap/omap-abe-twl6040.c | 10 +- sound/soc/omap/omap-hdmi-audio.c | 2 - sound/soc/omap/omap-pcm.c | 21 +- sound/soc/omap/omap-twl4030.c | 12 +- sound/soc/omap/rx51.c | 6 +- sound/soc/pxa/hx4700.c | 11 +- sound/soc/pxa/palm27x.c | 11 +- sound/soc/pxa/ttc-dkb.c | 15 +- sound/soc/pxa/z2.c | 10 +- sound/soc/qcom/Kconfig | 25 + sound/soc/qcom/Makefile | 11 + sound/soc/qcom/lpass-cpu.c | 491 ++++++++++++ sound/soc/qcom/lpass-lpaif-ipq806x.h | 172 ++++ sound/soc/qcom/lpass-platform.c | 526 +++++++++++++ sound/soc/qcom/lpass.h | 51 ++ sound/soc/qcom/storm.c | 162 ++++ sound/soc/samsung/h1940_uda1380.c | 9 +- sound/soc/samsung/littlemill.c | 12 +- sound/soc/samsung/lowland.c | 14 +- sound/soc/samsung/rx1950_uda1380.c | 9 +- sound/soc/samsung/smartq_wm8987.c | 11 +- sound/soc/samsung/speyside.c | 14 +- sound/soc/samsung/tobermory.c | 13 +- sound/soc/sh/Kconfig | 6 + sound/soc/sh/fsi.c | 71 +- sound/soc/sh/rcar/Makefile | 7 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/core.c | 278 ++----- sound/soc/sh/rcar/dma.c | 616 +++++++++++++++ sound/soc/sh/rcar/dvc.c | 45 +- sound/soc/sh/rcar/gen.c | 152 +--- sound/soc/sh/rcar/rsnd.h | 92 ++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++ sound/soc/sh/rcar/src.c | 250 ++++-- sound/soc/sh/rcar/ssi.c | 73 +- sound/soc/soc-core.c | 124 +-- sound/soc/soc-dapm.c | 182 ++++- sound/soc/soc-jack.c | 42 +- sound/soc/soc-pcm.c | 17 +- sound/soc/tegra/tegra_alc5632.c | 14 +- sound/soc/tegra/tegra_max98090.c | 26 +- sound/soc/tegra/tegra_rt5640.c | 10 +- sound/soc/tegra/tegra_rt5677.c | 20 +- sound/soc/tegra/tegra_wm8903.c | 21 +- sound/soc/tegra/tegra_wm9712.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 36 +- 188 files changed, 8983 insertions(+), 3633 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/max98925.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt create mode 100644 Documentation/devicetree/bindings/sound/storm.txt delete mode 100644 sound/soc/atmel/atmel-pcm.c create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h create mode 100644 sound/soc/codecs/wm8804-i2c.c create mode 100644 sound/soc/codecs/wm8804-spi.c delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (94%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (99%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (79%) rename sound/soc/intel/{ => atom}/sst/sst.h (98%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (92%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (97%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (95%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (99%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (71%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (86%) rename sound/soc/intel/{ => boards}/byt-max98090.c (94%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (97%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (95%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (75%) rename sound/soc/intel/{ => boards}/haswell.c (96%) rename sound/soc/intel/{ => boards}/mfld_machine.c (96%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (99%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (97%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (99%) rename sound/soc/intel/{ => common}/sst-firmware.c (99%) create mode 100644 sound/soc/intel/common/sst-ipc.c create mode 100644 sound/soc/intel/common/sst-ipc.h create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (98%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (77%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (87%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (88%) create mode 100644 sound/soc/qcom/Kconfig create mode 100644 sound/soc/qcom/Makefile create mode 100644 sound/soc/qcom/lpass-cpu.c create mode 100644 sound/soc/qcom/lpass-lpaif-ipq806x.h create mode 100644 sound/soc/qcom/lpass-platform.c create mode 100644 sound/soc/qcom/lpass.h create mode 100644 sound/soc/qcom/storm.c create mode 100644 sound/soc/sh/rcar/dma.c create mode 100644 sound/soc/sh/rcar/rsrc-card.c

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Re: [alsa-devel] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Dan Carpenter 13 Apr '15

13 Apr '15

Hello Jyri Sarha, The patch a75a053f1eef: "ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used" from Mar 20, 2015, leads to the following Sparse warning: sound/soc/davinci/davinci-mcasp.c:1098:45: warning: Variable length array is used. sound/soc/davinci/davinci-mcasp.c 1088 static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, 1089 struct snd_pcm_hw_rule *rule) 1090 { 1091 struct davinci_mcasp_ruledata *rd = rule->private; 1092 struct snd_interval *ci = 1093 hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); 1094 int sbits = params_width(params); 1095 int rate = params_rate(params); 1096 int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? 1097 rd->mcasp->tdm_slots : ci->max; 1098 unsigned int list[ci->max - ci->min + 1]; ^^^^^^^^^^^^^^^^^^^^^ I wasnt able to determine where these values are capped. The worry is that the kernel only has an 8k stack so if they are too high it could oops. I think these values can come from the user in snd_pcm_hw_params_user() but I'm an newbie to the code and I didn't see where the limits were enforced. 1099 int c1, c, count = 0; 1100 1101 for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { 1102 uint bclk_freq = c1*sbits*rate; 1103 int ppm; regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T450
by Jo-Philipp Wich 13 Apr '15

13 Apr '15

The Lenovo Thinkpad T450 requires the ALC292_FIXUP_TPT440_DOCK as well in order to get working sound output on the docking stations headphone jack. Patch tested on a Thinkpad T450 (20BVCTO1WW) using kernel 4.0-rc7 in conjunction with a ThinkPad Ultradock. Signed-off-by: Jo-Philipp Wich <jow(a)openwrt.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9d12c0..52aa8ae 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5053,6 +5053,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Expose codec type sysfs
by Takashi Iwai 13 Apr '15

13 Apr '15

The type field of HD-audio codec object should be exposed to user-space so that it can identify which driver type to bind (legacy / asoc). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_sysfs.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index 18aea43d230d..0a6ce3b84cc4 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -36,6 +36,7 @@ static ssize_t type##_show(struct device *dev, \ } \ static DEVICE_ATTR_RO(type) +CODEC_ATTR(type); CODEC_ATTR(vendor_id); CODEC_ATTR(subsystem_id); CODEC_ATTR(revision_id); @@ -45,6 +46,7 @@ CODEC_ATTR_STR(vendor_name); CODEC_ATTR_STR(chip_name); static struct attribute *hdac_dev_attrs[] = { + &dev_attr_type.attr, &dev_attr_vendor_id.attr, &dev_attr_subsystem_id.attr, &dev_attr_revision_id.attr, -- 2.3.5

1 1

[alsa-devel] [PATCH] ASoC: Intel: fix ifnullfree.cocci warnings
by kbuild test robot 13 Apr '15

13 Apr '15

sound/soc/intel/common/sst-ipc.c:287:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values. NULL check before some freeing functions is not needed. Based on checkpatch warning "kfree(NULL) is safe this check is probably not required" and kfreeaddr.cocci by Julia Lawall. Generated by: scripts/coccinelle/free/ifnullfree.cocci CC: Jin Yao <yao.jin(a)linux.intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sst-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/intel/common/sst-ipc.c +++ b/sound/soc/intel/common/sst-ipc.c @@ -283,8 +283,7 @@ void sst_ipc_fini(struct sst_generic_ipc if (ipc->tx_thread) kthread_stop(ipc->tx_thread); - if (ipc->msg) - kfree(ipc->msg); + kfree(ipc->msg); } EXPORT_SYMBOL_GPL(sst_ipc_fini);

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Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Ville Syrjälä 13 Apr '15

13 Apr '15

On Mon, Apr 13, 2015 at 12:43:14PM +0300, Mika Kahola wrote: > On Fri, 2015-04-10 at 16:10 +0200, Takashi Iwai wrote: > > At Fri, 10 Apr 2015 16:27:39 +0300, > > Mika Kahola wrote: > > > > > > On Thu, 2015-04-09 at 17:17 +0200, Takashi Iwai wrote: > > > > At Thu, 9 Apr 2015 15:51:27 +0200, > > > > Daniel Vetter wrote: > > > > > > > > > > On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > > > > > > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > > > > > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > > > > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > > > > > > connected to my Haswell box. Before applying the patch I tested as a > > > > > > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > > > > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > > > > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > > > > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > > > > > > turned out that the first commit that I failed to get audio working was > > > > > > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > > > > > > > > > > > I rerun the bisecting and this time the result was > > > > > > > > > > > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > > > > > > Merge: 2d846c7 a59d719 > > > > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > > > > Date: Wed Apr 8 11:29:56 2015 +0200 > > > > > > > > > > > > Merge branch 'for-linus' into for-next > > > > > > > > > > > > Back merge HD-audio quirks to for-next branch, so that we can apply > > > > > > a couple of more quirks. > > > > > > > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > > > > > Adding Takashi and intel audio folks. > > > > > > > > The bisecting looks odd. The commit you pointed is a back-merge from > > > > 4.0-rc to next branch, so this merge itself shouldn't bring so many > > > > stuff -- at least about the sound part. > > > > > > > > The diff in sound/* is found below. As you can see, the only change > > > > relevant with HDMI is the chunk in sound/pci/hda/hda_intel.c for > > > > HD-audio controller, but it's specific to Skylake, thus this must be > > > > irrelevant with your hardware. > > > > > > > > Please double-check. > > > > > > I double-checked the bisecting and I ended up with the same result. You're > > > right, it does seems odd why audio on my Haswell box stopped working. > > > > > > Any ideas how to proceed from here? > > > > Try to check whether the reverting the diff of sound/* manually fixes > > the problem. (i.e. apply diff -R for the diff I showed in the > > previous mail.) > > > > If it doesn't fix but still reverting the whole commit does fix, it > > means that something else got broken by the merge. > > > > > > Takashi > > > > I reverted the diff but unfortunately there were no difference. > > For curiosity, I tested the audio with the latest -nightly and I got > the audio working again. I wonder what has changed but it seems that > the problem is now solved. BTW we now have a WARN triggering on BSW due to hda trying to query the current cdclk frequency. https://bugs.freedesktop.org/show_bug.cgi?id=90012 I'm not sure if hda really needs to know the cdclk freq on BSW or not. But in either case I think we should just kill the HAS_DDI check. Once these patches land the code will anyway work on all platforms. Mika, can you add that to your list of things to do on top of these patches? > > -Mika- > > > > > > > > > Cheers, > > > Mika > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > === > > > > % git diff 0a599838737a2527c35e4d94f794aefe59df1781^..0a599838737a2527c35e4d94f794aefe59df1781 sound > > > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > > > index 9bcc5457a83e..e1c210515581 100644 > > > > --- a/sound/pci/hda/hda_intel.c > > > > +++ b/sound/pci/hda/hda_intel.c > > > > @@ -1967,7 +1967,7 @@ static const struct pci_device_id azx_ids[] = { > > > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > > > /* Sunrise Point */ > > > > { PCI_DEVICE(0x8086, 0xa170), > > > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > > > /* Sunrise Point-LP */ > > > > { PCI_DEVICE(0x8086, 0x9d70), > > > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > > > index e0c06f9a0e80..7f46d063af57 100644 > > > > --- a/sound/pci/hda/patch_realtek.c > > > > +++ b/sound/pci/hda/patch_realtek.c > > > > @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) > > > > { > > > > /* We currently only handle front, HP */ > > > > static hda_nid_t pins[] = { > > > > - 0x0f, 0x10, 0x14, 0x15, 0 > > > > + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 > > > > }; > > > > hda_nid_t *p; > > > > for (p = pins; *p; p++) > > > > @@ -2870,6 +2870,8 @@ static void alc283_init(struct hda_codec *codec) > > > > > > > > if (!hp_pin) > > > > return; > > > > + > > > > + msleep(30); > > > > hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > > > > > > > /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ > > > > @@ -3564,6 +3566,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3619,6 +3622,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_write_coef_idx(codec, 0x45, 0xc489); > > > > snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > > > > alc_process_coef_fw(codec, coef0255); > > > > @@ -3688,6 +3692,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3742,6 +3747,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3796,6 +3802,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > break; > > > > case 0x10ec0233: > > > > @@ -3841,6 +3848,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) > > > > > > > > switch (codec->core.vendor_id) { > > > > case 0x10ec0255: > > > > + case 0x10ec0256: > > > > alc_process_coef_fw(codec, coef0255); > > > > msleep(300); > > > > val = alc_read_coef_idx(codec, 0x46); > > > > @@ -4993,6 +5001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > > > > SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), > > > > SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), > > > > SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > > > + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), > > > > SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > > > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), > > > > SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), > > > > @@ -5173,6 +5182,16 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > > > {0x17, 0x40000000}, > > > > {0x1d, 0x40700001}, > > > > {0x21, 0x02211050}), > > > > + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, > > > > + {0x12, 0x90a60140}, > > > > + {0x13, 0x40000000}, > > > > + {0x14, 0x90170110}, > > > > + {0x19, 0x411111f0}, > > > > + {0x1a, 0x411111f0}, > > > > + {0x1b, 0x411111f0}, > > > > + {0x1d, 0x40700001}, > > > > + {0x1e, 0x411111f0}, > > > > + {0x21, 0x02211020}), > > > > SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, > > > > {0x12, 0x90a60130}, > > > > {0x13, 0x40000000}, > > > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > > > > index 353532b8aee4..8f63e8c8b46c 100644 > > > > --- a/sound/usb/quirks.c > > > > +++ b/sound/usb/quirks.c > > > > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > > > > > > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > > > > { > > > > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > > > > - return chip->usb_id == USB_ID(0x045E, 0x076D); > > > > + /* devices which do not support reading the sample rate. */ > > > > + switch (chip->usb_id) { > > > > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > > > > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > > > > + return true; > > > > + } > > > > + return false; > > > > } > > > > > > > > /* Marantz/Denon USB DACs need a vendor cmd to switch > > > > > > > > > > > > > _______________________________________________ > Intel-gfx mailing list > Intel-gfx(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/intel-gfx -- Ville Syrjälä Intel OTC

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[alsa-devel] [PATCH] ctl: fix to handle several elements added by one operation for userspace element
by Takashi Sakamoto 13 Apr '15

13 Apr '15

An element instance can have several elements with the same feature. Some userspace applications can add such an element instance by add operation with the number of elements. Then, the element instance gets a memory object to keep states of these elements. But the element instance has just one memory object for the elements. This causes the same result to each read/write operations to the different elements. This commit fixes this bug by allocating enough memory objects to the element instance for each of elements. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index ccb1ca2..5b2c352 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1029,7 +1029,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, struct user_element { struct snd_ctl_elem_info info; struct snd_card *card; - void *elem_data; /* element data */ + char *elem_data; /* element data */ unsigned long elem_data_size; /* size of element data in bytes */ void *tlv_data; /* TLV data */ unsigned long tlv_data_size; /* TLV data size */ @@ -1078,9 +1078,12 @@ static int snd_ctl_elem_user_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *src = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - memcpy(&ucontrol->value, ue->elem_data, ue->elem_data_size); + memcpy(&ucontrol->value, src, size); mutex_unlock(&ue->card->user_ctl_lock); return 0; } @@ -1090,11 +1093,14 @@ static int snd_ctl_elem_user_put(struct snd_kcontrol *kcontrol, { int change; struct user_element *ue = kcontrol->private_data; + unsigned int size = ue->elem_data_size; + char *dst = ue->elem_data + + snd_ctl_get_ioff(kcontrol, &ucontrol->id) * size; mutex_lock(&ue->card->user_ctl_lock); - change = memcmp(&ucontrol->value, ue->elem_data, ue->elem_data_size) != 0; + change = memcmp(&ucontrol->value, dst, size) != 0; if (change) - memcpy(ue->elem_data, &ucontrol->value, ue->elem_data_size); + memcpy(dst, &ucontrol->value, size); mutex_unlock(&ue->card->user_ctl_lock); return change; } @@ -1278,7 +1284,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (err < 0) return err; memcpy(&kctl->id, &info->id, sizeof(kctl->id)); - kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size * count, GFP_KERNEL); if (kctl->private_data == NULL) { kfree(kctl); -- 2.1.0

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[alsa-devel] [PATCH] ASoC: Intel: fix array_size.cocci warnings
by kbuild test robot 12 Apr '15

12 Apr '15

sound/soc/intel/haswell/sst-haswell-ipc.c:646:28-29: WARNING: Use ARRAY_SIZE Use ARRAY_SIZE instead of dividing sizeof array with sizeof an element Semantic patch information: This makes an effort to find cases where ARRAY_SIZE can be used such as where there is a division of sizeof the array by the sizeof its first element or by any indexed element or the element type. It replaces the division of the two sizeofs by ARRAY_SIZE. Generated by: scripts/coccinelle/misc/array_size.cocci CC: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sst-haswell-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -643,7 +643,7 @@ static void hsw_fw_ready(struct sst_hsw /* log the FW version info got from the mailbox here. */ memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); pinfo = &fw_info[0]; - for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + for (i = 0; i < ARRAY_SIZE(tmp); i++) tmp[i] = strsep(&pinfo, " "); dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " "version: %s.%s, build %s, source commit id: %s\n",

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Re: [alsa-devel] Issues and/or possible bugs in alsa
by Yomi Ogunwumi 12 Apr '15

12 Apr '15

I don't think I've disabled anything. • Yomi On Apr 12, 2015 12:31 PM, "Raymond Yau" <superquad.vortex2(a)gmail.com> wrote: > > > > I'm not sure if it worked with pervious kernel versions. There was a > time I rolled back earlier kernel versions — The result was the same, it > didn't work. > > > > It might have worked with Windows 8, but I had installed Arch soon after > obtaining the laptop. > > >Second. When I plug in these headphones, I can't use the internal > microphone on this laptop, it mutes. (AFAICT) These headphones do not have > a microphone built in, they're just an ordinary pair of headphones. > However, I was given a workaround in #alsa on freenode. In order to use the > laptop's internal microphone with my headphones plugged in, I have to do : > sudo hda-verb /dev/snd/hwC1D0 0x22 SET_CONNECT_SEL 6 — it works, but it > isn't really ideal. > > 0x22 [Audio Selector] wcaps 0x30010b: Stereo Amp-In > Amp-In caps: N/A > Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 > 0x00] [0x00 0x00] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12* > > Do you mean disable jack detection cannot work around this jack detection > fault ? > > Using hda-emu, the driver seem select internal mic when jack detection is > disabled on startup > > >> > >> How about disable jack detect using hint jack_detect=0 ? > >> > >> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… > >> > >> you have to select mic jack using capture source and manually mute the > speaket >

1 0

[alsa-devel] [PATCH 1/2] ASoC: n810: Consistently pass the card DAPM context to n810_ext_control()
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

Some callers of n810_ext_control() pass the card DAPM context and some pass the CODEC DAPM context. Given that some of the widgets that are accessed in the function are in the card's context, always passing it is the obvious choice. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/n810.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 5d7f9ce..617eae3 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -98,12 +98,11 @@ static int n810_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - n810_ext_control(&codec->dapm); + n810_ext_control(&rtd->card->dapm); return clk_prepare_enable(sys_clkout2); } -- 1.8.0

3 3

[alsa-devel] [PATCH] ASoC: davinci-evm: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/davinci/davinci-evm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 8c2b9be..731fb0d 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -117,7 +117,6 @@ static const struct snd_soc_dapm_route audio_map[] = { static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct snd_soc_codec *codec = rtd->codec; struct device_node *np = card->dev->of_node; int ret; @@ -136,9 +135,9 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) } /* not connected */ - snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM"); - snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); return 0; } -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: mop500_ab8500: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/ux500/mop500_ab8500.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index aa65370..b81a7a4c 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -362,7 +362,7 @@ struct snd_soc_ops mop500_ab8500_ops[] = { int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; struct device *dev = rtd->card->dev; struct mop500_ab8500_drvdata *drvdata; int ret; @@ -407,23 +407,23 @@ int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); - ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + ret = snd_soc_dapm_disable_pin(dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(dapm, "DMic 6"); return ret; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: wm1133-ev1: Use card DAPM context to access widgets
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/wm1133-ev1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index 0653aa8..b454972 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -202,7 +202,6 @@ static struct snd_soc_jack_pin mic_jack_pins[] = { static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, @@ -216,7 +215,7 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias"); return 0; } -- 1.8.0

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[alsa-devel] [PATCH 1/2] ASoC: atmel: Add dependency to SND_SOC_I2C_AND_SPI where necessary
by Lars-Peter Clausen 12 Apr '15

12 Apr '15

The SND_AT91_SOC_SAM9G20_WM8731 and SND_AT91_SOC_SAM9X5_WM8731 machine driver symbols select SND_SOC_WM8731 which depends on SND_SOC_I2C_AND_SPI. So the machine driver symbols need to depend on SND_SOC_I2C_AND_SPI as well, otherwise we might end up with a invalid configuration, which will sooner or later upset the randconfig auto-builders. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/atmel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 1579e99..2f185e5 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,7 +25,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 -- 1.8.0

2 2

[alsa-devel] [PATCH] ASoC: Intel: remove unneeded variable
by Julia Lawall 12 Apr '15

12 Apr '15

Removes unneeded variable used to store return value. Generated by: scripts/coccinelle/misc/returnvar.cocci CC: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- sst-baytrail-ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/intel/baytrail/sst-baytrail-ipc.c +++ b/sound/soc/intel/baytrail/sst-baytrail-ipc.c @@ -511,7 +511,6 @@ static int sst_byt_process_notification( struct sst_byt_stream *stream; u64 header; u8 msg_id, stream_id; - int handled = 1; header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); msg_id = sst_byt_header_msg_id(header); @@ -531,7 +530,7 @@ static int sst_byt_process_notification( break; } - return handled; + return 1; } static irqreturn_t sst_byt_irq_thread(int irq, void *context)

2 1

[alsa-devel] [PATCH 0/2 RFC] control core refactoring
by Takashi Sakamoto 12 Apr '15

12 Apr '15

This patchset is for refactoring control core code. The first patch is re-send http://mailman.alsa-project.org/pipermail/alsa-devel/2015-February/087700.h… The second patch is new. As long as I investigated, the changed code was written in 1999 and its basic design have never been changed. In my opinion, the design is a bit complicated and refactoring it is better to decrease maintaining cost. Takashi Sakamoto (2): ALSA: ctl: evaluate macro instead of numerical value ALSA: ctl: refactoring for read operation sound/core/control.c | 87 ++++++++++++++++++++++++++++++++-------------------- 1 file changed, 54 insertions(+), 33 deletions(-) -- 2.1.0

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Re: [alsa-devel] Playback - Overwrite buffer with silence
by Nilhcraiv 12 Apr '15

12 Apr '15

Hello Clemens, Fisrt of all thanks for your answer. In my code there are error checking, of course. The code of this email thread is only just to simplify the question. Clemens wrote: It is impossible for the boundar to be zero. I am sure, if you compile this code, "boundary" variable is always zero. But the question is, How I can put silence data in buffer if the received data is not available at time? (to avoid under-run) Is the doc wrong? #define FRAMES 240 #define PERIOD 1*FRAMES #define BUFFER_SIZE PERIOD*2 #define CHANNELS 1 #define SAMPLE_RATE 48000 snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int exact_rate,dir; snd_pcm_uframes_t boundary=0; snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); /*PlayBack hw */ snd_pcm_open(&pcm_handle, "default", stream, 0); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE); exact_rate = SAMPLE_RATE; snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, CHANNELS); snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, BUFFER_SIZE); snd_pcm_hw_params(pcm_handle, hwparams); /*Avoid under-run sw*/ snd_pcm_sw_params_current(pcm_handle, swparams); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is allways ZERO*/ snd_pcm_sw_params_set_silence_threshold(pcm_handle, swparams, 0); /*According to the doc.*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary); /*According to the doc.*/ snd_pcm_sw_params(pcm_handle,swparams); /*END Avoid under-run*/ snd_pcm_prepare(pcm_handle); Thanks! Nilhcraiv wrote: > If I obtain the boundary for ring pointers in frames at the start > (snd_pcm_sw_params_get_boundary), the boundary is always zero. It is impossible for the boundar to be zero. > Here you can see a light version of my code (without error checking): Error checking would be important. Regards, Clemens

2 1

[alsa-devel] [PATCH v5 0/5] ALSA: jack: Refactoring for jack kctls
by Jie Yang 12 Apr '15

12 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For hda, a. call snd_jack_kctl_new()(in __snd_hda_jack_add_kctl()) to new a jack_kctl(also new kctl); and use snd_hda_ctl_add to add kctl to card; b. new a hda_jack_tbl, appoint jack_tbl->jack_kctl = jack_kctl; c. for non-phantom jack, call snd_jack_new() with this jack_kctl, then the jack_kctl will be added to the new created jack; Changes in v5: 1. remove SND_KCTL_JACK config item, we need jack kctl once CONFIG_SND_JACK is selected. Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (5): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: add a parameter to pass kctl for snd_jack_new ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove export snd_kctl_jack_new() include/sound/jack.h | 25 ++++++- sound/core/Kconfig | 3 - sound/core/Makefile | 3 +- sound/core/ctljack.c | 1 - sound/core/jack.c | 152 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 1 - sound/pci/hda/hda_jack.c | 52 +++++++------- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 10 files changed, 201 insertions(+), 45 deletions(-) -- 1.9.1

2 6

[alsa-devel] [PATCH 0/4] ALSA: ctl: fix userspace control element operations
by Takashi Sakamoto 12 Apr '15

12 Apr '15

This patchset is to fix some issues related to operations for userspace control element. Three patches are related to identical information, the other is related to several element added by one operation. Unfortunately, current alsa-lib has no functionality to achieve the operation to add several element. To demonstrate this and test these patches, I prepare for a small GObject introspection library, alsa-gir. https://github.com/takaswie/alsa-gir This reposiroty includes sample scripts written by python. The file of sample/ctl.py demonstrates userspace control elements. If this library helps your review for this patchset, I'm feeling happy. Takashi Sakamoto (4): ALSA: ctl: confirm to return all identical information in 'activate' event ALSA: ctl: fix a bug to return no identical information in info operation for userspace controls ALSA: ctl: fill identical information to return value when adding userspace elements ALSA: ctl: fix to handle several elements added by one operation for userspace element sound/core/control.c | 44 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 37 insertions(+), 7 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH] ASoC: rnsd: fix build regression without CONFIG_OF
by Arnd Bergmann 12 Apr '15

12 Apr '15

The r-car sound driver only works when CONFIG_OF is set, and after a recent change has a compile-time dependency as well: sound/built-in.o: In function `rsnd_dma_request_channel': :(.text+0x9fb84): undefined reference to `of_dma_request_slave_channel' This could be fixed either by adding a static inline wrapper for the function, or by adding a Kconfig dependency. This implements the second approach, which seems appropriate because the driver in fact has a hard dependency. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 72adc61f4637aa3 ("ASoC: rsnd: 1st DMAC dma-names cares subnode") diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 2b3030415573..07114b0b0dc1 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" + depends on DMA_OF select SND_SIMPLE_CARD select REGMAP_MMIO help

2 1

[alsa-devel] [GIT PULL] Last minute ASoC fix for v4.0
by Mark Brown 12 Apr '15

12 Apr '15

The following changes since commit c517d838eb7d07bbe9507871fab3931deccff539: Linux 4.0-rc1 (2015-02-22 18:21:14 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.0-rc7 for you to fetch changes up to 14f0413ce3c400a74213a861bc8f47dca5e4d231: ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 (2015-04-11 22:09:14 +0100) ---------------------------------------------------------------- ASoC: Last minute fix for v4.0 This patch backs out a change that came in during the merge window which selects a configuration for GPIO4 on pcm512x CODECs that may not be suitable for all systems using the device. Changes for v4.1 will make this properly configurable but for now it's safest to revert to the v3.19 behaviour and leave the pin configuration alone. Sorry for sending this direct at the last minute but due to the GPIO misuse it'd be really good to get it in the release and I'd not realised it hadn't been sent yet - between some travel, a job change and other non-urgent fixes coming in I'd lost track of the urgency. It's been in -next for several weeks now, is isolated to the driver and fairly clear to inspection. ---------------------------------------------------------------- Howard Mitchell (1): ASoC: pcm512x: Remove hardcoding of pll-lock to GPIO4 sound/soc/codecs/pcm512x.c | 19 ------------------- 1 file changed, 19 deletions(-)

1 0

[alsa-devel] Playback - Overwrite buffer with silence
by Nilhcraiv 11 Apr '15

11 Apr '15

Hello, I am working an application and I have problems playing the audio with ALSA. In my application, I receive the sound data, this data must be decoded to be reproduce. In this process, always obtain an under-runerror. To fix this problem, I followed the documentation i.e. I use snd_pcm_sw_params_set_silence_size() and snd_pcm_sw_params_set_silence_threshold() to overwrite the buffer with silence. --> http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___s_w___params.… but this does not work. Does the doc is wrong? If I obtain the boundary for ring pointers in frames at the start (snd_pcm_sw_params_get_boundary), the boundary is always zero. If I obtain the boundary each time (in the received_data() function) it is more than the buffer size and I get error when I put it in snd_pcm_sw_params_set_silence_size(), ¿Why?, Is it correct to use the boundary value?, ¿How I can avoid under-run by putting silence if I have not data at time? Here you can see a light version of my code (without error checking): #define FRAMES 240 #define PERIOD 1*FRAMES #define BUFFER_SIZE PERIOD*2 #define CHANNELS 1 #define SAMPLE_RATE 48000 snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int exact_rate,dir; snd_pcm_uframes_t boundary=0; snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; snd_pcm_hw_params_alloca(&hwparams); snd_pcm_sw_params_alloca(&swparams); /*PlayBack hw */ snd_pcm_open(&pcm_handle, "default", stream, 0); snd_pcm_hw_params_any(pcm_handle, hwparams); snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE); exact_rate = SAMPLE_RATE; snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &exact_rate, 0); snd_pcm_hw_params_set_channels(pcm_handle, hwparams, CHANNELS); snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, BUFFER_SIZE); snd_pcm_hw_params(pcm_handle, hwparams); /*Avoid under-run sw*/ snd_pcm_sw_params_current(pcm_handle, swparams); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is allways ZERO*/ snd_pcm_sw_params_set_silence_threshold(pcm_handle, swparams, 0); /*According to the doc.*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary); /*According to the doc.*/ snd_pcm_sw_params(pcm_handle,swparams); /*END Avoid under-run*/ snd_pcm_prepare(pcm_handle); void received_data(void *t_data)/*short version of the function*/ { short *decoded_data = NULL; int len=0; decoded_data = (short *)malloc(PERIOD * FRAME_SIZE); len = decoder(&decoded_data,t_data); snd_pcm_sw_params_get_boundary(swparams, &boundary); /*Here, boundary is always a large number*/ snd_pcm_sw_params_set_silence_size(pcm_handle, swparams, boundary);/*Here, boundary in frames is always a large number, and there are an error...*/ snd_pcm_sw_params(pcm_handle,swparams); pcmreturn = snd_pcm_writei(pcm_handle, decoded_data, len);/*In most cases there are UNDERRUN...*/ xrun_recovery(&pcm_handle, pcmreturn);/*Recover in case of an error.*/ } Thanks in advance! Vicente

2 1

[alsa-devel] [PATCH v2] ALSA: emu10k1: add toggles for E-mu 1010 optical ports
by Michael Gernoth 11 Apr '15

11 Apr '15

The optical ports on the E-mu 1010 (and dock) can be configured for ADAT- or S/PDIF-mode, which is currently hardcoded to ADAT. Add two mixer elements to expose this setting. Tested on an E-mu 1010 PCIe with connected Micro Dock. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emumixer.c | 118 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 118 insertions(+) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 1de3302..55e5716 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -806,6 +806,108 @@ static struct snd_kcontrol_new snd_emu1010_internal_clock = .put = snd_emu1010_internal_clock_put }; +static int snd_emu1010_optical_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_out; + return 0; +} + +static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_out != val); + if (change) { + emu->emu1010.optical_out = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_out = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Output Mode", + .count = 1, + .info = snd_emu1010_optical_out_info, + .get = snd_emu1010_optical_out_get, + .put = snd_emu1010_optical_out_put +}; + +static int snd_emu1010_optical_in_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_in; + return 0; +} + +static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0]; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_in != val); + if (change) { + emu->emu1010.optical_in = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_in = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Input Mode", + .count = 1, + .info = snd_emu1010_optical_in_info, + .get = snd_emu1010_optical_in_get, + .put = snd_emu1010_optical_in_put +}; + static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2051,6 +2153,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } else if (emu->card_capabilities->emu_model) { /* all other e-mu cards for now */ @@ -2086,6 +2196,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } if ( emu->card_capabilities->i2c_adc) { -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: emu10k1: add toggles for E-mu 1010 optical ports
by Michael Gernoth 11 Apr '15

11 Apr '15

The optical ports on the E-mu 1010 (and dock) can be configured for ADAT- or SP/DIF-mode, which is currently hardcoded to ADAT. Add two mixer elements to expose this setting. Tested on an E-mu 1010 PCIe with connected micro dock. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emumixer.c | 120 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 120 insertions(+) diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 1de3302..e0d6053 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -806,6 +806,110 @@ static struct snd_kcontrol_new snd_emu1010_internal_clock = .put = snd_emu1010_internal_clock_put }; +static int snd_emu1010_optical_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_out; + return 0; +} + +static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0] ; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_out != val); + if (change) { + emu->emu1010.optical_out = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_out = +{ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Output Mode", + .count = 1, + .info = snd_emu1010_optical_out_info, + .get = snd_emu1010_optical_out_get, + .put = snd_emu1010_optical_out_put +}; + +static int snd_emu1010_optical_in_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { + "SPDIF", "ADAT" + }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_emu1010_optical_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->emu1010.optical_in; + return 0; +} + +static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_emu10k1 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + u32 tmp; + int change = 0; + + val = ucontrol->value.enumerated.item[0] ; + /* Limit: uinfo->value.enumerated.items = 2; */ + if (val >= 2) + return -EINVAL; + change = (emu->emu1010.optical_in != val); + if (change) { + emu->emu1010.optical_in = val; + tmp = (emu->emu1010.optical_in ? EMU_HANA_OPTICAL_IN_ADAT : 0) | + (emu->emu1010.optical_out ? EMU_HANA_OPTICAL_OUT_ADAT : 0); + snd_emu1010_fpga_write(emu, EMU_HANA_OPTICAL_TYPE, tmp); + } + return change; +} + +static struct snd_kcontrol_new snd_emu1010_optical_in = +{ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Optical Input Mode", + .count = 1, + .info = snd_emu1010_optical_in_info, + .get = snd_emu1010_optical_in_get, + .put = snd_emu1010_optical_in_put +}; + static int snd_audigy_i2c_capture_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2051,6 +2155,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } else if (emu->card_capabilities->emu_model) { /* all other e-mu cards for now */ @@ -2086,6 +2198,14 @@ int snd_emu10k1_mixer(struct snd_emu10k1 *emu, snd_ctl_new1(&snd_emu1010_internal_clock, emu)); if (err < 0) return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_out, emu)); + if (err < 0) + return err; + err = snd_ctl_add(card, + snd_ctl_new1(&snd_emu1010_optical_in, emu)); + if (err < 0) + return err; } if ( emu->card_capabilities->i2c_adc) { -- 2.1.4

2 2

[alsa-devel] [PATCH] ALSA: emu10k1: handle dock disconnects
by Michael Gernoth 11 Apr '15

11 Apr '15

When the dock on an E-mu 1010 card is disconnected, all outputs get muted by the hardware. Add logic to detect a disconnect and unmute. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index b4458a6..ee50c1d 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -707,6 +707,7 @@ static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; u32 tmp, tmp2, reg; + u32 last_reg = 0; int err; for (;;) { @@ -782,7 +783,15 @@ static int emu1010_firmware_thread(void *data) msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (!reg && last_reg) { + /* Audio Dock removed */ + dev_info(emu->card->dev, + "emu1010: Audio Dock detached\n"); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + + last_reg = reg; } dev_info(emu->card->dev, "emu1010: firmware thread stopping\n"); return 0; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: seq: fill client ID in return value of pool operation
by Takashi Sakamoto 11 Apr '15

11 Apr '15

The returned value of 'get/seq client pool' operation has zeroed value for its client ID, against requested client ID. This commit fix the bug by filling it with index value of referred client object. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 4828765..edbdab8 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1879,6 +1879,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr == NULL) return -ENOENT; memset(&info, 0, sizeof(info)); + info.client = cptr->number; info.output_pool = cptr->pool->size; info.output_room = cptr->pool->room; info.output_free = info.output_pool; -- 2.1.0

2 1

[alsa-devel] [PATCH 0/2] Cygnus Audio Driver
by Scott Branden 11 Apr '15

11 Apr '15

Please provide comments on the initial version of this driver. This patchset contains the devicetree bindings and core audio driver for the Cygnus SoC. There is an open question on how to fit this driver into the clock framework (if at all). The audio PLL is embedded in the audio block and only used by the audio block. The audio PLL registers are also in the middle of the audio register map. In addition, the audio PLL is adjustable to less than 1 Hz. The existing clock driver framework does not provide a mechanism to take advantage of the resolution of the hardware. Can the audio PLL remain within the audio driver and/or any modifications required? Lori Hikichi (2): ASoC: cygnus-audio: adding device tree bindings ASoC: add core audio driver for Broadcom Cygnus SOC. .../bindings/sound/brcm,cygnus-audio.txt | 68 + sound/soc/bcm/Kconfig | 11 + sound/soc/bcm/Makefile | 5 +- sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++ sound/soc/bcm/cygnus-pcm.h | 45 + sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 84 + 7 files changed, 2743 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-pcm.h create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 2.3.3

4 10

Re: [alsa-devel] [PATCH v3] ASoC: max98090: add shutdown callback for max98090
by anish kumar 11 Apr '15

11 Apr '15

On Fri, Apr 10, 2015 at 5:45 PM, anish singh <anish198519851985(a)gmail.com> wrote: > > ---------- Forwarded message ---------- > From: Caesar Wang <wxt(a)rock-chips.com> > Date: Wed, Apr 8, 2015 at 4:05 AM > Subject: [PATCH v3] ASoC: max98090: add shutdown callback for max98090 > To: Sonny Rao <sonnyrao(a)chromium.org>, Cheng-Yi Chiang > <cychiang(a)chromium.org>, Heiko Stuebner <heiko(a)sntech.de>, Mark Brown > <broonie(a)kernel.org>, zhengxing <zhengxing(a)rock-chips.com>, "jay.xu" > <xjq(a)rock-chips.com> > Cc: devicetree(a)vger.kernel.org, dianders(a)chromium.org, > dri-devel(a)lists.freedesktop.org, linux-kernel(a)vger.kernel.org, > linux-rockchip(a)lists.infradead.org, linux-arm-kernel(a)lists.infradead.org, > Caesar Wang <wxt(a)rock-chips.com>, Jaroslav Kysela <perex(a)perex.cz>, > alsa-devel(a)alsa-project.org, Takashi Iwai <tiwai(a)suse.de>, Dylan Reid > <dgreid(a)chromium.org>, Liam Girdwood <lgirdwood(a)gmail.com>, Lars-Peter > Clausen <lars(a)metafoo.de>, Jarkko Nikula <jarkko.nikula(a)linux.intel.com>, > Tushar Behera <tushar.b(a)samsung.com> > > > To fix pop noise when shutdown,the pop noise during shutdown > is the pmic cutoff power of codec without any notice. > > Signed-off-by: jay.xu <xjq(a)rock-chips.com> > Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> > Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> > > --- > > Changes in v3: > - modify the shutdown function before remove(..) > - fix the `Serien-cc` > > Changes in v2: > - remove the dev_info(..) > - fix the comment style > - add the max98090_i2c_shutdown() in remove() fuction > > sound/soc/codecs/max98090.c | 17 +++++++++++++++++ > 1 file changed, 17 insertions(+) > > diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c > index b112b1c..3e33ef2 100644 > --- a/sound/soc/codecs/max98090.c > +++ b/sound/soc/codecs/max98090.c > @@ -2605,8 +2605,24 @@ err_enable: > return ret; > } > Please check attached patch and see if it solves your problem. We tested it here with zcd on. +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + dev_info(&i2c->dev, "shut down device\n"); + + /* enable zero crossing */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_ZDENN_MASK-1); + + /* turn down volume levels for all the outputs */ + regmap_write(max98090->regmap, + M98090_REG_RCV_LOUTL_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LOUTR_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LEFT_SPK_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_RIGHT_SPK_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LEFT_HP_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_RIGHT_HP_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + if it doesn't solve your problem then please dump your registers and send it to us to look further. I assume dapm is turning off the widgets before shutting down the codec. > +static void max98090_i2c_shutdown(struct i2c_client *i2c) > +{ > + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); > + > + /* > + * Enable volume smoothing, disable zero cross. This will cause > + * a quick 40ms ramp to mute on shutdown. > + */ > + regmap_write(max98090->regmap, > + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); > + regmap_write(max98090->regmap, > + M98090_REG_DEVICE_SHUTDOWN, 0x00); > + msleep(40); > +} > + > static int max98090_i2c_remove(struct i2c_client *client) > { > + max98090_i2c_shutdown(client); > snd_soc_unregister_codec(&client->dev); > return 0; > } > @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { > .acpi_match_table = ACPI_PTR(max98090_acpi_match), > }, > .probe = max98090_i2c_probe, > + .shutdown = max98090_i2c_shutdown, > .remove = max98090_i2c_remove, > .id_table = max98090_i2c_id, > }; > -- > 1.9.1 > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 10 Apr '15

10 Apr '15

Remove rate constraints from generic ASoC AC'97 CODEC and make it selectable in config. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..5b3224c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..84cad9a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -211,8 +211,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

1 0

[alsa-devel] [PATCH 0/3] ASoC: Intel: IPC/mailbox code refactoring
by Jin Yao 10 Apr '15

10 Apr '15

Hi Mark, Currently in Intel SST driver, some very similar or duplicated IPC/mailbox processing code are used in different platforms. This is a patch series which refactors the common IPC/mailbox code into generic APIs and use the new APIs in Baytrail and Broadwell platforms. This patch series depends on Jie Yang's following patches for build: ASoC: Intel: create atom folder and move atom platform files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create common folder and move common files in Since now Jie Yang's patch series has been applied in intel branch so this patch series will be no build issue. Thanks Jin Yao Jin Yao (3): ASoC: Intel: Refactor common IPC/mailbox code into generic APIs ASoC: Intel: Use the generic IPC/mailbox APIs in Baytrail ASoC: Intel: Use the generic IPC/mailbox APIs in Broadwell sound/soc/intel/baytrail/sst-baytrail-ipc.c | 360 ++++++-------------------- sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/sst-ipc.c | 294 +++++++++++++++++++++ sound/soc/intel/common/sst-ipc.h | 91 +++++++ sound/soc/intel/haswell/sst-haswell-ipc.c | 382 +++++++--------------------- 5 files changed, 551 insertions(+), 579 deletions(-) create mode 100644 sound/soc/intel/common/sst-ipc.c create mode 100644 sound/soc/intel/common/sst-ipc.h -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl: Add the audio interfacce acronyms in Kconfig text
by Fabio Estevam 10 Apr '15

10 Apr '15

To keep consistency with the other Kconfig entries, use the audio interfacce acronyms (SSI and SPDIF) in the Kconfig menu text. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..fa44fc6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI in-tree drivers select it automatically. config SND_SOC_FSL_SSI - tristate "Synchronous Serial Interface module support" + tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO @@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI in-tree drivers select it automatically. config SND_SOC_FSL_SPDIF - tristate "Sony/Philips Digital Interface module support" + tristate "Sony/Philips Digital Interface (SPDIF) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: soc-core: call snd_soc_runtime_set_dai_fmt() before soc_new_pcm()
by Kuninori Morimoto 10 Apr '15

10 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current snd_soc_runtime_set_dai_fmt() is called after soc_probe_link_dais(). this means snd_soc_dai_set_fmt() will be called after soc_new_pcm(). Before appling 1efb53a220b78fdfdbb97b726a2156713e75bdab (ASoC: simple-card: Remove support for setting differing DAI formats) simple-card user had (1) snd_soc_dai_set_fmt() -> soc_new_pcm(), but, after that it is (2) soc_new_pcm() -> snd_soc_dai_set_fmt(). At least rsnd driver is assuming (1) pattern. This patch move snd_soc_dai_set_fmt() into soc_probe_link_dais() after the dai_link->init section to solve this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This patch solves my issue, but I don't know how much effect it has to other drivers. sound/soc/soc-core.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 76bfff2..9777e78 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1324,6 +1324,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } + if (dai_link->dai_fmt) + snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); + ret = soc_post_component_init(rtd, dai_link->name); if (ret) return ret; @@ -1642,12 +1645,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].dai_fmt) - snd_soc_runtime_set_dai_fmt(&card->rtd[i], - card->dai_link[i].dai_fmt); - } - snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Use devm_ioremap_resource()
by Fabio Estevam 10 Apr '15

10 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using platform_get_resource() and devm_ioremap_resource() can make the code a bit simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 143cc16..e8bb8ee 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1288,7 +1288,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) const struct of_device_id *of_id; const char *p, *sprop; const uint32_t *iprop; - struct resource res; + struct resource *res; void __iomem *iomem; char name[64]; @@ -1335,19 +1335,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->cpu_dai_drv.name = dev_name(&pdev->dev); - /* Get the addresses and IRQ */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return ret; - } - ssi_private->ssi_phys = res.start; - - iomem = devm_ioremap(&pdev->dev, res.start, resource_size(&res)); - if (!iomem) { - dev_err(&pdev->dev, "could not map device resources\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iomem = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(iomem)) + return PTR_ERR(iomem); + ssi_private->ssi_phys = res->start; ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { -- 1.9.1

2 1

Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Takashi Iwai 10 Apr '15

10 Apr '15

At Fri, 10 Apr 2015 16:27:39 +0300, Mika Kahola wrote: > > On Thu, 2015-04-09 at 17:17 +0200, Takashi Iwai wrote: > > At Thu, 9 Apr 2015 15:51:27 +0200, > > Daniel Vetter wrote: > > > > > > On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > > > > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > > > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > > > > connected to my Haswell box. Before applying the patch I tested as a > > > > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > > > > turned out that the first commit that I failed to get audio working was > > > > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > > > > > > > I rerun the bisecting and this time the result was > > > > > > > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > > > > Merge: 2d846c7 a59d719 > > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > > Date: Wed Apr 8 11:29:56 2015 +0200 > > > > > > > > Merge branch 'for-linus' into for-next > > > > > > > > Back merge HD-audio quirks to for-next branch, so that we can apply > > > > a couple of more quirks. > > > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > Adding Takashi and intel audio folks. > > > > The bisecting looks odd. The commit you pointed is a back-merge from > > 4.0-rc to next branch, so this merge itself shouldn't bring so many > > stuff -- at least about the sound part. > > > > The diff in sound/* is found below. As you can see, the only change > > relevant with HDMI is the chunk in sound/pci/hda/hda_intel.c for > > HD-audio controller, but it's specific to Skylake, thus this must be > > irrelevant with your hardware. > > > > Please double-check. > > I double-checked the bisecting and I ended up with the same result. You're > right, it does seems odd why audio on my Haswell box stopped working. > > Any ideas how to proceed from here? Try to check whether the reverting the diff of sound/* manually fixes the problem. (i.e. apply diff -R for the diff I showed in the previous mail.) If it doesn't fix but still reverting the whole commit does fix, it means that something else got broken by the merge. Takashi > > Cheers, > Mika > > > > thanks, > > > > Takashi > > > > === > > % git diff 0a599838737a2527c35e4d94f794aefe59df1781^..0a599838737a2527c35e4d94f794aefe59df1781 sound > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index 9bcc5457a83e..e1c210515581 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -1967,7 +1967,7 @@ static const struct pci_device_id azx_ids[] = { > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > /* Sunrise Point */ > > { PCI_DEVICE(0x8086, 0xa170), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > /* Sunrise Point-LP */ > > { PCI_DEVICE(0x8086, 0x9d70), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > index e0c06f9a0e80..7f46d063af57 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) > > { > > /* We currently only handle front, HP */ > > static hda_nid_t pins[] = { > > - 0x0f, 0x10, 0x14, 0x15, 0 > > + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 > > }; > > hda_nid_t *p; > > for (p = pins; *p; p++) > > @@ -2870,6 +2870,8 @@ static void alc283_init(struct hda_codec *codec) > > > > if (!hp_pin) > > return; > > + > > + msleep(30); > > hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > > > /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ > > @@ -3564,6 +3566,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3619,6 +3622,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_write_coef_idx(codec, 0x45, 0xc489); > > snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > > alc_process_coef_fw(codec, coef0255); > > @@ -3688,6 +3692,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3742,6 +3747,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3796,6 +3802,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3841,6 +3848,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > msleep(300); > > val = alc_read_coef_idx(codec, 0x46); > > @@ -4993,6 +5001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > > SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), > > SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), > > SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), > > SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), > > SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), > > @@ -5173,6 +5182,16 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > {0x17, 0x40000000}, > > {0x1d, 0x40700001}, > > {0x21, 0x02211050}), > > + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, > > + {0x12, 0x90a60140}, > > + {0x13, 0x40000000}, > > + {0x14, 0x90170110}, > > + {0x19, 0x411111f0}, > > + {0x1a, 0x411111f0}, > > + {0x1b, 0x411111f0}, > > + {0x1d, 0x40700001}, > > + {0x1e, 0x411111f0}, > > + {0x21, 0x02211020}), > > SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, > > {0x12, 0x90a60130}, > > {0x13, 0x40000000}, > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > > index 353532b8aee4..8f63e8c8b46c 100644 > > --- a/sound/usb/quirks.c > > +++ b/sound/usb/quirks.c > > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > > { > > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > > - return chip->usb_id == USB_ID(0x045E, 0x076D); > > + /* devices which do not support reading the sample rate. */ > > + switch (chip->usb_id) { > > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > > + return true; > > + } > > + return false; > > } > > > > /* Marantz/Denon USB DACs need a vendor cmd to switch > > > >

1 0

[alsa-devel] [PATCH] ALSA: emu10k1: don't deadlock in proc-functions
by Michael Gernoth 10 Apr '15

10 Apr '15

The functions snd_emu10k1_proc_spdif_read and snd_emu1010_fpga_read acquire the emu_lock before accessing the FPGA. The function used to access the FPGA (snd_emu1010_fpga_read) also tries to take the emu_lock which causes a deadlock. Remove the outer locking in the proc-functions (guarding only the already safe fpga read) to prevent this deadlock. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emuproc.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 2ca9f2e..415d23c 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -245,27 +245,19 @@ static void snd_emu10k1_proc_spdif_read(struct snd_info_entry *entry, u32 rate; if (emu->card_capabilities->emu_model) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x38, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); if ((value & 0x1) == 0) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x2a, &value); snd_emu1010_fpga_read(emu, 0x2b, &value2); - spin_unlock_irqrestore(&emu->emu_lock, flags); rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "ADAT Locked : %u\n", rate); } else { snd_iprintf(buffer, "ADAT Unlocked\n"); } - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x20, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); if ((value & 0x4) == 0) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, 0x28, &value); snd_emu1010_fpga_read(emu, 0x29, &value2); - spin_unlock_irqrestore(&emu->emu_lock, flags); rate = 0x1770000 / (((value << 5) | value2)+1); snd_iprintf(buffer, "SPDIF Locked : %d\n", rate); } else { @@ -415,9 +407,7 @@ static void snd_emu_proc_emu1010_reg_read(struct snd_info_entry *entry, snd_iprintf(buffer, "EMU1010 Registers:\n\n"); for(i = 0; i < 0x40; i+=1) { - spin_lock_irqsave(&emu->emu_lock, flags); snd_emu1010_fpga_read(emu, i, &value); - spin_unlock_irqrestore(&emu->emu_lock, flags); snd_iprintf(buffer, "%02X: %08X, %02X\n", i, value, (value >> 8) & 0x7f); } } -- 2.1.4

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[alsa-devel] ALSA bug tracker link
by maruthi srinivas 10 Apr '15

10 Apr '15

Hi, The link : https://bugtrack.alsa-project.org/alsa-bug isn't working. After searching in web , I found a discussion which says, that link will be down and we need to send alsa-devel mailing list for any bugs. Is there a different bug tracker being used now or reporting to alsa-devel mailing list is the only way ? Please confirm. Regards, Maruthi

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Use devm_snd_soc_register_component()
by Fabio Estevam 10 Apr '15

10 Apr '15

Using devm_snd_soc_register_component() can make the code shorter and cleaner. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..143cc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1393,8 +1393,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_register_component(&pdev->dev, &fsl_ssi_component, - &ssi_private->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_ssi_component, + &ssi_private->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto error_asoc_register; @@ -1407,13 +1407,13 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "could not claim irq %u\n", ssi_private->irq); - goto error_irq; + goto error_asoc_register; } } ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); if (ret) - goto error_irq; + goto error_asoc_register; /* * If codec-handle property is missing from SSI node, we assume @@ -1454,9 +1454,6 @@ done: error_sound_card: fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); -error_irq: - snd_soc_unregister_component(&pdev->dev); - error_asoc_register: if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); @@ -1472,7 +1469,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); - snd_soc_unregister_component(&pdev->dev); if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: ak4642: fixup channels_min
by Kuninori Morimoto 10 Apr '15

10 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ak4642 doesn't have Mono record, ak4643 have it, but not supported. This patch fixes channel mismatch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 24ceb36..13585e8 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -499,13 +499,13 @@ static struct snd_soc_dai_driver ak4642_dai = { .name = "ak4642-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE }, -- 1.9.1

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[alsa-devel] [PATCH 0/5] HD-audio widget power-save fixes
by Takashi Iwai 10 Apr '15

10 Apr '15

Hi, this is a series of small fixes for the recently added HD-audio widget power-save feature. Especially the last patch should fix the Realtek codec breakage. The patches are found in topic/hda branch of sound git tree, too. Takashi === Takashi Iwai (5): ALSA: hda/generic - Check power state cap at updating the widget power ALSA: hda/generic - Fix wrong initial power state for fixed pins ALSA: hda/generic - Make snd_hda_gen_path_power_filter() always applicable ALSA: hda/generic - Don't override power_filter when power_save_node is set ALSA: hda/realtek - Fix the regression by widget power-saving sound/pci/hda/hda_generic.c | 23 +++++++++++++++++++---- sound/pci/hda/patch_realtek.c | 5 +++-- 2 files changed, 22 insertions(+), 6 deletions(-) -- 2.3.5

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[alsa-devel] [PATCH 0/3] ARM: shmobile: armadillo: fixup CPU settings
by Kuninori Morimoto 10 Apr '15

10 Apr '15

Hi Mark, Lars Cc Simon These patches had been posted in ALSA/SH-ARM ML in http://thread.gmane.org/gmane.linux.ports.sh.devel/43079 by Lars. But, he doesn't have armadillo board, and I commented about FSI driver, and no one update these, these patches were marooned. Original 2/3 patch changed behavior for clock inversion on FSI driver. FSI + wm8978 on armadillo800eva worked without any issues, but, I don't know how much effect it has for other board (many board are using FSI). We used this inversion flags on each board for historical reasons, but, almost all these were not needed (except some picky board) on FSI. Maybe Lars's 2/3 patch is correct, but, it is difficult to check/confirm for all boards. And unfortunately, Renesas don't use FSI anymore. So, I think keeping current FSI driver as-is is more safety for old boards. I tested these patches on latest Mark's branch Lars-Peter Clausen (3): ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format ARM: shmobile: armadillo800eva: fix clock inversion ASoC: simple-card: Remove support for setting differing DAI formats arch/arm/mach-shmobile/board-armadillo800eva.c | 3 +-- include/sound/simple_card.h | 1 - sound/soc/generic/simple-card.c | 30 ++++++++---------------------- 3 files changed, 9 insertions(+), 25 deletions(-)

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[alsa-devel] [PATCH v2 RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 10 Apr '15

10 Apr '15

Add MAX98090 audio codec, I2S interface and the sound nodes to support audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio in Odroid-XU3 board using simple-audio-card DT binding. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ 2 files changed, 68 insertions(+) diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi index ac6f860..73886d6 100644 --- a/arch/arm/boot/dts/exynos5420.dtsi +++ b/arch/arm/boot/dts/exynos5420.dtsi @@ -415,6 +415,9 @@ <&clock_audss EXYNOS_I2S_BUS>, <&clock_audss EXYNOS_SCLK_I2S>; clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk0"; + #sound-dai-cells = <1>; samsung,idma-addr = <0x03000000>; pinctrl-names = "default"; pinctrl-0 = <&i2s0_bus>; @@ -429,6 +432,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk1"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s1_bus>; status = "disabled"; @@ -442,6 +448,9 @@ dma-names = "tx", "rx"; clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; clock-names = "iis", "i2s_opclk0"; + #clock-cells = <1>; + clock-output-names = "i2s_cdclk2"; + #sound-dai-cells = <1>; pinctrl-names = "default"; pinctrl-0 = <&i2s2_bus>; status = "disabled"; diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts index edc25cf..c038f0d 100644 --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts @@ -11,6 +11,7 @@ */ /dts-v1/; +#include <dt-bindings/sound/samsung-i2s.h> #include "exynos5800.dtsi" / { @@ -285,6 +286,64 @@ rtc@101E0000 { status = "okay"; }; + + sound: sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "Odroid-XU3"; + simple-audio-card,widgets = + "Headphone", "Headphone Jack", + "Speakers", "Speakers"; + simple-audio-card,routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "Headphone Jack", "MICBIAS", + "IN1", "Headphone Jack", + "Speakers", "SPKL", + "Speakers", "SPKR"; + + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&link0_codec>; + simple-audio-card,frame-master = <&link0_codec>; + + simple-audio-card,cpu { + sound-dai = <&i2s0 0>; + system-clock-frequency = <19200000>; + }; + + link0_codec: simple-audio-card,codec { + sound-dai = <&max98090>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + }; + }; +}; + +&clock_audss { + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, + <&clock_audss EXYNOS_MOUT_I2S>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + assigned-clock-parents = <&clock CLK_FIN_PLL>, + <&clock_audss EXYNOS_MOUT_AUDSS>; + assigned-clock-rates = <0>, + <0>, + <19200000>; +}; + +&hsi2c_5 { + status = "okay"; + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx3>; + interrupts = <2 0>; + clocks = <&i2s0 CLK_I2S_CDCLK>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; +}; + +&i2s0 { + status = "okay"; }; &hdmi { -- 2.0.0.390.gcb682f8

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Re: [alsa-devel] [PATCH 1/2] ASoC: davinci: Clear VCIF read FIFO on errors
by Peter Ujfalusi 10 Apr '15

10 Apr '15

On 04/10/2015 02:00 AM, Alexey Ignatov wrote: > Clearing VCIF read FIFO on overflow and underflow events fixes audio glitching > when DMA load is high. This happened when simultaneously capturing and encoding > video and audio, for example. > > Signed-off-by: Alexey Ignatov <lexszero(a)gmail.com> > --- > sound/soc/davinci/davinci-vcif.c | 60 +++++++++++++++++++++++++++++++++++++--- > 1 file changed, 56 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c > index 5bee0427..0d8014e 100644 > --- a/sound/soc/davinci/davinci-vcif.c > +++ b/sound/soc/davinci/davinci-vcif.c > @@ -45,6 +45,10 @@ > } \ > } while (0) > > +#define DAVINCI_VC_INT_RERR_MASK \ > + (DAVINCI_VC_INT_RERRUDR_MASK | \ > + DAVINCI_VC_INT_RERROVF_MASK) > + > struct davinci_vcif_dev { > struct davinci_vc *davinci_vc; > struct davinci_pcm_dma_params dma_params[2]; > @@ -87,6 +91,22 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) > writel(w, davinci_vc->base + DAVINCI_VC_CTRL); > } > > +static void davinci_vcif_interrupts(struct snd_pcm_substream *substream, > + int enable) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct davinci_vcif_dev *davinci_vcif_dev = > + snd_soc_dai_get_drvdata(rtd->cpu_dai); > + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; > + > + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { > + writel(DAVINCI_VC_INT_RERR_MASK, > + davinci_vc->base + DAVINCI_VC_INTCLR); > + writel(enable ? DAVINCI_VC_INT_RERR_MASK : 0, > + davinci_vc->base + DAVINCI_VC_INTEN); > + } > +} > + > static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, > struct snd_soc_dai *dai) > @@ -104,10 +124,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, > /* General line settings */ > writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL); > > - writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR); > - > - writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN); > - > w = readl(davinci_vc->base + DAVINCI_VC_CTRL); > > /* Determine xfer data type */ > @@ -149,6 +165,32 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, > return 0; > } > > +static irqreturn_t davinci_vcif_irq_handler(int irq, void *data) > +{ > + struct davinci_vcif_dev *davinci_vcif_dev = data; > + struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; > + uint32_t w; > + > + w = readl(davinci_vc->base + DAVINCI_VC_INTSTATUS); > + > + if (w & DAVINCI_VC_INT_RERR_MASK) { > + pr_debug("vc overflow or underflow occurred, resetting fifo\n"); > + > + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); > + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RFIFOCL, 1); > + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); > + > + w = readl(davinci_vc->base + DAVINCI_VC_CTRL); > + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RFIFOCL, 0); > + writel(w, davinci_vc->base + DAVINCI_VC_CTRL); > + > + writel(DAVINCI_VC_INT_RERR_MASK, > + davinci_vc->base + DAVINCI_VC_INTCLR); > + } > + > + return IRQ_HANDLED; > +} > + > static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, > struct snd_soc_dai *dai) > { > @@ -159,10 +201,12 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, > case SNDRV_PCM_TRIGGER_RESUME: > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > davinci_vcif_start(substream); > + davinci_vcif_interrupts(substream, 1); > break; > case SNDRV_PCM_TRIGGER_STOP: > case SNDRV_PCM_TRIGGER_SUSPEND: > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + davinci_vcif_interrupts(substream, 0); > davinci_vcif_stop(substream); > break; > default: > @@ -178,6 +222,7 @@ static int davinci_vcif_startup(struct snd_pcm_substream *substream, > struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); > > snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); > + > return 0; > } > > @@ -238,6 +283,13 @@ static int davinci_vcif_probe(struct platform_device *pdev) > > dev_set_drvdata(&pdev->dev, davinci_vcif_dev); > > + ret = devm_request_irq(&pdev->dev, IRQ_MBXINT, davinci_vcif_irq_handler, > + 0, "vcif", davinci_vcif_dev); > + if (ret != 0) { > + dev_err(&pdev->dev, "could not request irq: %d\n", ret); > + return ret; > + } I would make this optional to avoid regression if the arch patch is not present. > + > ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, > &davinci_vcif_dai, 1); > if (ret != 0) { > -- Péter

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[alsa-devel] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 10 Apr '15

10 Apr '15

This adds basic support for the Mediatek AFE unit for MT8173 SoC. This patch is based on Linux 4.0-rc1, Sascha's clk patch [1], and his SCPSYS power patch [2]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). The AFE unit is a DMA master, so no external DMA engine is needed. Each external interface is presented as a DAI to ASoC. A memory interface must be connected via the interconnect to an external interface. The connection pathes are configured through the device tree. [1] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000079.html [2] http://lists.infradead.org/pipermail/linux-mediatek/2015-March/000132.html Koro Chen (2): ASoC: mediatek: Add binding support for AFE driver ASoC: mediatek: Add AFE platform driver .../devicetree/bindings/sound/mtk-afe-pcm.txt | 105 ++ include/dt-bindings/sound/mtk-afe.h | 36 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mediatek/Kconfig | 8 + sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mtk-afe-common.h | 105 ++ sound/soc/mediatek/mtk-afe-connection.c | 416 ++++++ sound/soc/mediatek/mtk-afe-connection.h | 30 + sound/soc/mediatek/mtk-afe-pcm.c | 1497 ++++++++++++++++++++ 10 files changed, 2201 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt create mode 100644 include/dt-bindings/sound/mtk-afe.h create mode 100644 sound/soc/mediatek/Kconfig create mode 100644 sound/soc/mediatek/Makefile create mode 100644 sound/soc/mediatek/mtk-afe-common.h create mode 100644 sound/soc/mediatek/mtk-afe-connection.c create mode 100644 sound/soc/mediatek/mtk-afe-connection.h create mode 100644 sound/soc/mediatek/mtk-afe-pcm.c --

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[alsa-devel] [PATCH 00/14] parport: check success of attach call
by Sudip Mukherjee 10 Apr '15

10 Apr '15

Currently we are not checking if attach has succeeded or not. Also parport_register_driver() always return 0 even if attach fails. But in many places where attach has been used, it can fail. So, modified the parport code to check the return value of attach and accordingly return either 0 or error code from parport_register_driver. Sudip Mukherjee (14): parport: return value of attach and parport_register_driver ALSA: portman2x4: return proper error values from attach ALSA: mts64: return proper error values from attach staging: panel: return proper error values from attach spi: lm70llp: return proper error values from attach spi: butterfly: return proper error values from attach [SCSI] ppa: return proper error values from attach [SCSI] imm: return proper error values from attach pps: return proper error values from attach pps: return proper error values from attach net: plip: return proper error values from attach i2c-parport: return proper error values from attach ppdev: return proper error values from attach char: lp: return proper error values from attach drivers/char/lp.c | 16 +++++++++++----- drivers/char/ppdev.c | 10 +++++++--- drivers/i2c/busses/i2c-parport.c | 7 ++++--- drivers/net/plip/plip.c | 16 ++++++++++------ drivers/parport/share.c | 20 +++++++++++++++----- drivers/pps/clients/pps_parport.c | 7 ++++--- drivers/pps/generators/pps_gen_parport.c | 9 +++++---- drivers/scsi/imm.c | 4 ++-- drivers/scsi/ppa.c | 4 ++-- drivers/spi/spi-butterfly.c | 7 ++++--- drivers/spi/spi-lm70llp.c | 7 ++++--- drivers/staging/panel/panel.c | 11 ++++++----- include/linux/parport.h | 2 +- sound/drivers/mts64.c | 13 ++++++++----- sound/drivers/portman2x4.c | 15 ++++++++++----- 15 files changed, 93 insertions(+), 55 deletions(-) -- 1.8.1.2

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Re: [alsa-devel] [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
by Mark Brown 10 Apr '15

10 Apr '15

On Thu, Apr 02, 2015 at 11:47:18AM -0700, Lori Hikichi wrote: > On 15-03-30 11:42 PM, Mark Brown wrote: > >>+config SND_SOC_CYGNUS > >>+ tristate "SoC platform audio for Broadcom Cygnus chips" > >>+ depends on ARCH_BCM_CYGNUS || COMPILE_TEST > >>+ default ARCH_BCM_CYGNUS > Okay. You don't need to reply to every single comment, the general assumption will be that if there's no other followup all review comments will be addressed. It's better to just reply to things where there's something more detailed to say, if you explicitly reply to everything then that makes it easier for actual replies to be missed since there's a lot of there's a lot of the mail that's just going to be skipped through. > >>+static void ringbuf_inc(void __iomem *audio_io, bool is_playback, > >>+ const struct ringbuf_regs *p_rbuf) > >So it looks like we're getting an interrupt per period and we have to > >manually advance to the next one? > Yes. OK, so that seems fragile - what happens if we're slightly late processing an interrupt or miss one entirely? Most hardware has some way to read back the current position which tends to be more reliable, is that not an option here?

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Re: [alsa-devel] [PATCH] ASoC: max98090: add shutdown callback for max98090
by Jianqun 10 Apr '15

10 Apr '15

Hi Anish Kumar 在 04/10/2015 03:38 AM, Anish Kumar 写道: >> To fix pop noise when shutdown,the pop noise during shutdown >> is the pmic cutoff power of codec without any notice. > Is the power turned off without any intimation to the codec driver? > Or is it physically removed? Can you describe the probelm > better? When device is shutdown with playing music, the origin driver does nothing for shutdown operation, as a result, pmic will cut off the power of codec IC, thus cause a pop noise from speaker and headphone. So the patch added a shutdown callback for driver, enable volume smoothing and disable zero cross, this is referenced from mic_input which does the samething before mic switch. I'm very happy if some suggestion from maximintegrate . :-) > > +static void max98090_i2c_shutdown(struct i2c_client *i2c) > +{ > + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); > + > + dev_info(&i2c->dev, "shut down device\n"); > + > + /* Enable volume smoothing, disable zero cross. This will cause > + * a quick 40ms ramp to mute on shutdown. > + */ > + regmap_write(max98090->regmap, > + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); > + regmap_write(max98090->regmap, > + M98090_REG_DEVICE_SHUTDOWN, 0x00); > + msleep(40); > +} > + > >

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[alsa-devel] ASoC: Failed to create DAPM debugfs
by Fabio Estevam 10 Apr '15

10 Apr '15

Hi, On a imx6q-hummingboard the following warning is seen when running linux-next 20150408: [ 2.270006] imx-spdif sound-spdif: ASoC: Failed to create DAPM debugfs directory Looking at sound/soc/fsl/imx-spdif.c and sound/soc/fsl/fsl_spdif.c I am not sure how to avoid such warning. Any ideas? Thanks, Fabio Estevam

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[alsa-devel] [PATCH v2] ASoC: fsl: Add the audio interface acronyms in Kconfig text
by Fabio Estevam 09 Apr '15

09 Apr '15

To keep consistency with the other Kconfig entries, use the audio interface acronyms (SSI and SPDIF) in the Kconfig menu text. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix typo in 'interfacce' sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..fa44fc6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI in-tree drivers select it automatically. config SND_SOC_FSL_SSI - tristate "Synchronous Serial Interface module support" + tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO @@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI in-tree drivers select it automatically. config SND_SOC_FSL_SPDIF - tristate "Sony/Philips Digital Interface module support" + tristate "Sony/Philips Digital Interface (SPDIF) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) -- 1.9.1

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Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Daniel Vetter 09 Apr '15

09 Apr '15

On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > connected to my Haswell box. Before applying the patch I tested as a > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > turned out that the first commit that I failed to get audio working was > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > I rerun the bisecting and this time the result was > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > Merge: 2d846c7 a59d719 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Apr 8 11:29:56 2015 +0200 > > Merge branch 'for-linus' into for-next > > Back merge HD-audio quirks to for-next branch, so that we can apply > a couple of more quirks. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Adding Takashi and intel audio folks. -Daniel > > > I don't have this sha1 anywhere. Can you please double-check? > > > > Also to avoid such issues please always add at least the commit subject > > and author, so that I can find it if it's somehow rebased. > > -Daniel > > > > > > > > To test this patch audio functionality I checkout the -nightly version > > > that works for me and apply the patch and test it. I'll come back with > > > the results later on. > > > > > > I had the module option i915.disable_power_well=0 > > > > > > The test routine that I used for audio testing was > > > speaker-test -c 2 -r 48000 -f S16_LE -t pink --device=plughw:0,3 > > > > > > On Tue, 2015-04-07 at 15:52 +0200, Daniel Vetter wrote: > > > > On Tue, Apr 07, 2015 at 12:29:25PM +0300, Mika Kahola wrote: > > > > > Definitely a good idea to check the audio part as well if there is > > > > > a doubt that by changing CD clock the audio would fail. I can check > > > > > this and I'll get back once I have the results. > > > > > > > > We force a full modeset, which should result in an interrupt on the audio > > > > side, which should result in the audio driver re-reading the current > > > > cdclk. If that's no the case it's buggy already. > > > > -Daniel > > > > > > -- > > > Mika Kahola, Intel OTC > > > > > > > > > > > > > -- > Mika Kahola, Intel OTC > > > -- Daniel Vetter Software Engineer, Intel Corporation http://blog.ffwll.ch

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[alsa-devel] [PATCH 1/3] ASoC: Create card debugfs directory earlier
by Lars-Peter Clausen 09 Apr '15

09 Apr '15

Create the card debugfs directory at the begining of the initilization rather then the end as various steps in the initilization sequence will try to register files and sub-directories in the card directory. Fixes: 4e2576bd36a1 ("ASoC: soc-core: initialize debugfs in snd_soc_instantiate_card()") Reported-by: Fabio Estevam <festevam(a)gmail.com> Reported-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2801578..7bd1599 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1555,6 +1555,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } + soc_init_card_debugfs(card); + card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; @@ -1684,8 +1686,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); - soc_init_card_debugfs(card); - return 0; probe_aux_dev_err: @@ -1699,6 +1699,7 @@ card_probe_error: if (card->remove) card->remove(card); + soc_cleanup_card_debugfs(card); snd_card_free(card->snd_card); base_error: -- 1.8.0

2 5

Re: [alsa-devel] [PATCH 0/5] HD-audio widget power-save fixes
by Takashi Iwai 09 Apr '15

09 Apr '15

At Thu, 09 Apr 2015 17:12:40 +0800, hwang4 wrote: > > > > On 2015年04月09日 16:40, Takashi Iwai wrote: > > Hi, > > > > this is a series of small fixes for the recently added HD-audio widget > > power-save feature. Especially the last patch should fix the Realtek > > codec breakage. > > > > The patches are found in topic/hda branch of sound git tree, too. > I tested the topic/hda branch on a machine with alc255 codec, the > internal speaker, headphone, internal mic and external mic all worked > very well. Great, I merged now. Takashi

1 0

[alsa-devel] [PATCH] ASoC: rt286: Restore default in probe
by Bard Liao 09 Apr '15

09 Apr '15

RT286 can't do register reset. If the hardware power is still existing in power off, rt286 will keep the register settings. So, we need to restore the default register value in probe to make sure the cache value is the same as the real register value. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 842cfb9..87af81b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1250,6 +1250,14 @@ static int rt286_i2c_probe(struct i2c_client *i2c, rt286->i2c = i2c; i2c_set_clientdata(i2c, rt286); + /* restore codec default */ + for (i = 0; i < INDEX_CACHE_SIZE; i++) + regmap_write(rt286->regmap, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + for (i = 0; i < ARRAY_SIZE(rt286_reg); i++) + regmap_write(rt286->regmap, rt286_reg[i].reg, + rt286_reg[i].def); + if (pdata) rt286->pdata = *pdata; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [patch] ASoC: Intel: do cast earlier in sst_cdev_tstamp()
by Dan Carpenter 09 Apr '15

09 Apr '15

My static checker complains about these because it looks like the multiply can overflow and then we cast to a larger data type. I don't think this is a problem, but it's also harmless to do the cast earlier so let's silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 718838b..7b50a9d 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -381,7 +381,7 @@ static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, tstamp->copied_total = fw_tstamp.ring_buffer_counter; tstamp->pcm_frames = fw_tstamp.frames_decoded; tstamp->pcm_io_frames = div_u64(fw_tstamp.hardware_counter, - (u64)((stream->num_ch) * SST_GET_BYTES_PER_SAMPLE(24))); + (u64)stream->num_ch * SST_GET_BYTES_PER_SAMPLE(24)); tstamp->sampling_rate = fw_tstamp.sampling_frequency; dev_dbg(dev, "PCM = %u\n", tstamp->pcm_io_frames);

2 1

Re: [alsa-devel] [PATCH 0/4] More aggressive PM for HD-audio
by Takashi Iwai 09 Apr '15

09 Apr '15

At Thu, 26 Mar 2015 20:24:43 +0800, Hui Wang wrote: > > [1 <text/plain; utf-8 (8bit)>] > On 03/21/2015 02:38 PM, Hui Wang wrote: > > On 03/21/2015 12:20 AM, David Henningsson wrote: > >> > >> On 2015-03-18 09:50, Takashi Iwai wrote: > >>> Hi, > >>> > >>> here is a patchset for supporting more aggressive PM for HD-audio. > >>> This allows to change the power state of each widget more dynamically > >>> with jack and stream states. It's activated only when the codec > >>> driver (or via sysfs or f/w patch) sets codec->power_mgmt flag. > >>> > >>> In theory, this should work for the recent Realtek codecs, but > >>> currently I have no machine for test. > >>> > >>> David, could you or your team check whether this works for ALC282 or > >>> such? Just add like: > >>> > >>> --- a/sound/pci/hda/patch_realtek.c > >>> +++ b/sound/pci/hda/patch_realtek.c > >>> @@ -5415,6 +5415,7 @@ static int patch_alc269(struct hda_codec *codec) > >>> > >>> spec = codec->spec; > >>> spec->gen.shared_mic_vref_pin = 0x18; > >>> + codec->power_mgmt = 1; > >>> > >>> snd_hda_pick_fixup(codec, alc269_fixup_models, > >>> alc269_fixup_tbl, alc269_fixups); > >>> > >>> > >>> The patchset is for for-next branch of sound git tree, but they might > >>> be applicable to 4.0-rc (or even older), too. The current patches are > >>> found in topic/hda-power branch. > >> > >> So I hoped to be able to look at this today, but it turns out the > >> machine I was thinking of using for testing has an ALC262 codec, > >> which hardly counts as "new". > >> > >> Hui, is this something you feel like taking on? Otherwise I'll try to > >> talk to someone in Taipei. > >> > > OK， I will look for the machine to do the test next week. > > > > Regards, > > Hui. > > > Sorry for late response, today is my first day in the office back from > vacation, I checked all machines in the Beijing office, none of them has > the ALC282 codec, I will continue to look for the machine from other office. > > And I did the test on the machines with the alc283, alc255, alc292 and > alc269, the testing result were same, there were no sound output from > internal speaker or headphone, and the internal mic or external mic > can't record any sound. The test steps as below: > > 1. power_save_node = 0 > checkout the hda-power branch, build the kernel based on this branch. > Install the kernel to the above machines and boot into the desktop > test internal speaker and internal mic, works very well, plug a headset, > test headphone and external mic, works very well. > run pm_suspend, wait 5 seconds, wakeup the system, redo the above test, > everything works very well. OK, this is expected. The patch shouldn't touch this case. > 2. power_save_node = 1 > enable the power_save_node as below: > @@ -5426,6 +5426,8 @@ static int patch_alc269(struct hda_codec *codec) > > alc_auto_parse_customize_define(codec); > > + codec->power_save_node = 1; > + > if (has_cdefine_beep(codec)) > spec->gen.beep_nid = 0x01; > rebuild the kernel, install the kernel to the above machines and boot > into the desktop > test internal speaker and internal mic, we can play sound to internal > speaker without any errors, but I can't hear any sound from the speaker; > I can use the internal mic to record without errors, but recorded file > did not include any sound pcm (maybe all 0x00 or 0xff) > I plug a headset into the headset jack, the detection works very well, > but I can't hear sound from headphone when play a sound, and I can't use > headset mic to record any sound as well. > > > And I attached 2 alsa-info.txt, one is the power_save_node=0, the other > is the power_save_node=1 Thanks. The alsa-info.sh outputs show no difference but the power state, so the widget attributes seem kept with the power state change, as it seems. Could you give alsa-info.sh output *during* playing with power_save_node=1? Takashi

4 8

[alsa-devel] [PATCH v2] ASoC: fsl_ssi: Do not create debugfs directory
by Fabio Estevam 09 Apr '15

09 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 81c7cfd1b22a0 ("ASoC: Move debugfs registration to the component level") ASoC core code deals with registering debugfs, so we should not be creating the debugfs directory locally in order to avoid the following warning: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Only remove dir creation/remove as suggested by Mark Brown sound/soc/fsl/fsl_ssi.h | 1 - sound/soc/fsl/fsl_ssi_dbg.c | 13 +++---------- 2 files changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..68aef90 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -211,7 +211,6 @@ struct device; #if IS_ENABLED(CONFIG_DEBUG_FS) struct fsl_ssi_dbg { - struct dentry *dbg_dir; struct dentry *dbg_stats; struct { diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..46cea84 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -142,22 +142,15 @@ static const struct file_operations fsl_ssi_stats_ops = { int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) { - ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); - if (!ssi_dbg->dbg_dir) + ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, NULL, + ssi_dbg, &fsl_ssi_stats_ops); + if (!ssi_dbg->dbg_stats) return -ENOMEM; - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, - ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); - if (!ssi_dbg->dbg_stats) { - debugfs_remove(ssi_dbg->dbg_dir); - return -ENOMEM; - } - return 0; } void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) { debugfs_remove(ssi_dbg->dbg_stats); - debugfs_remove(ssi_dbg->dbg_dir); } -- 1.9.1

3 2

Re: [alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Takashi Iwai 09 Apr '15

09 Apr '15

At Wed, 8 Apr 2015 23:24:52 -0700, Matthew Garrett wrote: > > On Wed, Apr 8, 2015 at 10:20 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Wed, 8 Apr 2015 18:53:48 -0700, > > Matthew Garrett wrote: > > > > > > Modern hardware will often have multiple HDA devices, and the desired > > > power saving configuration may vary depending on the codecs attached to > > > each of them. Push the power_save value down to the individual codec > > > structures, keeping the module parameter as a global control mechanism > > > for compatibility purposes. > > > > This patch doesn't apply any longer at all to the recent code, since > > the power saving stuff has been already translated to the standard > > runtime PM. Take a look at linux-next tree. > > Ah! Wonderful! > > > Also we can't break the current behavior controlling the power save > > via option. Many programs rely on this mechanism, thus changing this > > would break its behavior. > > Yeah, I kept that behaviour - anything that touches > /sys/module/snd_hda_intel/parameters/power_save just changes the > settings for all codecs. > > > In the latest code, the common control via power_save module option is > > still there for compatibility, but you can also adjust the value for > > each codec in runtime attribute of each codec sysfs in addition. > > That's ideal - I'll pull down next and rewrite my udev rules. Thanks! Let me know if you find anything broken with the new code. It's pretty new, so any intensive testing is appreciated. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: hda - Always allow access for POWER_STATE verbs
by Takashi Iwai 09 Apr '15

09 Apr '15

The hdac regmap code checks whether the codec is powered on while accessing, but there must be an exception -- the verbs to control the power state. Currently HD-audio driver doesn't access them via regmap, so this patch doesn't fix any current behavior, but it's just for future. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_regmap.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 64876fa357c9..51f1b5c8a91c 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -239,7 +239,7 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) int verb = get_verb(reg); int err; - if (!codec_is_running(codec)) + if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) return -EAGAIN; reg |= (codec->addr << 28); if (is_stereo_amp_verb(reg)) @@ -265,16 +265,16 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) unsigned int verb; int i, bytes, err; - if (!codec_is_running(codec)) - return codec->lazy_cache ? 0 : -EAGAIN; - reg &= ~0x00080000U; /* drop GET bit */ reg |= (codec->addr << 28); + verb = get_verb(reg); + + if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) + return codec->lazy_cache ? 0 : -EAGAIN; if (is_stereo_amp_verb(reg)) return hda_reg_write_stereo_amp(codec, reg, val); - verb = get_verb(reg); if (verb == AC_VERB_SET_PROC_COEF) return hda_reg_write_coef(codec, reg, val); -- 2.3.5

1 0

[alsa-devel] [PATCH 0/5 alsa-lib] amixer fixes for enumerated elements
by Takashi Sakamoto 09 Apr '15

09 Apr '15

The amixer still has issues according to sset operation to enumerated items. The issues are: - Fail to handle omitted value of each channel. - The name of enumerated element is truncated. - Unclear separators for values of channels. This patchset fixes these issues, including a small code refactoring for better reading. Takashi Sakamoto (5): amixer: gather local variables in the beginning of functions amixer: fix a bug to handle omitted value in parameter for enumerated element amixer: expand local storage for item name according to kernel code amixer: arrange validation logic amixer: use the same characters for separator. amixer/amixer.c | 60 ++++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 42 insertions(+), 18 deletions(-) -- 2.1.0

2 10

Re: [alsa-devel] Audio Jack Out does not work
by Dan Carpenter 09 Apr '15

09 Apr '15

So it's 03ad6a8c93b6df2 ('ALSA: hda - Fix "PCM" name being used on one DAC when there are two DACs') which causes the problem? Have you tried to just revert that patch? git show 03ad6a8c93b6df2d65c305b5b5f9474068b45bfb | patch -p1 -R regards, dan carpenter

4 13

[alsa-devel] [PATCH] ALSA: hda - make power_save threshold per-codec
by Matthew Garrett 09 Apr '15

09 Apr '15

Modern hardware will often have multiple HDA devices, and the desired power saving configuration may vary depending on the codecs attached to each of them. Push the power_save value down to the individual codec structures, keeping the module parameter as a global control mechanism for compatibility purposes. Signed-off-by: Matthew Garrett <mjg59(a)coreos.com> --- Documentation/sound/alsa/HD-Audio.txt | 5 +++-- sound/pci/hda/hda_codec.c | 28 ++++++++++++++++++++-------- sound/pci/hda/hda_codec.h | 19 ++++--------------- sound/pci/hda/hda_controller.c | 6 +++--- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 7 +++---- sound/pci/hda/hda_sysfs.c | 30 ++++++++++++++++++++++++++++++ sound/pci/hda/hda_tegra.c | 4 ++-- 8 files changed, 66 insertions(+), 35 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39..99f2fb8 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -569,8 +569,9 @@ Power-Saving The power-saving is a kind of auto-suspend of the device. When the device is inactive for a certain time, the device is automatically turned off to save the power. The time to go down is specified via -`power_save` module option, and this option can be changed dynamically -via sysfs. +`power_save` module option. This option can be changed dynamically +via sysfs, either globally (via /sys/module/snd_hda_*/parameters/power_save) +or per-codec (via the power_save sysfs parameter in the codec directory) The power-saving won't work when the analog loopback is enabled on some codecs. Make sure that you mute all unneeded signal routes when diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2..5e8394a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -895,7 +895,6 @@ int snd_hda_bus_new(struct snd_card *card, bus->private_data = temp->private_data; bus->pci = temp->pci; bus->modelname = temp->modelname; - bus->power_save = temp->power_save; bus->ops = temp->ops; mutex_init(&bus->cmd_mutex); @@ -1436,8 +1435,9 @@ static void snd_hda_codec_dev_release(struct device *dev) * Returns 0 if successful, or a negative error code. */ int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) + unsigned int codec_addr, + struct hda_codec **codecp, + int power_save) { struct hda_codec *codec; char component[31]; @@ -1495,6 +1495,7 @@ int snd_hda_codec_new(struct hda_bus *bus, INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; + codec->power_save = power_save; codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM @@ -5112,23 +5113,34 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) codec->power_transition = 0; } -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - /* Transition to powered down */ static void __snd_hda_power_down(struct hda_codec *codec) { if (!codec->power_on || codec->power_count || codec->power_transition) return; - if (power_save(codec)) { + if (codec->power_save >= 0) { codec->power_transition = -1; /* avoid reentrance */ queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); + msecs_to_jiffies(codec->power_save * 1000)); } } /** + * + * snd_hda_power_save_set - set the power-down time for the codec + * + * @codec: HD-audio codec + * @value: seconds to wait before powering down the codec + */ +void snd_hda_power_save_set(struct hda_codec *codec, int value) +{ + codec->power_save = value; + snd_hda_power_save(codec, 0, false); +} +EXPORT_SYMBOL_GPL(snd_hda_power_save_set); + +/** * snd_hda_power_save - Power-up/down/sync the codec * @codec: HD-audio codec * @delta: the counter delta to change diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f..faaab58 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -106,7 +106,6 @@ struct hda_bus_template { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; }; @@ -123,7 +122,6 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -398,6 +396,7 @@ struct hda_codec { struct snd_array jacks; #endif + int power_save; int depop_delay; /* depop delay in ms, -1 for default delay time */ /* fix-up list */ @@ -423,7 +422,7 @@ enum { int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); + struct hda_codec **codecp, int power_save); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); @@ -588,9 +587,11 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM +void snd_hda_power_save_set(struct hda_codec *codec, int value); void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); void snd_hda_update_power_acct(struct hda_codec *codec); #else +static inline void snd_hda_power_save_set(struct hda_codec *codec, int value) {} static inline void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) {} #endif @@ -634,18 +635,6 @@ static inline void snd_hda_power_down(struct hda_codec *codec) snd_hda_power_save(codec, -1, false); } -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 17c2637..4bb13ae 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1813,7 +1813,7 @@ static int get_jackpoll_interval(struct azx *chip) /* Codec initialization */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to) + int power_save) { struct hda_bus_template bus_temp; int c, codecs, err; @@ -1827,7 +1827,6 @@ int azx_codec_create(struct azx *chip, const char *model, bus_temp.ops.attach_pcm = azx_attach_pcm_stream; bus_temp.ops.bus_reset = azx_bus_reset; #ifdef CONFIG_PM - bus_temp.power_save = power_save_to; bus_temp.ops.pm_notify = azx_power_notify; #endif #ifdef CONFIG_SND_HDA_DSP_LOADER @@ -1886,7 +1885,8 @@ int azx_codec_create(struct azx *chip, const char *model, for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(chip->bus, c, &codec, + power_save); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10f..d187121 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -45,7 +45,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, - int *power_save_to); + int power_save); int azx_codec_configure(struct azx *chip); int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a8a1e14..d51779a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -174,7 +174,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -187,7 +186,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +static int power_save = -1; #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -754,7 +753,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) if (!chip->bus || chip->disabled) continue; list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_power_save_set(c, power_save); } mutex_unlock(&card_list_lock); return 0; @@ -1898,7 +1897,7 @@ static int azx_probe_continue(struct azx *chip) /* create codec instances */ err = azx_codec_create(chip, model[dev], azx_max_codecs[chip->driver_type], - power_save_addr); + power_save); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a..69bbb0b 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -44,8 +44,37 @@ static ssize_t power_off_acct_show(struct device *dev, return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct)); } +static ssize_t power_save_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + + return sprintf(buf, "%d\n", jiffies_to_msecs(codec->power_save)); +} + +static ssize_t power_save_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct hda_codec *codec = dev_get_drvdata(dev); + unsigned long val; + int err = kstrtol(buf, 0, &val); + + if (err < 0) + return err; + + if (val < -1) + return -EINVAL; + + snd_hda_power_save_set(codec, val); + return count; +} + static DEVICE_ATTR_RO(power_on_acct); static DEVICE_ATTR_RO(power_off_acct); +static DEVICE_ATTR_RW(power_save); + #endif /* CONFIG_PM */ #define CODEC_INFO_SHOW(type) \ @@ -759,6 +788,7 @@ static struct attribute *hda_dev_attrs[] = { #ifdef CONFIG_PM &dev_attr_power_on_acct.attr, &dev_attr_power_off_acct.attr, + &dev_attr_power_save.attr, #endif #ifdef CONFIG_SND_HDA_RECONFIG &dev_attr_init_verbs.attr, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f..69b76fc 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -82,7 +82,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +static int power_save = -1; #endif /* @@ -503,7 +503,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_codec_create(chip, NULL, 0, power_save); if (err < 0) goto out_free; -- 2.3.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: tegra_alc5632: Use card DAPM context to access widgets
by Lars-Peter Clausen 08 Apr '15

08 Apr '15

The dapm field of the snd_soc_codec struct will eventually be removed (replaced with the DAPM context from the component embedded inside the CODEC). Replace its usage with the card's DAPM context. The idea is that DAPM is hierarchical and with the card at the root it is possible to access widgets from other contexts through the card context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/tegra/tegra_alc5632.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 6dcd06a..ba272e2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -101,9 +101,6 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, @@ -118,7 +115,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_alc5632_hp_jack_gpio); } - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "MICBIAS1"); return 0; } -- 1.8.0

2 4

[alsa-devel] [PATCH v4 0/5] ALSA: jack: Refactoring for jack kctls
by Jie Yang 08 Apr '15

08 Apr '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this new implementation, both HDA and soc jack kctls works fine, they are compatible for old HDA jack kctls, too. For soc, a. snd_jack_new() with NULL jack_kctl; b. for each pin, call snd_jack_add_kctls() to add one kctl(it will call snd_ctl_add() to add kctl to card); For hda, a. call snd_jack_kctl_new()(in __snd_hda_jack_add_kctl()) to new a jack_kctl(also new kctl); and use snd_hda_ctl_add to add kctl to card; b. new a hda_jack_tbl, appoint jack_tbl->jack_kctl = jack_kctl; c. for non-phantom jack, call snd_jack_new() with this jack_kctl, then the jack_kctl will be added to the new created jack; Changes in v4: 1. use snd_ctl_find_id() to get avaliable index; 2. add initial kctl for snd_jack_new(compatible for HDA); 3. add struct snd_jack_kctl * field to struct hda_jack_tbl; 4. new kctls for soc jack during jack pins creating. 5. add a patch to remove exporting snd_kctl_jack_new(). Changes in v3: 1. replace bit index with bit mask in jack_kctl; 2. add exception for SND_JACK_HEADSET and SND_JACK_AVOUT, only create one jack kctl for these two combo jacks, respectively. 3. add NULL check, mem kfree, and fix some potential risk. Change in v2: 1. define jack_kctl struct, and put jack kctl related stuff there; 2. add a patch to remove the existing controls for HDA jack. Jie Yang (5): ALSA: jack: create jack kcontrols for every jack input ALSA: jack: add a parameter to pass kctl for snd_jack_new ALSA: hda - Update to use the new jack kctls method ASoC: jack: create kctls according to jack pins info ALSA: jack: remove export snd_kctl_jack_new() include/sound/jack.h | 25 ++++++- sound/core/ctljack.c | 1 - sound/core/jack.c | 152 +++++++++++++++++++++++++++++++++++++++- sound/pci/hda/hda_jack.c | 52 +++++++------- sound/pci/hda/hda_jack.h | 4 +- sound/pci/oxygen/xonar_wm87x6.c | 2 +- sound/soc/soc-jack.c | 3 +- 7 files changed, 200 insertions(+), 39 deletions(-) -- 1.9.1

6 20

[alsa-devel] [PATCH] ASoC: rt5645: Adds push button support for rt5650
by Bard Liao 08 Apr '15

08 Apr '15

This patch exports rt5650_set_button_detect function for machine driver to support rt5650 headset button detection function. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 147 +++++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5645.h | 7 ++- 2 files changed, 146 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 69528ae..e547a994 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2368,6 +2368,8 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, static int rt5645_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_PREPARE: if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { @@ -2398,8 +2400,9 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); - snd_soc_update_bits(codec, RT5645_GEN_CTRL1, - RT5645_DIG_GATE_CTRL, 0); + if (!rt5645->en_button_func) + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, 0); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2 | @@ -2414,6 +2417,37 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void rt5645_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + if (enable) { + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x8); + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x8000); + snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + } else { + snd_soc_update_bits(codec, RT5650_4BTN_IL_CMD2, 0x8000, 0x0); + snd_soc_update_bits(codec, RT5645_INT_IRQ_ST, 0x8, 0x0); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC L power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "ADC R power"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "LDO2"); + snd_soc_dapm_disable_pin_unlocked(&codec->dapm, + "Mic Det Power"); + snd_soc_dapm_sync_unlocked(&codec->dapm); + } +} + static int rt5645_jack_detect(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); @@ -2449,17 +2483,24 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec) val = snd_soc_read(codec, RT5645_IN1_CTRL3) & 0x7; dev_dbg(codec->dev, "val = %d\n", val); - if (val == 1 || val == 2) - jack_type = SND_JACK_HEADSET; - else - jack_type = SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(&codec->dapm, "micbias1"); snd_soc_dapm_disable_pin(&codec->dapm, "micbias2"); if (rt5645->pdata.jd_mode == 0) snd_soc_dapm_disable_pin(&codec->dapm, "LDO2"); snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power"); snd_soc_dapm_sync(&codec->dapm); + + if (val == 1 || val == 2) { + jack_type = SND_JACK_HEADSET; + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, true); + } else { + jack_type = SND_JACK_HEADPHONE; + } + + } else { /* jack out */ + if (rt5645->en_button_func) + rt5645_enable_push_button_irq(codec, false); } snd_soc_jack_report(rt5645->hp_jack, jack_type, SND_JACK_HEADPHONE); @@ -2480,6 +2521,42 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(rt5645_set_jack_detect); +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int ret; + + rt5645->jack = jack; + rt5645->codec_irq.gpiod_dev = codec->dev; + rt5645->codec_irq.name = "rt5650 irq"; + rt5645->codec_irq.report = SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3; + rt5645->codec_irq.debounce_time = 150; + rt5645->codec_irq.wake = true; + rt5645->codec_irq.data = (struct rt5645_priv *)rt5645; + rt5645->codec_irq.jack_status_check = rt5645_irq_detection; + + ret = snd_soc_jack_add_gpios(rt5645->jack, 1, + &rt5645->codec_irq); + if (ret) { + dev_err(codec->dev, "Adding jack GPIO failed\n"); + return ret; + } + + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); + regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, + RT5645_HP_CB_MASK, RT5645_HP_CB_PU); + regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + rt5645->en_button_func = true; + + return 0; +} +EXPORT_SYMBOL_GPL(rt5650_set_button_detect); + static void rt5645_jack_detect_work(struct work_struct *work) { struct rt5645_priv *rt5645 = @@ -2498,6 +2575,62 @@ static irqreturn_t rt5645_irq(int irq, void *data) return IRQ_HANDLED; } +static int rt5645_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5650_4BTN_IL_CMD1); + pr_debug("val=0x%x\n", val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5650_4BTN_IL_CMD1, val); + + return btn_type; +} + +static int rt5645_irq_detection(void *data) +{ + struct rt5645_priv *rt5645 = (struct rt5645_priv *)data; + struct snd_soc_jack *jack = &rt5645->jack; + int btn_type, report = SND_JACK_HEADSET; + + btn_type = rt5645_button_detect(rt5645->codec); + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + dev_err(rt5645->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + /* keep the jack status */ + report = jack->status; + break; + } + + pr_debug("report=%x\n", report); + + return report; +} + static int rt5645_probe(struct snd_soc_codec *codec) { struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index db78e94..bf29d8b 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2184,7 +2184,9 @@ struct rt5645_priv { struct i2c_client *i2c; struct snd_soc_jack *hp_jack; struct snd_soc_jack *mic_jack; + struct snd_soc_jack *jack; struct delayed_work jack_detect_work; + struct snd_soc_jack_gpio codec_irq; int codec_type; int sysclk; @@ -2196,9 +2198,12 @@ struct rt5645_priv { int pll_src; int pll_in; int pll_out; + + bool en_button_func; }; int rt5645_set_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack); - +int rt5650_set_button_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); #endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu <xjq(a)rock-chips.com> Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- Changes in v3: - modify the shutdown function before remove(..) - fix the `Serien-cc` Changes in v2: - remove the dev_info(..) - fix the comment style - add the max98090_i2c_shutdown() in remove() fuction sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c..3e33ef2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2605,8 +2605,24 @@ err_enable: return ret; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { .acpi_match_table = ACPI_PTR(max98090_acpi_match), }, .probe = max98090_i2c_probe, + .shutdown = max98090_i2c_shutdown, .remove = max98090_i2c_remove, .id_table = max98090_i2c_id, }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Remove custom fsl_ssi_debugfs_create/remove()
by Fabio Estevam 08 Apr '15

08 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 81c7cfd1b22a0 ("ASoC: Move debugfs registration to the component level") ASoC core code deals with registering debugfs, so we should remove the custom fsl_ssi_debugfs_create/remove() to avoid the following warning: fsl-ssi-dai 2028000.ssi: ASoC: Failed to create component debugfs directory Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 11 +---------- sound/soc/fsl/fsl_ssi.h | 13 ------------- sound/soc/fsl/fsl_ssi_dbg.c | 22 ---------------------- 3 files changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..b8cdb8f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1411,10 +1411,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) } } - ret = fsl_ssi_debugfs_create(&ssi_private->dbg_stats, &pdev->dev); - if (ret) - goto error_irq; - /* * If codec-handle property is missing from SSI node, we assume * that the machine driver uses new binding which does not require @@ -1441,7 +1437,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); dev_err(&pdev->dev, "failed to register platform: %d\n", ret); - goto error_sound_card; + goto error_irq; } done: @@ -1451,9 +1447,6 @@ done: return 0; -error_sound_card: - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - error_irq: snd_soc_unregister_component(&pdev->dev); @@ -1468,8 +1461,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) { struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); - fsl_ssi_debugfs_remove(&ssi_private->dbg_stats); - if (ssi_private->pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 5065105..4767df1 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -241,10 +241,6 @@ struct fsl_ssi_dbg { void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *ssi_dbg, u32 sisr); -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev); - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg); - #else struct fsl_ssi_dbg { @@ -254,15 +250,6 @@ static inline void fsl_ssi_dbg_isr(struct fsl_ssi_dbg *stats, u32 sisr) { } -static inline int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, - struct device *dev) -{ - return 0; -} - -static inline void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ -} #endif /* ! IS_ENABLED(CONFIG_DEBUG_FS) */ #endif diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 5469ffb..118064c 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -139,25 +139,3 @@ static const struct file_operations fsl_ssi_stats_ops = { .llseek = seq_lseek, .release = single_release, }; - -int fsl_ssi_debugfs_create(struct fsl_ssi_dbg *ssi_dbg, struct device *dev) -{ - ssi_dbg->dbg_dir = debugfs_create_dir(dev_name(dev), NULL); - if (!ssi_dbg->dbg_dir) - return -ENOMEM; - - ssi_dbg->dbg_stats = debugfs_create_file("stats", S_IRUGO, - ssi_dbg->dbg_dir, ssi_dbg, &fsl_ssi_stats_ops); - if (!ssi_dbg->dbg_stats) { - debugfs_remove(ssi_dbg->dbg_dir); - return -ENOMEM; - } - - return 0; -} - -void fsl_ssi_debugfs_remove(struct fsl_ssi_dbg *ssi_dbg) -{ - debugfs_remove(ssi_dbg->dbg_stats); - debugfs_remove(ssi_dbg->dbg_dir); -} -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: remove unused functions
by Sudip Mukherjee 08 Apr '15

08 Apr '15

these functions were never called by anyone. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/intel/atom/sst/sst_pvt.c | 24 ------------------------ 1 file changed, 24 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index 2d74249..adb32fe 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -111,30 +111,6 @@ int sst_wait_interruptible(struct intel_sst_drv *sst_drv_ctx, } -static unsigned long long read_shim_data(struct intel_sst_drv *sst, int addr) -{ - unsigned long long val = 0; - - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - val = sst_shim_read64(sst->shim, addr); - break; - } - return val; -} - -static void write_shim_data(struct intel_sst_drv *sst, int addr, - unsigned long long data) -{ - switch (sst->dev_id) { - case SST_MRFLD_PCI_ID: - case SST_BYT_ACPI_ID: - sst_shim_write64(sst->shim, addr, (u64) data); - break; - } -} - /* * sst_wait_timeout - wait on event for timeout * -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: bebob: fix to processing in big-endian machine for sending cue
by Takashi Sakamoto 08 Apr '15

08 Apr '15

Some M-Audio devices require to receive bootup command just after powering on, while codes in BeBoB driver doesn't work properly in big-endian machine because the command should be aligned by little-endian. This commit fixes this bug. This fix should go to stable kernel. Cc: Takayuki Shiroma <t.shiroma.oki(a)gmail.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_maudio.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c index a422aaa..9ee25a6 100644 --- a/sound/firewire/bebob/bebob_maudio.c +++ b/sound/firewire/bebob/bebob_maudio.c @@ -96,10 +96,10 @@ int snd_bebob_maudio_load_firmware(struct fw_unit *unit) struct fw_device *device = fw_parent_device(unit); int err, rcode; u64 date; - __be32 cues[3] = { - MAUDIO_BOOTLOADER_CUE1, - MAUDIO_BOOTLOADER_CUE2, - MAUDIO_BOOTLOADER_CUE3 + __le32 cues[3] = { + cpu_to_le32(MAUDIO_BOOTLOADER_CUE1), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE2), + cpu_to_le32(MAUDIO_BOOTLOADER_CUE3) }; /* check date of software used to build */ -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Create AFG sysfs node at last
by Takashi Iwai 08 Apr '15

08 Apr '15

... so that user-space can know that the whole nodes have been created. Unfortunately, this can't be implemented easily in race-free way, so it's a kind of compromise. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_sysfs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index b358d5157802..18aea43d230d 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -313,12 +313,12 @@ static void widget_tree_free(struct hdac_device *codec) if (!tree) return; + free_widget_node(tree->afg, &widget_afg_group); if (tree->nodes) { for (p = tree->nodes; *p; p++) free_widget_node(*p, &widget_node_group); kfree(tree->nodes); } - free_widget_node(tree->afg, &widget_afg_group); if (tree->root) kobject_put(tree->root); kfree(tree); @@ -362,13 +362,6 @@ static int widget_tree_create(struct hdac_device *codec) if (!tree->root) return -ENOMEM; - if (codec->afg) { - err = add_widget_node(tree->root, codec->afg, - &widget_afg_group, &tree->afg); - if (err < 0) - return err; - } - tree->nodes = kcalloc(codec->num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL); if (!tree->nodes) @@ -381,6 +374,13 @@ static int widget_tree_create(struct hdac_device *codec) return err; } + if (codec->afg) { + err = add_widget_node(tree->root, codec->afg, + &widget_afg_group, &tree->afg); + if (err < 0) + return err; + } + kobject_uevent(tree->root, KOBJ_CHANGE); return 0; } -- 2.3.4

1 1

[alsa-devel] [PATCH v2] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

To fix pop noise when shutdown,the pop noise during shutdown is the pmic cutoff power of codec without any notice. Signed-off-by: jay.xu <xjq(a)rock-chips.com> Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Serien-cc: linux-kernel(a)vger.kernel.org Serien-cc: devicetree(a)vger.kernel.org Serien-cc: dianders(a)chromium.org --- Changes in v2: - remove the dev_info(..) - fix the comment style - add the max98090_i2c_shutdown() in remove() fuction sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b112b1c..e59013a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2607,10 +2607,26 @@ err_enable: static int max98090_i2c_remove(struct i2c_client *client) { + max98090_i2c_shutdown(client); snd_soc_unregister_codec(&client->dev); return 0; } +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + /* + * Enable volume smoothing, disable zero cross. This will cause + * a quick 40ms ramp to mute on shutdown. + */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + #ifdef CONFIG_PM static int max98090_runtime_resume(struct device *dev) { @@ -2697,6 +2713,7 @@ static struct i2c_driver max98090_i2c_driver = { }, .probe = max98090_i2c_probe, .remove = max98090_i2c_remove, + .shutdown = max98090_i2c_shutdown, .id_table = max98090_i2c_id, }; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: Intel: Fix a buffer overflow issue
by Jie Yang 08 Apr '15

08 Apr '15

0day robot reported a buffer overflow issue: ... sound/soc/intel/haswell/sst-haswell-pcm.c:1107 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 sound/soc/intel/haswell/sst-haswell-pcm.c:1109 hsw_pcm_probe() error: buffer\ overflow 'hsw_dais' 4 <= 4 ... Fix it by initializing the index(i) to correct value. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 157b3a6..23ae040 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -1103,7 +1103,7 @@ static int hsw_pcm_probe(struct snd_soc_platform *platform) return 0; err: - for (;i >= 0; i--) { + for (--i; i >= 0; i--) { if (hsw_dais[i].playback.channels_min) snd_dma_free_pages(&priv_data->dmab[i][0]); if (hsw_dais[i].capture.channels_min) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/1] ASoC: max98090: add shutdown callback for max98090
by Caesar Wang 08 Apr '15

08 Apr '15

Tested on veyron devices,play music then shutdown device,here carefully to speaker or headphone. Caesar Wang (1): ASoC: max98090: add shutdown callback for max98090 sound/soc/codecs/max98090.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) -- 1.9.1

3 6

Re: [alsa-devel] Add delay in alc283_init
by Takashi Iwai 08 Apr '15

08 Apr '15

At Wed, 8 Apr 2015 08:56:48 +0000, Kailang wrote: > > Hi Takashi, > > This patch will make more stable for getting pin sense during resume state. > This is tested for Chrome Box machine. > Many Thanks. Thanks, applied now with Cc to stable. Takashi > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-add-delay-before-get-pin-sense.patch <application/octet-stream (base64)>] >

1 0

[alsa-devel] 1st draft: ALSA Json Gateway and HTML5 UI
by Fulup Ar Foll 07 Apr '15

07 Apr '15

Hi, I would need some help to finish a json version of amixer. Porting cget is done, but for cset I have an issue with "snd_hctl_find_elem (handle, id)". I'm not sure what's the right method to pass it. I've tried every options I could imagine. But if it works on Mint/32bit, it miserably fail on OpenSuse/64. Furthermore amixer.c as this strange comment /* FIXME: Remove it when hctl find works ok !!! */ and they do not use snd_hctl_find_elem. I posted a small C code to highlight the issue at [1]https://gist.github.com/fulup-bzh/891e5627be325150337f My question is simple, did I miss something ? should I move to lower level API ? Note: While the painting for my json-gateway is not dry, I nevertheless pushed a initial running version with cget at [2]https://github.com/fulup-bzh/AlsaJsonGateway I also have a small HTML5/UI at [3]http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2 While UI code is very dependant on each soundcard [in my case Scarlett/Focurite] the Json-Gateway is 100% independent of the board. If anything one have an idea, please let me know. Fulup References 1. https://gist.github.com/fulup-bzh/891e5627be325150337f 2. https://github.com/fulup-bzh/AlsaJsonGateway 3. http://breizhme.net/alsajson/mixers/ajg#/scarlett?card=2

2 1

[alsa-devel] [PATCH][ASoC]Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver
by Maciej S. Szmigiero 07 Apr '15

07 Apr '15

Add ability to remove rate constraints from generic ASoC AC'97 CODEC driver via passed platform data, make it selectable in config. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. There is no change in behavior when no platform data is passed. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/include/sound/soc-ac97.h b/include/sound/soc-ac97.h new file mode 100644 index 0000000..ceb4e2f --- /dev/null +++ b/include/sound/soc-ac97.h @@ -0,0 +1,17 @@ +/* + * Platform data for generic ASoC AC97 CODEC driver. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#ifndef __LINUX_SND__SOC_AC97_H +#define __LINUX_SND__SOC_AC97_H + +struct snd_soc_ac97_codec_platform_data { + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + +#endif /* __LINUX_SND__SOC_AC97_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0bddd92..92d6d39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -210,8 +210,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..c3fb845 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -20,9 +20,16 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> +#include <sound/soc-ac97.h> #include <sound/initval.h> #include <sound/soc.h> +struct ac97_private { + struct snd_ac97 *ac97; + bool playback_rate_constrained; + bool capture_rate_constrained; +}; + static const struct snd_soc_dapm_widget ac97_widgets[] = { SND_SOC_DAPM_INPUT("RX"), SND_SOC_DAPM_OUTPUT("TX"), @@ -33,22 +40,53 @@ static const struct snd_soc_dapm_route ac97_routes[] = { { "TX", NULL, "AC97 Playback" }, }; +static const unsigned int default_rates[] = { + 8000, 11025, 22050, 44100, 48000 +}; + +static const struct snd_pcm_hw_constraint_list default_rate_constraints = { + .count = ARRAY_SIZE(default_rates), + .list = default_rates, +}; + +static int ac97_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + bool constrained; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + constrained = ac97->playback_rate_constrained; + else + constrained = ac97->capture_rate_constrained; + + if (!constrained) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + else + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &default_rate_constraints); + + return 0; +} + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; - return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); + return snd_ac97_set_rate(ac97->ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { + .startup = ac97_startup, .prepare = ac97_prepare, }; @@ -58,20 +96,21 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; static int ac97_soc_probe(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97; + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); + struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; int ret; @@ -83,31 +122,28 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return ret; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); - ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97); + ret = snd_ac97_mixer(ac97_bus, &ac97_template, &ac97->ac97); if (ret < 0) return ret; - snd_soc_codec_set_drvdata(codec, ac97); - return 0; } #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - snd_ac97_suspend(ac97); + snd_ac97_suspend(ac97->ac97); return 0; } static int ac97_soc_resume(struct snd_soc_codec *codec) { + struct ac97_private *ac97 = snd_soc_codec_get_drvdata(codec); - struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); - - snd_ac97_resume(ac97); + snd_ac97_resume(ac97->ac97); return 0; } @@ -129,6 +165,27 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { static int ac97_probe(struct platform_device *pdev) { + struct ac97_private *ac97 = + devm_kzalloc(&pdev->dev, sizeof(struct ac97_private), + GFP_KERNEL); + struct snd_soc_ac97_codec_platform_data *pdata = + pdev->dev.platform_data; + + if (!ac97) + return -ENOMEM; + + if (pdata) { + ac97->playback_rate_constrained = + pdata->playback_rate_constrained; + ac97->capture_rate_constrained = + pdata->capture_rate_constrained; + } else { + ac97->playback_rate_constrained = 1; + ac97->capture_rate_constrained = 1; + } + + platform_set_drvdata(pdev, ac97); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_ac97, &ac97_dai, 1); }

2 6

[alsa-devel] [PATCH 00/14] Fix fallout from per-user struct clk patches
by Russell King - ARM Linux 07 Apr '15

07 Apr '15

Sorry for posting this soo close to the 4.1 merge window, I had completely forgotten about this chunk of work I did earlier this month. The per-user struct clk patches rather badly broke clkdev and various other places. This was reported, but was forgotten about. Really, the per-user clk stuff should've been reverted, but we've lived with it far too long for that. So, our only other option is to now rush these patches into 4.1 and hope for the best. The series cleans up quite a number of places too... arch/arm/mach-davinci/da850.c | 1 + arch/arm/mach-lpc32xx/clock.c | 5 +- arch/arm/mach-omap1/board-nokia770.c | 2 +- arch/arm/mach-omap2/clkt2xxx_virt_prcm_set.c | 12 +--- arch/arm/mach-omap2/omap_device.c | 24 +++----- arch/arm/mach-pxa/eseries.c | 1 + arch/arm/mach-pxa/lubbock.c | 1 + arch/arm/mach-pxa/tosa.c | 1 + arch/arm/plat-orion/common.c | 6 +- arch/sh/kernel/cpu/sh4a/clock-sh7734.c | 3 +- arch/sh/kernel/cpu/sh4a/clock-sh7757.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-sh7785.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-sh7786.c | 4 +- arch/sh/kernel/cpu/sh4a/clock-shx3.c | 4 +- drivers/clk/clk-s2mps11.c | 4 +- drivers/clk/clkdev.c | 83 ++++++++++++++++++++-------- drivers/media/platform/omap3isp/isp.c | 18 ------ drivers/media/platform/omap3isp/isp.h | 1 - include/linux/clk.h | 27 ++++----- include/linux/clkdev.h | 6 +- include/media/omap3isp.h | 6 -- sound/soc/sh/migor.c | 3 +- 22 files changed, 108 insertions(+), 112 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

10 39

[alsa-devel] [PATCH] ALSA: hda_intel: add AZX_DCAPS_I915_POWERWELL for SKL and BSW
by libin.yang＠intel.com 07 Apr '15

07 Apr '15

From: Libin Yang <libin.yang(a)intel.com> HDMI/DP codec on SKL/BSW is in the power well. The power well must be turned on before probing the HDMI/DP codec. This is a temporary patch, which will power on the powerwell by adding AZX_DCAPS_I915_POWERWELL for SKL and BSW. After restructuring and new flag is added, this patch will be reverted. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 15a8299..e1c2105 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -297,8 +297,12 @@ enum { AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\ AZX_DCAPS_SNOOP_TYPE(SCH)) +#define AZX_DCAPS_INTEL_BRASWELL \ + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL) + #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -1991,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Braswell */ { PCI_DEVICE(0x8086, 0x2284), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL }, /* ICH6 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_INTEL_ICH }, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

2 5

[alsa-devel] [PATCH v2 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v1: - Corrected email address on two of the patches Thanks, Charles sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

1 3

[alsa-devel] bug
by Nikita N. 07 Apr '15

07 Apr '15

is any programmer listening here? -- http://www.fastmail.com - A no graphics, no pop-ups email service

10 35

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix ruledata setup in davinci_mcasp_startup
by Peter Ujfalusi 07 Apr '15

07 Apr '15

Passing &mcasp->ruledata[dir] to snd_pcm_hw_rule_add() is not correct since commit: 7b3d165a2821 ASoC: davinci-mcasp: Index ruledata in drvdata with substream->stream now sets up the struct based on the substream->stream (0 or 1) while we pass a pointer which we take with dir (1 or 2). This will lead kernel crash. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 0b6b1b286201..bb4b78eada58 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1128,6 +1128,8 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + struct davinci_mcasp_ruledata *ruledata = + &mcasp->ruledata[substream->stream]; u32 max_channels = 0; int i, dir; @@ -1149,7 +1151,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->serial_dir[i] == dir) max_channels++; } - mcasp->ruledata[substream->stream].serializers = max_channels; + ruledata->serializers = max_channels; max_channels *= mcasp->tdm_slots; /* * If the already active stream has less channels than the calculated @@ -1172,12 +1174,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { int ret; - mcasp->ruledata[substream->stream].mcasp = mcasp; + ruledata->mcasp = mcasp; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1185,7 +1187,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (ret) @@ -1193,7 +1195,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, davinci_mcasp_hw_rule_channels, - &mcasp->ruledata[dir], + ruledata, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_FORMAT, -1); if (ret) -- 2.3.5

2 1

[alsa-devel] [PATCH 1/4] ASoC: wm8804: Add support for hardware reset line
by Charles Keepax 07 Apr '15

07 Apr '15

It is best to use the physical reset if it is available. This patch adds support for a GPIO controlled physical reset for the chip. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8804.c | 26 +++++++++++++++++++++----- 1 files changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index f44da83..cc168c4 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/gpio/consumer.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/of_device.h> @@ -61,6 +62,8 @@ struct wm8804_priv { struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; int mclk_div; + + struct gpio_desc *reset; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -182,7 +185,7 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct wm8804_priv *wm8804) +static int wm8804_soft_reset(struct wm8804_priv *wm8804) { return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } @@ -586,6 +589,14 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + wm8804->reset = devm_gpiod_get_optional(dev, "wlf,reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm8804->reset)) { + ret = PTR_ERR(wm8804->reset); + dev_err(dev, "Failed to get reset line: %d\n", ret); + return ret; + } + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; @@ -620,6 +631,9 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) return ret; } + if (wm8804->reset) + gpiod_set_value_cansleep(wm8804->reset, 1); + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); if (ret < 0) { dev_err(dev, "Failed to read device ID: %d\n", ret); @@ -648,10 +662,12 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) } dev_info(dev, "revision %c\n", id1 + 'A'); - ret = wm8804_reset(wm8804); - if (ret < 0) { - dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; + if (!wm8804->reset) { + ret = wm8804_soft_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } } ret = snd_soc_register_codec(dev, &soc_codec_dev_wm8804, -- 1.7.2.5

2 5

[alsa-devel] [PATCH] ALSA: hda_intel: add AZX_DCAPS_I915_POWERWELL for skl
by libin.yang＠intel.com 07 Apr '15

07 Apr '15

From: Libin Yang <libin.yang(a)intel.com> HDMI/DP codec on SKL is in the power well. The power well must be turned on before probing the HDMI/DP codec. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 15a8299..4744a96 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -298,7 +298,8 @@ enum { AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_SKYLAKE \ - (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG) + (AZX_DCAPS_INTEL_PCH | AZX_DCAPS_SEPARATE_STREAM_TAG |\ + AZX_DCAPS_I915_POWERWELL) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ -- 1.9.1

4 19

[alsa-devel] steinberg ur44 - soundcard not connecting, acl problem (systemd?)
by James Miller 06 Apr '15

06 Apr '15

Hi, I've just bought a steinberg ur44. It is reported as having worked in an earlier kernel to the 3.18 rt custom/3.19 low latency generic that I am using. The low latency kernel is generic but the realtime is compiled with the alsa quirks as described here: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074120.html On plugging it in, the card is recognised by udev but is disconnected after seemingly being set up. The same error seems to be occurring when I start the machine with the card plugged in. There seems to be an ACL error reported in journalctl. I include the appropriate lines from the udevadm and from journalctl udevadm: sudo udevadm monitor monitor will print the received events for: UDEV - the event which udev sends out after rule processing KERNEL - the kernel uevent KERNEL[2741.779546] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5 (usb) KERNEL[2741.779705] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.0 (usb) KERNEL[2741.781720] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1 (usb) KERNEL[2741.787500] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) KERNEL[2741.787611] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) KERNEL[2741.787813] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) KERNEL[2741.787933] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) KERNEL[2741.788073] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) KERNEL[2741.788189] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) KERNEL[2741.789228] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) KERNEL[2741.789294] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) KERNEL[2741.789315] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) KERNEL[2741.789365] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) KERNEL[2741.789418] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) KERNEL[2741.789436] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) KERNEL[2741.789487] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) KERNEL[2741.789503] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) KERNEL[2741.789556] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.2 (usb) KERNEL[2741.789577] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.3 (usb) KERNEL[2741.790043] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.4 (usb) UDEV [2741.810128] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5 (usb) UDEV [2741.810172] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.3 (usb) UDEV [2741.810195] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.2 (usb) UDEV [2741.810217] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1 (usb) UDEV [2741.810237] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) UDEV [2741.810260] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) UDEV [2741.810279] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0c (sound) UDEV [2741.810301] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.4 (usb) UDEV [2741.815664] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) UDEV [2741.815903] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/pcmC3D0p (sound) UDEV [2741.820639] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) UDEV [2741.821530] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midi3 (sound) UDEV [2741.823891] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) UDEV [2741.823931] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/dmmidi3 (sound) UDEV [2741.826596] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) UDEV [2741.827398] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/midiC3D0 (sound) UDEV [2741.837736] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) UDEV [2741.839478] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3 (sound) UDEV [2741.840171] remove /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3 (sound) UDEV [2742.842156] add /devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.0 (usb) journactl (on hotplug add): Apr 06 19:26:09 super-R720 kernel: usb 1-5: USB disconnect, device number 4 Apr 06 19:26:09 super-R720 systemd-udevd[309]: error opening USB device 'descriptors' file Apr 06 19:26:11 super-R720 kernel: usb 1-5: new high-speed USB device number 5 using ehci-pci Apr 06 19:26:12 super-R720 kernel: usb 1-5: New USB device found, idVendor=0499, idProduct=150e Apr 06 19:26:12 super-R720 kernel: usb 1-5: New USB device strings: Mfr=1, Product=2, SerialNumber=0 Apr 06 19:26:12 super-R720 kernel: usb 1-5: Product: Steinberg UR44 Apr 06 19:26:12 super-R720 kernel: usb 1-5: Manufacturer: Yamaha Corporation Apr 06 19:26:12 super-R720 kernel: snd-usb-audio: probe of 1-5:1.1 failed with error -16 Apr 06 19:26:12 super-R720 mtp-probe[3550]: checking bus 1, device 5: "/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5" Apr 06 19:26:12 super-R720 mtp-probe[3550]: bus: 1, device: 5 was not an MTP device Apr 06 19:26:12 super-R720 systemd-udevd[3547]: error opening ATTR{/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3/ Apr 06 19:26:12 super-R720 systemd-udevd[3560]: Failed to apply ACL on /dev/dmmidi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3560]: Failed to apply ACL on /dev/dmmidi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3559]: Failed to apply ACL on /dev/midi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3559]: Failed to apply ACL on /dev/midi3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3558]: Failed to apply ACL on /dev/snd/midiC3D0: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3558]: Failed to apply ACL on /dev/snd/midiC3D0: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3549]: Failed to apply ACL on /dev/snd/pcmC3D0c: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3549]: Failed to apply ACL on /dev/snd/pcmC3D0c: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3548]: Failed to apply ACL on /dev/snd/pcmC3D0p: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3548]: Failed to apply ACL on /dev/snd/pcmC3D0p: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3547]: Failed to apply ACL on /dev/snd/controlC3: No such file or directory Apr 06 19:26:12 super-R720 systemd-udevd[3547]: Failed to apply ACL on /dev/snd/controlC3: No such file or directory journalctl -b: systemd-udevd[416]: error opening ATTR{/sys/devices/pci0000:00/0000:00:1a.7/usb1/1-5/1-5:1.1/sound/card3/controlC3/. I can compile a kernel reasonably well, and can understand and program in c, c++, but am not well versed in lower level usb hardware coding. I'd be grateful for any help, many thanks! Many thanks -- James Stewart Miller Bsc(hons) Psych.

1 1

[alsa-devel] [asoc:topic/intel 37/37] sound/soc/intel/atom/sst/sst_drv_interface.c:277:19: sparse: cast removes address space of expression
by kbuild test robot 06 Apr '15

06 Apr '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: b97169da06992ef04081e66ed22bbdb23dbf6610 commit: b97169da06992ef04081e66ed22bbdb23dbf6610 [37/37] ASoC: Intel: create atom folder and move atom platform files in reproduce: # apt-get install sparse git checkout b97169da06992ef04081e66ed22bbdb23dbf6610 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/intel/atom/sst/sst_drv_interface.c:277:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:278:17: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:285:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: sparse: incorrect type in assignment (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: expected void [noderef] <asn:2>*addr sound/soc/intel/atom/sst/sst_drv_interface.c:285:14: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:372:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:373:17: got void * >> sound/soc/intel/atom/sst/sst_drv_interface.c:560:19: sparse: cast removes address space of expression >> sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: expected void const volatile [noderef] <asn:2>*src sound/soc/intel/atom/sst/sst_drv_interface.c:561:25: got void * -- >> sound/soc/intel/atom/sst/sst_loader.c:55:26: sparse: incorrect type in argument 1 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:55:26: expected void [noderef] <asn:2>*to sound/soc/intel/atom/sst/sst_loader.c:55:26: got void *dst >> sound/soc/intel/atom/sst/sst_loader.c:55:31: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:55:31: expected void const *from sound/soc/intel/atom/sst/sst_loader.c:55:31: got void const [noderef] <asn:2>*src >> sound/soc/intel/atom/sst/sst_loader.c:220:43: sparse: incorrect type in argument 2 (different address spaces) sound/soc/intel/atom/sst/sst_loader.c:220:43: expected void *destn sound/soc/intel/atom/sst/sst_loader.c:220:43: got void [noderef] <asn:2>* -- >> sound/soc/intel/atom/sst/sst_pvt.c:114:20: sparse: symbol 'read_shim_data' was not declared. Should it be static? >> sound/soc/intel/atom/sst/sst_pvt.c:127:6: sparse: symbol 'write_shim_data' was not declared. Should it be static? Please review and possibly fold the followup patch. vim +277 sound/soc/intel/atom/sst/sst_drv_interface.c 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 271 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 272 /* update bytes sent */ 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 273 stream->cumm_bytes += bytes; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 274 dev_dbg(dev, "bytes copied %d inc by %ld\n", stream->cumm_bytes, bytes); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 275 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 276 memcpy_fromio(&fw_tstamp, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @277 ((void *)(ctx->mailbox + ctx->tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @278 +(str_id * sizeof(fw_tstamp))), 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 279 sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 280 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 281 fw_tstamp.bytes_copied = stream->cumm_bytes; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 282 dev_dbg(dev, "bytes sent to fw %llu inc by %ld\n", 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 283 fw_tstamp.bytes_copied, bytes); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 284 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @285 addr = ((void *)(ctx->mailbox + ctx->tstamp)) + 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 286 (str_id * sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 287 offset = offsetof(struct snd_sst_tstamp, bytes_copied); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 288 sst_shim_write(addr, offset, fw_tstamp.bytes_copied); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 289 return 0; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 290 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 291 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 292 static int sst_cdev_set_metadata(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 293 unsigned int str_id, struct snd_compr_metadata *metadata) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 294 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 295 int retval = 0; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 296 struct stream_info *str_info; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 297 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 298 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 299 dev_dbg(dev, "set metadata for stream %d\n", str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 300 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 301 str_info = get_stream_info(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 302 if (!str_info) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 303 return -EINVAL; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 304 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 305 dev_dbg(dev, "pipe id = %d\n", str_info->pipe_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 306 retval = sst_prepare_and_post_msg(ctx, str_info->task_id, IPC_CMD, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 307 IPC_IA_SET_STREAM_PARAMS_MRFLD, str_info->pipe_id, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 308 sizeof(*metadata), metadata, NULL, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 309 true, true, true, false); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 310 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 311 return retval; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 312 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 313 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 314 static int sst_cdev_stream_pause(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 315 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 316 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 317 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 318 return sst_pause_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 319 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 320 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 321 static int sst_cdev_stream_pause_release(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 322 unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 323 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 324 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 325 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 326 return sst_resume_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 327 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 328 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 329 static int sst_cdev_stream_start(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 330 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 331 struct stream_info *str_info; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 332 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 333 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 334 str_info = get_stream_info(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 335 if (!str_info) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 336 return -EINVAL; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 337 str_info->prev = str_info->status; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 338 str_info->status = STREAM_RUNNING; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 339 return sst_start_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 340 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 341 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 342 static int sst_cdev_stream_drop(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 343 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 344 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 345 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 346 return sst_drop_stream(ctx, str_id); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 347 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 348 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 349 static int sst_cdev_stream_drain(struct device *dev, unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 350 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 351 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 352 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 353 return sst_drain_stream(ctx, str_id, false); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 354 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 355 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 356 static int sst_cdev_stream_partial_drain(struct device *dev, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 357 unsigned int str_id) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 358 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 359 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 360 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 361 return sst_drain_stream(ctx, str_id, true); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 362 } 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 363 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 364 static int sst_cdev_tstamp(struct device *dev, unsigned int str_id, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 365 struct snd_compr_tstamp *tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 366 { 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 367 struct snd_sst_tstamp fw_tstamp = {0,}; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 368 struct stream_info *stream; 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 369 struct intel_sst_drv *ctx = dev_get_drvdata(dev); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 370 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 371 memcpy_fromio(&fw_tstamp, 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @372 ((void *)(ctx->mailbox + ctx->tstamp) 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 @373 +(str_id * sizeof(fw_tstamp))), 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 374 sizeof(fw_tstamp)); 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 375 7adab122 sound/soc/intel/sst/sst_drv_interface.c Vinod Koul 2014-10-30 376 stream = get_stream_info(ctx, str_id); :::::: The code at line 277 was first introduced by commit :::::: 7adab122a57c5ade8efc2e4de67c72b084c31cda ASoC: Intel: sst - add compressed ops handling :::::: TO: Vinod Koul <vinod.koul(a)intel.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center http://lists.01.org/mailman/listinfo/kbuild Intel Corporation

2 2

[alsa-devel] [PATCH v3 0/5] Intel SST driver restructure
by Jie Yang 06 Apr '15

06 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Changes in v3: Fix ATT43256.config 0day build issue. Changes in v2: Add "Reviewed-by" and "Acked-by" items from each platforms' maintainer. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 2 +- sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_ipc.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_loader.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 2 +- sound/soc/intel/{ => atom}/sst/sst_stream.c | 2 +- sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 4 +-- sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 4 +-- sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 4 +-- sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 77 insertions(+), 73 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (99%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (99%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (99%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (98%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (99%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (99%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

2 6

[alsa-devel] [PATCH] ASoC: wm5102: Remove set of volume update bits for output 3R
by Charles Keepax 06 Apr '15

06 Apr '15

The earpiece on wm5102 is mono, thus there is no output 3R. Don't toggle the volume update bits for this output, although worth noting that doing so had no negative effects it is just redundant. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 6d0fe0a..0c6d1bc 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1861,7 +1861,6 @@ static unsigned int wm5102_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_2L, ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_DAC_DIGITAL_VOLUME_3L, - ARIZONA_DAC_DIGITAL_VOLUME_3R, ARIZONA_DAC_DIGITAL_VOLUME_4L, ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_DAC_DIGITAL_VOLUME_5L, -- 1.7.2.5

2 1

[alsa-devel] [PATCH/alsa-lib] amixer: fix 'cset' operation to set several channels
by Takashi Sakamoto 06 Apr '15

06 Apr '15

For an enumerated element with several channels, when given comma-separated string values, 'cset' operation sets these values just to the first channel, instead of corresponding channels. This commit adds missing-increment to fix this bug. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- amixer/amixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 36c92eb..e9fe735 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -1290,8 +1290,10 @@ static int sset_enum(snd_mixer_elem_t *elem, unsigned int argc, char **argv) if (snd_mixer_selem_set_enum_item(elem, item++, ival) >= 0) check_flag = 1; /* skip separators */ - while (*ptr == ',' || isspace(*ptr)) + while (*ptr == ',' || isspace(*ptr)) { ptr++; + chn++; + } } } return check_flag; -- 2.1.0

2 3

[alsa-devel] [PATCH] ALSA: emu10k1: add Audigy 5/Rx
by Michael Gernoth 05 Apr '15

05 Apr '15

The Audigy 5/Rx is essentially an Audigy 4 behind a PLX PCIe- bridge with an additional TOSLINK output. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emu10k1_main.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index b4458a6..8e510e9 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1325,6 +1325,22 @@ static int snd_emu10k1_dev_free(struct snd_device *device) } static struct snd_emu_chip_details emu_chip_details[] = { + /* Audigy 5/Rx SB1550 */ + /* Tested by michael(a)gernoth.net 28 Mar 2015 */ + /* DSP: CA10300-IAT LF + * DAC: Cirrus Logic CS4382-KQZ + * ADC: Philips 1361T + * AC97: Sigmatel STAC9750 + * CA0151: None + */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10241102, + .driver = "Audigy2", .name = "SB Audigy 5/Rx [SB1550]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .adc_1361t = 1, /* 24 bit capture instead of 16bit */ + .ac97_chip = 1} , /* Audigy4 (Not PRO) SB0610 */ /* Tested by James(a)superbug.co.uk 4th April 2006 */ /* A_IOCFG bits -- 2.1.4

2 1

[alsa-devel] SoundBlaster X-fi HD SPDIF fix
by Richard Körber 05 Apr '15

05 Apr '15

Hi! I own a Creative USB SoundBlaster X-fi HD and tried to use its SPDIF optical output. However it didn't work with ALSA out of the box. The reason is that for SPDIF output device #1 must be used. The fix is: In /usr/share/alsa/cards/USB-Audio.conf, change the iec958_device block like this: USB-Audio.pcm.iec958_device { "USB Sound Blaster HD" 1 } Digital output works fine for me after that. BTW, I tried this line in /etc/asound.conf cards.USB-Audio.pcm.iec958_device."USB Sound Blaster HD" 1 but it didn't have any effect. Why was that? Thank you! Regards Richard Körber

2 8

[alsa-devel] [PATCH 0/16] fix error return code
by Julia Lawall 05 Apr '15

05 Apr '15

The complate semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ // identify a function that returns a negative return value at least once. f(...) { ... when any ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ... when any } @r exists@ identifier ret,ok.f,fn; expression e1,e2,e3,e4,e5,e6,x; statement S,S1; position p1,p2,p3; @@ // identify a case where the return variable is set to a non-negative value // and then returned in error-handling code f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1$ when != &ret when any ( if (<+... ret = e5 ...+>) S1 | if (<+... &ret ...+>) S1 | if@p2(<+...x = fn(...)...+>) { ... when != ret = e6 when forall return@p3 ret; } | break; | x = fn(...) ... when != $ret = e4\|ret++\|ret--\|ret+=e4\|ret-=e4$ when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(<+...$x != 0\|x < 0\|x == NULL\|IS_ERR(x)$...+>) { ... when != ret = e2 when forall return@p3 ret; } ) ) ... when any } @printer depends on r@ position p; identifier ok.f,pr; constant char [] c; @@ f(...) { <...pr@p(...,c,...)...> } @bad0 exists@ identifier r.ret,ok.f,g != {ERR_PTR,IS_ERR}; position p != printer.p; @@ f(...) { ... when any g@p(...,ret,...) ... when any } @bad depends on !bad0 exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ // ignore the above if there is some path where the variable is set to // something else ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1\|&ret$ ... when any if@p2(...) S2 @bad1 depends on !bad0 && !bad exists@ position r.p2; statement S2; identifier r.ret; expression e1; constant c; @@ ret = -c ... when != $ret = e1\|ret++\|ret--\|ret+=e1\|ret-=e1$ when != &ret when any if@p2(...) S2 @bad2 depends on !bad0 && !bad && !bad1 exists@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; constant c; @@ // likewise ignore it if there has been an intervening return ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when != if (...) { ... return -c; } when any if@p2(...) S2 @script:python depends on !bad0 && !bad && !bad1 && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 2

Re: [alsa-devel] [PATCH] ALSA: hda/realtek - Support for microphone in Dell's XPS 13 9343 with ALC288.
by Takashi Iwai 05 Apr '15

05 Apr '15

At Thu, 2 Apr 2015 12:33:40 +0100, me(a)jdiez.me wrote: > > From: Jose Diez <me(a)jdiez.me> > > Original patch by Kailang Yang did not compile on 4.0-rc5. It was > referencing a member field `core` in `struct hda_codec` in order to > access the `vendor_id`. This has always been in the codec structure > itself. > > This patch fixes the build and is verified to work properly on 4.0-rc5, > running actual hardware. > > Signed-off-by: Jose Diez <me(a)jdiez.me> I prefer splitting the Dell-specific part as its own quirk, as mentioned in the review of Kailang's patch. Then it'll be applicable to both 4.1 and older kernels. thanks, Takashi > --- > sound/pci/hda/patch_realtek.c | 122 ++++++++++++++++++++++++++++++++++++++++++ > 1 file changed, 122 insertions(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7438213..a8fc2c8 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3583,6 +3583,14 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > WRITE_COEF(0x32, 0x42a3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x76, 0x000e), > WRITE_COEF(0x6c, 0x2400), > @@ -3613,6 +3621,10 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3642,6 +3654,14 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > WRITE_COEF(0x26, 0x008c), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0), > + UPDATE_COEF(0x56, 0x0006, 0), > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), > + UPDATE_COEF(0x66, 0x0008, 0x0008), > + UPDATE_COEF(0x67, 0x2000, 0x2000), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x19, 0xa208), > WRITE_COEF(0x2e, 0xacf0), > @@ -3674,6 +3694,13 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > alc_process_coef_fw(codec, coef0233); > snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0x000c, 0); > + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > + alc_process_coef_fw(codec, coef0286); > + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > + break; > case 0x10ec0292: > snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > alc_process_coef_fw(codec, coef0292); > @@ -3709,6 +3736,14 @@ static void alc_headset_mode_default(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x4f, 0xfcc0, 0xc400), /* Set to TRS type */ > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x76, 0x000e), > WRITE_COEF(0x6c, 0x2400), > @@ -3737,6 +3772,11 @@ static void alc_headset_mode_default(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + break; > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3765,6 +3805,13 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x6b, 0xd429), > WRITE_COEF(0x76, 0x0008), > @@ -3791,6 +3838,12 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xd400); > + msleep(300); > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3819,6 +3872,13 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > WRITE_COEF(0x32, 0x4ea3), > {} > }; > + static struct coef_fw coef0286[] = { > + UPDATE_COEF(0x50, 0x2000, 0x2000), > + UPDATE_COEF(0x56, 0x0006, 0x0006), > + UPDATE_COEF(0x66, 0x0008, 0), > + UPDATE_COEF(0x67, 0x2000, 0), > + {} > + }; > static struct coef_fw coef0292[] = { > WRITE_COEF(0x6b, 0xe429), > WRITE_COEF(0x76, 0x0008), > @@ -3845,6 +3905,12 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > case 0x10ec0283: > alc_process_coef_fw(codec, coef0233); > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_update_coef_idx(codec, 0x4f, 0xfcc0, 0xe400); > + msleep(300); > + alc_process_coef_fw(codec, coef0286); > + break; > case 0x10ec0292: > alc_process_coef_fw(codec, coef0292); > break; > @@ -3869,6 +3935,11 @@ static void alc_determine_headset_type(struct hda_codec *codec) > conteol) */ > {} > }; > + static struct coef_fw coef0286[] = { > + //UPDATE_COEF(0x4f, 0xfcc0, 0x5400), /* Combo Jack auto detect */ > + UPDATE_COEF(0x4f, 0xfcc0, 0xd400), > + {} > + }; > static struct coef_fw coef0293[] = { > UPDATE_COEF(0x4a, 0x000f, 0x0008), /* Combo Jack auto detect */ > WRITE_COEF(0x45, 0xD429), /* Set to ctia type */ > @@ -3896,6 +3967,13 @@ static void alc_determine_headset_type(struct hda_codec *codec) > val = alc_read_coef_idx(codec, 0x46); > is_ctia = (val & 0x0070) == 0x0070; > break; > + case 0x10ec0286: > + case 0x10ec0288: > + alc_process_coef_fw(codec, coef0286); > + msleep(350); > + val = alc_read_coef_idx(codec, 0x50); > + is_ctia = (val & 0x0070) == 0x0070; > + break; > case 0x10ec0292: > alc_write_coef_idx(codec, 0x6b, 0xd429); > msleep(300); > @@ -3990,8 +4068,15 @@ static void alc_update_headset_jack_cb(struct hda_codec *codec, > struct hda_jack_callback *jack) > { > struct alc_spec *spec = codec->spec; > + int present; > spec->current_headset_type = ALC_HEADSET_TYPE_UNKNOWN; > snd_hda_gen_hp_automute(codec, jack); > + present = spec->gen.hp_jack_present ? 0x40 : 0; > + if (codec->bus->pci->subsystem_vendor == 0x1028 && > + codec->vendor_id == 0x10ec0288) > + /* Headset Mic enable and disable, ony for Dell Dino */ > + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, > + present); > } > > static void alc_probe_headset_mode(struct hda_codec *codec) > @@ -4418,6 +4503,8 @@ enum { > ALC286_FIXUP_HP_GPIO_LED, > ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, > ALC280_FIXUP_HP_DOCK_PINS, > + ALC286_FIXUP_DELL1_MIC_NO_PRESENCE, > + ALC286_FIXUP_DELL_XPS_13_GPIO6, > }; > > static const struct hda_fixup alc269_fixups[] = { > @@ -4906,6 +4993,27 @@ static const struct hda_fixup alc269_fixups[] = { > .chained = true, > .chain_id = ALC280_FIXUP_HP_GPIO4 > }, > + [ALC286_FIXUP_DELL1_MIC_NO_PRESENCE] = { > + .type = HDA_FIXUP_PINS, > + .v.pins = (const struct hda_pintbl[]) { > + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ > + { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */ > + { } > + }, > + .chained = true, > + .chain_id = ALC269_FIXUP_HEADSET_MODE > + }, > + [ALC286_FIXUP_DELL_XPS_13_GPIO6] = { > + .type = HDA_FIXUP_VERBS, > + .v.verbs = (const struct hda_verb[]) { > + {0x01, AC_VERB_SET_GPIO_MASK, 0x40}, > + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x40}, > + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, > + { } > + }, > + .chained = true, > + .chain_id = ALC286_FIXUP_DELL1_MIC_NO_PRESENCE > + }, > }; > > static const struct snd_pci_quirk alc269_fixup_tbl[] = { > @@ -5132,6 +5240,13 @@ static const struct hda_model_fixup alc269_fixup_models[] = { > {0x1b, 0x411111f0}, \ > {0x1e, 0x411111f0} > > +#define ALC286_STANDARD_PINS \ > + {0x17, 0x411111f0}, \ > + {0x18, 0x411111f0}, \ > + {0x19, 0x411111f0}, \ > + {0x1a, 0x411111f0}, \ > + {0x1e, 0x411111f0} > + > #define ALC290_STANDARD_PINS \ > {0x12, 0x99a30130}, \ > {0x13, 0x40000000}, \ > @@ -5317,6 +5432,13 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > {0x19, 0x03a11020}, > {0x1d, 0x40e00001}, > {0x21, 0x0321101f}), > + SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC286_FIXUP_DELL_XPS_13_GPIO6, > + ALC286_STANDARD_PINS, > + {0x12, 0x90a60120}, > + {0x13, 0x40000000}, > + {0x14, 0x90170110}, > + {0x1d, 0x4076832d}, > + {0x21, 0x0321101f}), > SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, > ALC290_STANDARD_PINS, > {0x14, 0x411111f0}, > -- > 2.3.5 >

1 0

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 05 Apr '15

05 Apr '15

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

2 20

[alsa-devel] [PATCH] sound: Use const struct ac97_quirk
by Joe Perches 05 Apr '15

05 Apr '15

Use const to reduce data by ~3Kb. Signed-off-by: Joe Perches <joe(a)perches.com> --- include/sound/ac97_codec.h | 4 +++- sound/pci/ac97/ac97_codec.c | 3 ++- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/via82xx.c | 2 +- 7 files changed, 10 insertions(+), 7 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index d315a08..0e9d75b 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -608,7 +608,9 @@ struct ac97_quirk { int type; /* quirk type above */ }; -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override); +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, + const char *override); int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate); /* diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5bca1a3..82259ca 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -2902,7 +2902,8 @@ static int apply_quirk_str(struct snd_ac97 *ac97, const char *typestr) * Return: Zero if successful, or a negative error code on failure. */ -int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, const char *override) +int snd_ac97_tune_hardware(struct snd_ac97 *ac97, + const struct ac97_quirk *quirk, const char *override) { int result; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 850a8c9..66ddd98 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -747,7 +747,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip) snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read); } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x11d4, /* AD */ .subdevice = 0x1889, /* AD1889 */ diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d5f15c9..42a20c8 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1390,7 +1390,7 @@ static irqreturn_t snd_atiixp_interrupt(int irq, void *dev_id) * ac97 mixer section */ -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x103c, .subdevice = 0x006b, diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 802c33f..963b912 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -43,7 +43,7 @@ static char *ac97_quirk; module_param(ac97_quirk, charp, 0444); MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds."); -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { #if 0 /* Not yet confirmed if all 5536 boards are HP only */ { .subvendor = PCI_VENDOR_ID_AMD, diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 2c5484e..749069a 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -1795,7 +1795,7 @@ static struct ac97_pcm ac97_pcm_defs[] = { }, }; -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x0e11, .subdevice = 0x000e, diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8622283..3dd038b 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1812,7 +1812,7 @@ static void snd_via82xx_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97 = NULL; } -static struct ac97_quirk ac97_quirks[] = { +static const struct ac97_quirk ac97_quirks[] = { { .subvendor = 0x1106, .subdevice = 0x4161,

2 1

[alsa-devel] linux-next - strange audio bug
by Valdis Kletnieks 04 Apr '15

04 Apr '15

Next-20150310 works OK, next-20150324 has the bug. Dell Latitude E6530 laptop, lspci tells me: 01:00.1 Audio device: NVIDIA Corporation GF108 High Definition Audio Controller (rev a1) Subsystem: Dell Device 0535 ... Kernel driver in use: snd_hda_intel Symptoms: Audio works perfectly at boot, and for some amount of time after. But somewhere around 6-8 hours of uptime, it gets into a strange state where I'll get no audio... and then it will cut in for all of 2 seconds or so (fairly consistent, that part), and then cut back out for anywhere from 10 seconds to a minute, then another 2 seconds of sound.. lather rinse repeat. Before I start the thankless task of bisecting this (which means I'll be able to do only 1, *maybe* 2 steps a day), does this ring any bells?

4 3

[alsa-devel] [PATCH] ALSA: hda: fix possible null dereference
by Sudip Mukherjee 04 Apr '15

04 Apr '15

we are dereferencing pcm first then checking pcm. instead now lets put them in same if condition so that pcm is checked first. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/hda/hda_codec.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 145cae7..7df18e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3717,9 +3717,8 @@ static int add_std_chmaps(struct hda_codec *codec) struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (pcm->own_chmap) - continue; - if (!pcm || !hinfo->substreams) + if (!pcm || pcm->own_chmap || + !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix subsystem ID read regression
by David Henningsson 04 Apr '15

04 Apr '15

A regression was introduced in 7639a06c23c7d4cda3: if AC_PAR_SUBSYSTEM_ID reads as zero, one should retry using AC_VERB_GET_SUBSYSTEM_ID. This seems to hit many codecs (my own laptop included), and causes quirks for some machines not to apply correctly. Reported-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/hda/hdac_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6e8ee1d..53b6b95 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -91,7 +91,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->power_caps = snd_hdac_read_parm(codec, fg, AC_PAR_POWER_STATE); /* reread ssid if not set by parameter */ - if (codec->subsystem_id == -1) + if (codec->subsystem_id == -1 || codec->subsystem_id == 0) snd_hdac_read(codec, fg, AC_VERB_GET_SUBSYSTEM_ID, 0, &codec->subsystem_id); -- 1.9.1

2 1

Re: [alsa-devel] [RFC] ALSA: usb-audio: reduce "cannot get freq at ep" spew
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:36:28 +0000, Eric Wong wrote: > > Eric Wong <normalperson(a)yhbt.net> wrote: > > I also had another generic patch prepared before I noticed Joe's > > quirk addition for the MS Lifecam HD-5000. > > This is the generic patch I prepared before I noticed Joe's quirk > addition (applies cleanly to 3.19 and 4.0-rc6): > > ------------------------------8<--------------------------- > Subject: [PATCH] ALSA: usb-audio: reduce "cannot get freq at ep" spew > > If a device fails to support reading the sample rate, it will likely > fail again and there is no point in logging the message every time the > sample rate is set. > > This reduces dmesg noise when using the Benchmark DAC1 PRE as a USB > sound card. Hm, this shuts up all the rest for the device, so it also hides if the message comes from multiple EPs. Takashi > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > --- > sound/usb/clock.c | 7 +++++-- > sound/usb/usbaudio.h | 3 ++- > 2 files changed, 7 insertions(+), 3 deletions(-) > > diff --git a/sound/usb/clock.c b/sound/usb/clock.c > index 03fed66..512fe12 100644 > --- a/sound/usb/clock.c > +++ b/sound/usb/clock.c > @@ -307,8 +307,11 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, > USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, > UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, > data, sizeof(data))) < 0) { > - dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > - iface, fmt->altsetting, ep); > + if (!chip->no_get_freq) { > + dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", > + iface, fmt->altsetting, ep); > + chip->no_get_freq = 1; > + } > return 0; /* some devices don't support reading */ > } > > diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h > index 91d0380..be65671 100644 > --- a/sound/usb/usbaudio.h > +++ b/sound/usb/usbaudio.h > @@ -43,7 +43,8 @@ struct snd_usb_audio { > unsigned int in_pm:1; > unsigned int autosuspended:1; > unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ > - > + unsigned int no_get_freq:1; > + > int num_interfaces; > int num_suspended_intf; > > -- > EW >

1 0

Re: [alsa-devel] [PATCH] ALSA: usb-audio: don't try to get Benchmark DAC1 sample rate
by Takashi Iwai 04 Apr '15

04 Apr '15

At Tue, 31 Mar 2015 07:34:05 +0000, Eric Wong wrote: > > Adding this quirk allows us to avoid the noisy > "cannot get freq at ep 0x1" message in dmesg output every time > playback starts. > > This ought to affect other Benchmark DAC1 variations using the same > "Microchip Technology, Inc." chip as well, but I have only tested > with the "Pre" variant. > > Signed-off-by: Eric Wong <normalperson(a)yhbt.net> > Cc: Joe Turner <joe(a)oampo.co.uk> > Cc: Takashi Iwai <tiwai(a)suse.de> Applied, thanks. Takashi > --- > I also had another generic patch prepared before I noticed Joe's > quirk addition for the MS Lifecam HD-5000. > > sound/usb/quirks.c | 9 +++++++-- > 1 file changed, 7 insertions(+), 2 deletions(-) > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > index 753a47d..9a28365 100644 > --- a/sound/usb/quirks.c > +++ b/sound/usb/quirks.c > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > { > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > - return chip->usb_id == USB_ID(0x045E, 0x076D); > + /* devices which do not support reading the sample rate. */ > + switch (chip->usb_id) { > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > + return true; > + } > + return false; > } > > /* Marantz/Denon USB DACs need a vendor cmd to switch > -- > EW >

1 0

[alsa-devel] Support Dell headset mode for ALC256
by Kailang 04 Apr '15

04 Apr '15

Hi Takashi, Dell has new platform with ALC256. Attach patch is support headset mode for ALC256. Thanks. BR, Kailang

2 1

Re: [alsa-devel] [PATCH 76/86] sound/pci: use uapi/linux/pci_ids.h directly
by Takashi Iwai 04 Apr '15

04 Apr '15

At Sun, 29 Mar 2015 15:42:59 +0200, Michael S. Tsirkin wrote: > > Header moved from linux/pci_ids.h to uapi/linux/pci_ids.h, > use the new header directly so we can drop > the wrapper in include/linux/pci_ids.h. > > Signed-off-by: Michael S. Tsirkin <mst(a)redhat.com> Do we need this change in the driver code at all...? I thought include/uapi is already included in the include path. Takashi > --- > sound/pci/ctxfi/xfi.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c > index f2f3277..7e26876 100644 > --- a/sound/pci/ctxfi/xfi.c > +++ b/sound/pci/ctxfi/xfi.c > @@ -11,7 +11,7 @@ > #include <linux/init.h> > #include <linux/pci.h> > #include <linux/moduleparam.h> > -#include <linux/pci_ids.h> > +#include <uapi/linux/pci_ids.h> > #include <linux/module.h> > #include <sound/core.h> > #include <sound/initval.h> > -- > MST >

1 0

[alsa-devel] [PATCH v1.1 07/14] media: omap3isp: remove unused clkdev
by Laurent Pinchart 04 Apr '15

04 Apr '15

From: Russell King <rmk+kernel(a)arm.linux.org.uk> No merged platform supplies xclks via platform data. As we want to slightly change the clkdev interface, rather than fixing this unused code, remove it instead. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> Signed-off-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> --- drivers/media/platform/omap3isp/isp.c | 24 ------------------------ drivers/media/platform/omap3isp/isp.h | 1 - include/media/omap3isp.h | 6 ------ 3 files changed, 31 deletions(-) In case this would be helpful, I've rebased the original patch on top of the linuxtv master branch scheduled for merge in v4.1. diff --git a/drivers/media/platform/omap3isp/isp.c b/drivers/media/platform/omap3isp/isp.c index ff51c4f..18d0a87 100644 --- a/drivers/media/platform/omap3isp/isp.c +++ b/drivers/media/platform/omap3isp/isp.c @@ -304,7 +304,6 @@ static struct clk *isp_xclk_src_get(struct of_phandle_args *clkspec, void *data) static int isp_xclk_init(struct isp_device *isp) { - struct isp_platform_data *pdata = isp->pdata; struct device_node *np = isp->dev->of_node; struct clk_init_data init; unsigned int i; @@ -335,26 +334,6 @@ static int isp_xclk_init(struct isp_device *isp) xclk->clk = clk_register(NULL, &xclk->hw); if (IS_ERR(xclk->clk)) return PTR_ERR(xclk->clk); - - /* When instantiated from DT we don't need to register clock - * aliases. - */ - if (np) - continue; - - if (!pdata || (pdata->xclks[i].con_id == NULL && - pdata->xclks[i].dev_id == NULL)) - continue; - - xclk->lookup = kzalloc(sizeof(*xclk->lookup), GFP_KERNEL); - if (xclk->lookup == NULL) - return -ENOMEM; - - xclk->lookup->con_id = pdata->xclks[i].con_id; - xclk->lookup->dev_id = pdata->xclks[i].dev_id; - xclk->lookup->clk = xclk->clk; - - clkdev_add(xclk->lookup); } if (np) @@ -376,9 +355,6 @@ static void isp_xclk_cleanup(struct isp_device *isp) if (!IS_ERR(xclk->clk)) clk_unregister(xclk->clk); - - if (xclk->lookup) - clkdev_drop(xclk->lookup); } } diff --git a/drivers/media/platform/omap3isp/isp.h b/drivers/media/platform/omap3isp/isp.h index 431224e..e579943 100644 --- a/drivers/media/platform/omap3isp/isp.h +++ b/drivers/media/platform/omap3isp/isp.h @@ -132,7 +132,6 @@ enum isp_xclk_id { struct isp_xclk { struct isp_device *isp; struct clk_hw hw; - struct clk_lookup *lookup; struct clk *clk; enum isp_xclk_id id; diff --git a/include/media/omap3isp.h b/include/media/omap3isp.h index 0f0c08b..048f8f9 100644 --- a/include/media/omap3isp.h +++ b/include/media/omap3isp.h @@ -150,13 +150,7 @@ struct isp_platform_subdev { struct isp_bus_cfg *bus; }; -struct isp_platform_xclk { - const char *dev_id; - const char *con_id; -}; - struct isp_platform_data { - struct isp_platform_xclk xclks[2]; struct isp_platform_subdev *subdevs; void (*set_constraints)(struct isp_device *isp, bool enable); }; -- Regards, Laurent Pinchart

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Re: [alsa-devel] [PATCH 1/6] ASoC: jz4740: Remove Makefile entry for removed file
by Greg Kroah-Hartman 03 Apr '15

03 Apr '15

On Fri, Mar 27, 2015 at 10:51:43AM +0000, Markos Chandras wrote: > On 04/22/2014 09:46 PM, Lars-Peter Clausen wrote: > > Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > jz4740-pcm.c file, but neglected to remove the Makefile entries. > > > > Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") > > Reported-by: kbuild test robot <fengguang.wu(a)intel.com> > > Reported-by: Ralf Baechle <ralf(a)linux-mips.org> > > Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> > > --- > > sound/soc/jz4740/Makefile | 2 -- > > 1 file changed, 2 deletions(-) > > > > diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile > > index be873c1..d32c540 100644 > > --- a/sound/soc/jz4740/Makefile > > +++ b/sound/soc/jz4740/Makefile > > @@ -1,10 +1,8 @@ > > # > > # Jz4740 Platform Support > > # > > -snd-soc-jz4740-objs := jz4740-pcm.o > > snd-soc-jz4740-i2s-objs := jz4740-i2s.o > > > > -obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o > > obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o > > > > # Jz4740 Machine Support > > > Hi, > > This patch (eebdec044e82) is present in 3.15-rc1 but the build failure > that was introduced by 0406a40a0 ("ASoC: jz4740: Use the generic > dmaengine PCM driver") is present in 3.14-rc1 so 3.14 is still broken. > > Greg, would it be possible to cherry pick eebdec044e82 ("ASoC: jz4740: > Remove Makefile entry for removed file") to 3.14 stable branch? It seems > to apply without conflicts. Now applied, thanks. greg k-h

1 0

[alsa-devel] dummy codec + simple card combination
by Belisko Marek 03 Apr '15

03 Apr '15

Hi, I'm trying to add audio support for gta04 (for gsm + bt and fm audio). All audio sources are clock masters so basically connection is dummy codecs -> cpu dai (mcbsp ports). For implementing such thing I can see 2 options: 1. simple card driver which can easily describe cpu dai interface but dummy codec cannot because there are no DT bindings (there was sever attempts to add DT bindings for dummy-codec but AFAIK none was accepted) This turns it to only solution to create gta04 soundcard which will get from DT cpu dai links and use in card driver dummy codecs directly. Is my assumption correct? I would like to get some feedback before wasting my time with implementing something which cannot be pushed mainline. Thanks for all suggestions. BR, marek -- as simple and primitive as possible ------------------------------------------------- Marek Belisko - OPEN-NANDRA Freelance Developer Ruska Nova Ves 219 | Presov, 08005 Slovak Republic Tel: +421 915 052 184 skype: marekwhite twitter: #opennandra web: http://open-nandra.com

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[alsa-devel] Advanced Gtk+ Sequencer aka GSequencer now on GitHub
by Joël Krähemann 03 Apr '15

03 Apr '15

Hi all, me just migrated gsequencer to GitHub. http://www.gsequencer.org Further me did renamed binary from `ags` to `gsequencer`. About the state of the sequencer: Version 0.4.2-44 is believed to be stable. Since me don't have the time to test it well, me need people to test it. Please, let me know your experiences. MIDI support is intended to be added for release Version 0.4.3 and many other new exciting features. If someone can help with ALSA rawmidi, please let me know. I don't want to do JACK, since it is believed to be somehow the Qt thing. What me can say so far about MIDI me read years ago about it in "The computer music tutorial - The MIT Press, ISBN 0-262-68082-3". And MIDI consist of messages which are routeable. Someone giving specification how the MIDI message struct is passed to ALSA and how it looks like would be great. Thank you in advance Joël

1 0

[alsa-devel] [PATCH] ASoC: cs4271: Increase delay time after reset
by pascal.huerst＠gmail.com 02 Apr '15

02 Apr '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> The delay time after a reset in the codec probe callback was too short, and did not work on certain hw because the codec needs more time to power on. This increases the delay time from 1us to 1ms. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/cs4271.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 7d3a6ac..e770ee6 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -561,10 +561,10 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ gpio_direction_output(cs4271->gpio_nreset, 0); - udelay(1); + mdelay(1); gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ - udelay(1); + mdelay(1); } ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, -- 1.9.3

3 2

[alsa-devel] [PATCH] ASoC: atmel: ac97c: Remove unused variable
by Fabio Estevam 02 Apr '15

02 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Remove the unused 'match' variable to fix the following build warning: sound/atmel/ac97c.c:919:29: warning: unused variable 'match' [-Wunused-variable] Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/atmel/ac97c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index cf4cedf..6dad042 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -916,7 +916,6 @@ static struct ac97c_platform_data *atmel_ac97c_probe_dt(struct device *dev) { struct ac97c_platform_data *pdata; struct device_node *node = dev->of_node; - const struct of_device_id *match; if (!node) { dev_err(dev, "Device does not have associated DT data\n"); -- 1.9.1

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[alsa-devel] [RFC 0/11] dw_hdmi cleanups, audio preparation, helpers and ahb audio support
by Russell King - ARM Linux 02 Apr '15

02 Apr '15

I'm sending this series for comments, and to allow people to update their code bases (for those who are using my AHB audio driver on iMX6). This is really several series, and I'm not expecting this to be ready for the upcoming merge window. That said, if anyone wants to say that they're happy with some of the initial four patches, I'm happy to queue those for David, if he's willing to take them. This series applies on top of my previous series, which finished with the patch "drm: bridge/dw_hdmi: adjust n/cts setting order". The first four patches are a few cleanups to the dw_hdmi driver. The next two patches introduce new interfaces to the dw_hdmi driver to support AHB audio, including the errata found on iMX6 which requires N to be programmed to zero when the audio FIFO is not full. I'm expecting some discussion with these last two as we try to work out how to deal with the two variants of audio support for this part (AHB audio vs I2S audio). The following three add various helpers to the DRM and ALSA subsystems for audio support which is non-specific to the AHB audio driver. For DRM, this is a helper macro to obtain the pointer into the short audio descriptors. For ALSA, it's a set of helpers to restrict an audio PCM device's capabilities according to the ELD, and to generate the IEC958 channel status data. If these helpers are acceptable, we can convert a number of drivers to them. The ELD helper isn't quite finished, but should be sufficient for initial testing. The last two patches add support for the dw_hdmi audio. This is an ALSA driver rather than an ASoC driver as it's a pretty simple affair. I couldn't find a suitable location in the sound/ subtree to place it, so I've placed it next to the bridge - yes, we probably need to find it a better home in the sound/ subtree, but it currently makes use of dw_hdmi.h in the directory it lives. drivers/gpu/drm/bridge/Kconfig | 11 + drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.c | 566 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw_hdmi-ahb-audio.h | 14 + drivers/gpu/drm/bridge/dw_hdmi.c | 250 ++++++++----- drivers/gpu/drm/bridge/dw_hdmi.h | 3 + include/drm/bridge/dw_hdmi.h | 7 + include/drm/drm_edid.h | 19 + include/sound/pcm_drm_eld.h | 6 + include/sound/pcm_iec958.h | 9 + sound/core/Kconfig | 6 + sound/core/Makefile | 3 + sound/core/pcm_drm_eld.c | 91 +++++ sound/core/pcm_iec958.c | 70 ++++ 14 files changed, 964 insertions(+), 92 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: increase buffer_bytes_max
by Alexandre Belloni 02 Apr '15

02 Apr '15

atmel-pcm-dma is not limited to a buffer size of 64kB like atmel-pcm-pdc. Increase buffer_bytes_max to 512kB to allow for higher bit rates (i.e. 32bps at 192kHz) to work correctly. By default, keep the prealloc at 64kB. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/atmel-pcm-dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index b8e7bad05eb1..b6625c8c411b 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -54,7 +54,7 @@ static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ .periods_min = 8, .periods_max = 1024, /* no limit */ - .buffer_bytes_max = ATMEL_SSC_DMABUF_SIZE, + .buffer_bytes_max = 512 * 1024, }; /** @@ -119,7 +119,7 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = { .prepare_slave_config = atmel_pcm_configure_dma, .pcm_hardware = &atmel_pcm_dma_hardware, - .prealloc_buffer_size = ATMEL_SSC_DMABUF_SIZE, + .prealloc_buffer_size = 64 * 1024, }; int atmel_pcm_dma_platform_register(struct device *dev) -- 2.1.0

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[alsa-devel] ASoC: cs4271: init/timing problem
by Pascal Huerst 02 Apr '15

02 Apr '15

Hey all, We have a custom built hw, based on am335x and from time to time we need to rmmod/modprobe the ASoC machine driver. Depending on the hw, the first call to regmap_update_bits(..) in cs4271_codec_probe(..) fails with -EREMOTEIO. The error is originated in: drivers/i2c/busses/i2c-omap.c and happens if no i2c package acknowledge is received by the host. I think this is a timing issue and on some devices, the codec is just not ready yet, for communication. What is the right way to fix that? My attempt would be something like this: const int retries = 5; while (retries) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); if (ret == -EREMOTEIO) { retries--; udelay(1); continue; } else if (ret) return ret; break; } But I might also just add a proper delay before the first call to regmap_update_bits(..) So what is the proper way to handle this? Thanks Pascal

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Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 02 Apr '15

02 Apr '15

> > >> You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector > > Originally, after the install there was no "model" there. It did not work even then. This entry that you see is because of my trials to make it work. You have to post output of alsa-info.sh without specify any model > > >>You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged > Don't think that is the case. Because it works just fine on the Windows OS. In fact, as I have mentioned, there is some choppiness/stutter sound that I hear occasionally when I play in right now on Fedora/Ubuntu (while most of the time there is no sound). This behaviour goes away when I put the headphone (and then there is no sound at all from the main speaker) Do the front panel headphone still work when you use rear panel jacks for 5.1 sound in windows ? > > >>You have to provide pulseaudio verbose log > Pulseaudio is not a problem because I tested it by killing Pulseaudio too. The same problem persists. Though, if you insist I can surely get you the output. Killing pulseaudio won't work since it can auto-spwan http://lists.freedesktop.org/archives/pulseaudio-discuss/2015-March/023568.… Check the pulseaudio verbose log , as your line out and headphone share the same playback volume control, dB min of line out path and headphone path should be same value http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/m… <http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix…> http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… If Front is shared by HP and Line out, Why set volume to zero dB ? volume should remain unchanged , only need to restore save values after reboot > > >>You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume contro > How do I do that? Appreciate your help here! > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… Codec: Realtek ALC662 rev1 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0662 Subsystem Id: 0x8086434e Revision Id: 0x100101 Early Patching ~~~~~~~~~~~~~~ When CONFIG_SND_HDA_PATCH_LOADER=y is set, you can pass a "patch" as a firmware file for modifying the HD-audio setup before initializing the codec. This can work basically like the reconfiguration via sysfs in the above, but it does it before the first codec configuration. A patch file is a plain text file which looks like below: ------------------------------------------------------------------------ [codec] 0x12345678 0xabcd1234 2 [hint] multi_io = no ------------------------------------------------------------------------ The file needs to have a line `[codec]`. The next line should contain three numbers indicating the codec vendor-id (0x12345678 in the example), the codec subsystem-id (0xabcd1234) and the address (2) of the codec. The rest patch entries are applied to this specified codec until another codec entry is given. Passing 0 or a negative number to the first or the second value will make the check of the corresponding field be skipped. It'll be useful for really broken devices that don't initialize SSID properly. The hd-audio driver reads the file via request_firmware(). Thus, a patch file has to be located on the appropriate firmware path, typically, /lib/firmware. For example, when you pass the option `patch=hda-init.fw`, the file /lib/firmware/hda-init.fw must be present Post output of alsa-info.sh after you reboot with early patching You will get headphone playback volume control but loss those surround , center and lfe playback volume controls http://www.intel.com/support/motherboards/desktop/sb/CS-034206.htm Seem your motherboard does not support multistreaming as it is not in the supported list Do this mean that indep_hp should be explicitly disabled for alc662 ?

1 0

Re: [alsa-devel] [PATCH] fix dig out type to SPDIF when there are two digital pins with different type - VT1708S lost SPDIF device after upgrade
by Raymond Yau 02 Apr '15

02 Apr '15

> > > > Thank you for your reply, > > > > Please see below the output of alsa-info.sh before and after the headphone plugged in > > > > Before: http://www.alsa-project.org/db/?f=a316eebf7b150b5897838be37bce4e8c9d95eaae > > After: http://www.alsa-project.org/db/?f=0f163003e9005972db5f97f21a45be67816e3801 > > > > I have 6 Jacks at the rear panel and two at the top (one output one input) > > I am afraid I do not know how to switch channel mode to retask. > > > > Also, the computer is of a certain age, and the issue has started several month ago but I never really looked into it, so I am not sure that change in power management will be relevant has it seems quite recent. > > What is the model of your asus motherbroad ? > > Seem lost the digial device 1 for SPDIF out > > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > > Is this internal spdif connector ? > > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: Intel [HDA Intel], device 0: VT1708S Analog [VT1708S Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: Intel [HDA Intel], device 3: VT1708S Digital [VT1708S Digital] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > > Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 > Device: name="VT1708S Digital", type="HDMI", device=3 > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x15 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=5, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > > Node 0x20 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="HDMI Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x985601f0: [Fixed] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x12 > Node 0x21 [Pin Complex] wcaps 0x400701: Stereo Digital > Control: name="SPDIF Phantom Jack", index=0, device=0 > Pincap 0x00000010: OUT > Pin Default 0x074511f0: [Jack] SPDIF Out at Ext Rear Panel > Conn = Optical, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > > Power states: D0 D1 D2 D3 > Power: setting=D0, actual=D0 > Connection: 1 > 0x15 Do the driver need a pin fixup for the digital pin or more check when assign dig out type ? If codec has two digital pin complex with diffferent type, check whether the primary digital pin complex support DETECT case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) { codec_info(codec, "ignore pin 0x%x, too many assigned pins\n", nid); continue; } cfg->dig_out_pins[cfg->dig_outs] = nid; cfg->dig_out_type[cfg->dig_outs] = (loc == AC_JACK_LOC_HDMI) ? HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; cfg->dig_outs++; + if (cfg->dig_outs == 2) + if (cfg->dig_out_type[0] != cfg->dig_out_type[1]) + if (! snd_hda_query_pin_caps(codec, cfg->dig_out_pins[0]) & AC_PINCAP_PRES_DETECT)) + cfg-> dig_out_type[0] = HDA_PCM_TYPE_SPDIF; break;

1 0

[alsa-devel] [PATCH v2 0/5] Intel SST driver restructure
by Jie Yang 02 Apr '15

02 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Changes in v2: add "Reviewed-by" and "Acked-by" items from each platforms' maintainer. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

2 7

Re: [alsa-devel] ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used
by Dan Carpenter 01 Apr '15

01 Apr '15

Hello Jyri Sarha, The patch a75a053f1eef: "ASoC: davinci-mcasp: Set rule constraints if implicit BCLK divider is used" from Mar 20, 2015, leads to the following Smatch warning: sound/soc/davinci/davinci-mcasp.c:1152 davinci_mcasp_startup() error: buffer overflow 'mcasp->ruledata' 2 <= 2 sound/soc/davinci/davinci-mcasp.c 1138 1139 /* 1140 * Limit the maximum allowed channels for the first stream: 1141 * number of serializers for the direction * tdm slots per serializer 1142 */ 1143 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 1144 dir = TX_MODE; 1145 else 1146 dir = RX_MODE; dir is either 1 or 2. 1147 1148 for (i = 0; i < mcasp->num_serializer; i++) { 1149 if (mcasp->serial_dir[i] == dir) 1150 max_channels++; 1151 } 1152 mcasp->ruledata[dir].serializers = max_channels; ^^^^^^^^^^^^^ ->ruledata[] has only two elements so ->ruledata[2] is beyond the end of the array. 1153 max_channels *= mcasp->tdm_slots; regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: No not use IEC958_AES1_PRO_MODE_NOTID
by Jyri Sarha 01 Apr '15

01 Apr '15

No IEC958_AES?_PRO_* macros should be used in HDMI consumer audio mode and IEC958_AES1_PRO_MODE_NOTID should be applied to byte 1 when applicable. However IEC958_AES1_PRO_MODE_NOTID is defined as 0 so this fix does not affect the functionality in any way. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- When I refactored the OMAP HDMI audio functionality I intentionally wanted to keep all the HW bits identical to the earlier implementation so that I could compare the register values of the old and the new implementations one to one. One of these days I should go trough and check all the bits now that new implementation has settled in. Best regards, Jyri sound/soc/omap/omap-hdmi-audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f7eb42a..4775da4 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -142,8 +142,6 @@ static int hdmi_dai_hw_params(struct snd_pcm_substream *substream, iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE; - iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID; - iec->status[1] = IEC958_AES1_CON_GENERAL; iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC; -- 1.9.1

2 1

[alsa-devel] [PATCH 1/8] ASoC: wm8971: Use system_power_efficient_wq instead of custom workqueue
by Lars-Peter Clausen 01 Apr '15

01 Apr '15

The delayed work used by the wm8971 driver to manage the caps charging doesn't have any special requirements that would justify using a custom workqueue, just use the generic system_power_efficient_wq instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/wm8971.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 39ddb9b..44baacd 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -31,8 +31,6 @@ #define WM8971_REG_COUNT 43 -static struct workqueue_struct *wm8971_workq = NULL; - /* codec private data */ struct wm8971_priv { unsigned int sysclk; @@ -636,7 +634,8 @@ static int wm8971_resume(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_ON; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); } @@ -649,9 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) u16 reg; INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); - wm8971_workq = create_workqueue("wm8971"); - if (wm8971_workq == NULL) - return -ENOMEM; wm8971_reset(codec); @@ -659,7 +655,8 @@ static int wm8971_probe(struct snd_soc_codec *codec) reg = snd_soc_read(codec, WM8971_PWR1) & 0xfe3e; snd_soc_write(codec, WM8971_PWR1, reg | 0x01c0); codec->dapm.bias_level = SND_SOC_BIAS_STANDBY; - queue_delayed_work(wm8971_workq, &codec->dapm.delayed_work, + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, msecs_to_jiffies(1000)); /* set the update bits */ @@ -681,8 +678,6 @@ static int wm8971_remove(struct snd_soc_codec *codec) { wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8971_workq) - destroy_workqueue(wm8971_workq); return 0; } -- 1.8.0

3 9

[alsa-devel] [PATCH] ASoC: atmel-pcm-pdc: merge atmel-pcm back in
by Alexandre Belloni 01 Apr '15

01 Apr '15

atmel-pcm.c was split into two files to create a generic framework for both PDC and DMA. atmel-pcm-dma.c is using the generic dmaengine framework since 95e0e07e710e (ASoC: atmel-pcm: use generic dmaengine framework). Merge atmel-pcm.c in atmel-pcm-pdc.c as this is now the only user. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Makefile | 2 - sound/soc/atmel/atmel-pcm-pdc.c | 79 ++++++++++++++++++++++++++ sound/soc/atmel/atmel-pcm.c | 121 ---------------------------------------- sound/soc/atmel/atmel-pcm.h | 5 -- 4 files changed, 79 insertions(+), 128 deletions(-) delete mode 100644 sound/soc/atmel/atmel-pcm.c diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 466a821da98c..b327e5cc8de3 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -1,10 +1,8 @@ # AT91 Platform Support -snd-soc-atmel-pcm-objs := atmel-pcm.o snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o -obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o diff --git a/sound/soc/atmel/atmel-pcm-pdc.c b/sound/soc/atmel/atmel-pcm-pdc.c index a366b3503c28..da861b44413f 100644 --- a/sound/soc/atmel/atmel-pcm-pdc.c +++ b/sound/soc/atmel/atmel-pcm-pdc.c @@ -47,6 +47,85 @@ #include "atmel-pcm.h" +static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, + int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = ATMEL_SSC_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_coherent(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", + (void *)buf->area, (void *)(long)buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static int atmel_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + return remap_pfn_range(vma, vma->vm_start, + substream->dma_buffer.addr >> PAGE_SHIFT, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); + ret = atmel_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + out: + return ret; +} + +static void atmel_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + dma_free_coherent(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + /*--------------------------------------------------------------------------*\ * Hardware definition \*--------------------------------------------------------------------------*/ diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c deleted file mode 100644 index 8ae3fa5ac60a..000000000000 --- a/sound/soc/atmel/atmel-pcm.c +++ /dev/null @@ -1,121 +0,0 @@ -/* - * atmel-pcm.c -- ALSA PCM interface for the Atmel atmel SoC. - * - * Copyright (C) 2005 SAN People - * Copyright (C) 2008 Atmel - * - * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> - * - * Based on at91-pcm. by: - * Frank Mandarino <fmandarino(a)endrelia.com> - * Copyright 2006 Endrelia Technologies Inc. - * - * Based on pxa2xx-pcm.c by: - * - * Author: Nicolas Pitre - * Created: Nov 30, 2004 - * Copyright: (C) 2004 MontaVista Software, Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include <linux/module.h> -#include <linux/dma-mapping.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include "atmel-pcm.h" - -static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, - int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = ATMEL_SSC_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)(long)buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} -EXPORT_SYMBOL_GPL(atmel_pcm_mmap); - -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - pr_debug("atmel-pcm: allocating PCM playback DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - pr_debug("atmel-pcm: allocating PCM capture DMA buffer\n"); - ret = atmel_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} -EXPORT_SYMBOL_GPL(atmel_pcm_new); - -void atmel_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(atmel_pcm_free); - diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814eff21..6eaf081cad50 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -83,11 +83,6 @@ struct atmel_pcm_dma_params { #define ssc_readx(base, reg) (__raw_readl((base) + (reg))) #define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) -int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); -void atmel_pcm_free(struct snd_pcm *pcm); -int atmel_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma); - #if defined(CONFIG_SND_ATMEL_SOC_PDC) || \ defined(CONFIG_SND_ATMEL_SOC_PDC_MODULE) int atmel_pcm_pdc_platform_register(struct device *dev); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: rsnd: add Synchronous SRC mode
by Kuninori Morimoto 01 Apr '15

01 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas R-Car sound SRC (= Sampling Rate Converter) has Asynchronous/Synchronous SRC mode. Asynchronous mode is already supported via DPCM. This patch adds Synchronous mode on it. The condition of enabling Synchronous mode are - SoC is clock master - Sound uses SRC - Sound doesn't use DVC - Sound card uses DPCM (= rsrc-card card) amixer set "SRC Out Rate" on aplay xxx.wav & amixer set "SRC Out Rate" 48000 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/src.c | 126 ++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 121 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index a0a2bda..3beb32e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -22,13 +22,15 @@ struct rsnd_src { struct rsnd_src_platform_info *info; /* rcar_snd.h */ struct rsnd_mod mod; + struct rsnd_kctrl_cfg_s sen; /* sync convert enable */ + struct rsnd_kctrl_cfg_s sync; /* sync convert */ u32 convert_rate; /* sampling rate convert */ int err; }; #define RSND_SRC_NAME_SIZE 16 -#define rsnd_src_convert_rate(s) ((s)->convert_rate) +#define rsnd_enable_sync_convert(src) ((src)->sen.val) #define rsnd_src_of_node(priv) \ of_get_child_by_name(rsnd_priv_to_dev(priv)->of_node, "rcar_sound,src") @@ -233,6 +235,30 @@ int rsnd_src_ssi_irq_disable(struct rsnd_mod *ssi_mod) return 0; } +static u32 rsnd_src_convert_rate(struct rsnd_src *src) +{ + struct rsnd_mod *mod = &src->mod; + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + u32 convert_rate; + + if (!runtime) + return 0; + + if (!rsnd_enable_sync_convert(src)) + return src->convert_rate; + + convert_rate = src->sync.val; + + if (!convert_rate) + convert_rate = src->convert_rate; + + if (!convert_rate) + convert_rate = runtime->rate; + + return convert_rate; +} + unsigned int rsnd_src_get_ssi_rate(struct rsnd_priv *priv, struct rsnd_dai_stream *io, struct snd_pcm_runtime *runtime) @@ -333,6 +359,9 @@ static int rsnd_src_init(struct rsnd_mod *mod, src->err = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + /* * Initialize the operation of the SRC internal circuits * see rsnd_src_start() @@ -356,6 +385,9 @@ static int rsnd_src_quit(struct rsnd_mod *mod, src->convert_rate = 0; + /* reset sync convert_rate */ + src->sync.val = 0; + return 0; } @@ -672,6 +704,7 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); struct rsnd_src *src = rsnd_mod_to_src(mod); u32 convert_rate = rsnd_src_convert_rate(src); + u32 cr, route; uint ratio; int ret; @@ -692,13 +725,21 @@ static int rsnd_src_set_convert_rate_gen2(struct rsnd_mod *mod) if (ret < 0) return ret; - rsnd_mod_write(mod, SRC_SRCCR, 0x00011110); - + cr = 0x00011110; + route = 0x0; if (convert_rate) { - /* Gen1/Gen2 are not compatible */ - rsnd_mod_write(mod, SRC_ROUTE_MODE0, 1); + route = 0x1; + + if (rsnd_enable_sync_convert(src)) { + cr |= 0x1; + route |= rsnd_io_is_play(io) ? + (0x1 << 24) : (0x1 << 25); + } } + rsnd_mod_write(mod, SRC_SRCCR, cr); + rsnd_mod_write(mod, SRC_ROUTE_MODE0, route); + switch (rsnd_mod_id(mod)) { case 5: case 6: @@ -811,6 +852,80 @@ static int rsnd_src_stop_gen2(struct rsnd_mod *mod, return ret; } +static void rsnd_src_reconvert_update(struct rsnd_mod *mod) +{ + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + u32 convert_rate = rsnd_src_convert_rate(src); + u32 fsrate; + + if (!runtime) + return; + + if (!convert_rate) + convert_rate = runtime->rate; + + fsrate = 0x0400000 / convert_rate * runtime->rate; + + /* update IFS */ + rsnd_mod_write(mod, SRC_IFSVR, fsrate); +} + +static int rsnd_src_pcm_new(struct rsnd_mod *mod, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct rsnd_dai_stream *io = rsnd_mod_to_io(mod); + struct rsnd_dai *rdai = rsnd_io_to_rdai(io); + struct rsnd_src *src = rsnd_mod_to_src(mod); + int ret; + + /* + * enable SRC sync convert if possible + */ + + /* + * Gen1 is not supported + */ + if (rsnd_is_gen1(priv)) + return 0; + + /* + * SRC sync convert needs clock master + */ + if (!rsnd_rdai_is_clk_master(rdai)) + return 0; + + /* + * We can't use SRC sync convert + * if it has DVC + */ + if (rsnd_io_to_mod_dvc(io)) + return 0; + + /* + * enable sync convert + */ + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate Switch" : + "SRC In Rate Switch", + rsnd_src_reconvert_update, + &src->sen, 1); + if (ret < 0) + return ret; + + ret = rsnd_kctrl_new_s(mod, rtd, + rsnd_io_is_play(io) ? + "SRC Out Rate" : + "SRC In Rate", + rsnd_src_reconvert_update, + &src->sync, 192000); + + return ret; +} + static struct rsnd_mod_ops rsnd_src_gen2_ops = { .name = SRC_NAME, .dma_req = rsnd_src_dma_req, @@ -821,6 +936,7 @@ static struct rsnd_mod_ops rsnd_src_gen2_ops = { .start = rsnd_src_start_gen2, .stop = rsnd_src_stop_gen2, .hw_params = rsnd_src_hw_params, + .pcm_new = rsnd_src_pcm_new, }; struct rsnd_mod *rsnd_src_mod_get(struct rsnd_priv *priv, int id) -- 1.9.1

2 1

[alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 01 Apr '15

01 Apr '15

Joe Perches (25): arm: Use bool function return values of true/false not 1/0 arm64: Use bool function return values of true/false not 1/0 hexagon: Use bool function return values of true/false not 1/0 ia64: Use bool function return values of true/false not 1/0 mips: Use bool function return values of true/false not 1/0 powerpc: Use bool function return values of true/false not 1/0 s390: Use bool function return values of true/false not 1/0 sparc: Use bool function return values of true/false not 1/0 tile: Use bool function return values of true/false not 1/0 unicore32: Use bool function return values of true/false not 1/0 x86: Use bool function return values of true/false not 1/0 virtio_console: Use bool function return values of true/false not 1/0 csiostor: Use bool function return values of true/false not 1/0 dcache: Use bool function return values of true/false not 1/0 nfsd: nfs4state: Use bool function return values of true/false not 1/0 include/linux: Use bool function return values of true/false not 1/0 sound: Use bool function return values of true/false not 1/0 rcu: tree_plugin: Use bool function return values of true/false not 1/0 sched: Use bool function return values of true/false not 1/0 ftrace: Use bool function return values of true/false not 1/0 slub: Use bool function return values of true/false not 1/0 bridge: Use bool function return values of true/false not 1/0 netfilter: Use bool function return values of true/false not 1/0 security: Use bool function return values of true/false not 1/0 sound: wm5100-tables: Use bool function return values of true/false not 1/0 arch/arm/include/asm/dma-mapping.h | 8 ++-- arch/arm/include/asm/kvm_emulate.h | 2 +- arch/arm/mach-omap2/powerdomain.c | 14 +++--- arch/arm64/include/asm/dma-mapping.h | 2 +- arch/hexagon/include/asm/dma-mapping.h | 2 +- arch/ia64/include/asm/dma-mapping.h | 2 +- arch/mips/include/asm/dma-mapping.h | 2 +- arch/powerpc/include/asm/dcr-native.h | 2 +- arch/powerpc/include/asm/dma-mapping.h | 4 +- arch/powerpc/include/asm/kvm_book3s_64.h | 4 +- arch/powerpc/sysdev/dcr.c | 2 +- arch/s390/include/asm/dma-mapping.h | 2 +- arch/sparc/mm/init_64.c | 8 ++-- arch/tile/include/asm/dma-mapping.h | 2 +- arch/unicore32/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/archrandom.h | 2 +- arch/x86/include/asm/dma-mapping.h | 2 +- arch/x86/include/asm/kvm_para.h | 2 +- arch/x86/kvm/cpuid.h | 2 +- arch/x86/kvm/vmx.c | 72 ++++++++++++++-------------- drivers/char/virtio_console.c | 2 +- drivers/scsi/csiostor/csio_scsi.c | 4 +- fs/dcache.c | 12 ++--- fs/nfsd/nfs4state.c | 2 +- include/linux/blkdev.h | 2 +- include/linux/ide.h | 2 +- include/linux/kgdb.h | 2 +- include/linux/mfd/db8500-prcmu.h | 2 +- include/linux/mm.h | 2 +- include/linux/power_supply.h | 8 ++-- include/linux/ssb/ssb_driver_extif.h | 2 +- include/linux/ssb/ssb_driver_gige.h | 16 +++---- include/sound/soc.h | 4 +- kernel/rcu/tree_plugin.h | 4 +- kernel/sched/auto_group.h | 2 +- kernel/sched/completion.c | 16 ++++--- kernel/trace/ftrace.c | 10 ++-- mm/slub.c | 12 ++--- net/bridge/br_private.h | 2 +- net/ipv4/netfilter/ipt_ah.c | 2 +- net/netfilter/ipset/ip_set_hash_ip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipmark.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportip.c | 8 ++-- net/netfilter/ipset/ip_set_hash_ipportnet.c | 8 ++-- net/netfilter/ipset/ip_set_hash_net.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netiface.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netport.c | 8 ++-- net/netfilter/ipset/ip_set_hash_netportnet.c | 8 ++-- net/netfilter/xt_connlimit.c | 2 +- net/netfilter/xt_hashlimit.c | 2 +- net/netfilter/xt_ipcomp.c | 2 +- security/apparmor/file.c | 8 ++-- security/apparmor/policy.c | 10 ++-- sound/soc/codecs/wm5100-tables.c | 12 ++--- 55 files changed, 178 insertions(+), 176 deletions(-) -- 2.1.2

3 4

[alsa-devel] [PATCH v3 0/5] ALSA: hda - add hda controller to hda core
by Vinod Koul 01 Apr '15

01 Apr '15

This incorporates the comments from Takashi on 1st patch and uses the change from Takashi's unstable tree for controller and then adds the DSP loader code on top. This is based on sound/topic/hda-core Comments welcome Jeeja KP (1): ALSA: hda: move common hda controller register defines up Vinod Koul (4): ALSA: hda: add HDA_MAX_CODECS ALSA: hda - Add the controller helper codes to hda-core module ALSA: hda - move dsp lock to hdac ALSA: hda - add the DSP loader code include/sound/hda_registers.h | 177 +++++++++++++ include/sound/hdaudio.h | 185 ++++++++++++- sound/hda/Makefile | 3 +- sound/hda/hdac_bus.c | 13 +- sound/hda/hdac_controller.c | 401 ++++++++++++++++++++++++++++ sound/hda/hdac_stream.c | 565 ++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.c | 17 +- sound/pci/hda/hda_controller.h | 171 +----------- 8 files changed, 1347 insertions(+), 185 deletions(-) create mode 100644 include/sound/hda_registers.h create mode 100644 sound/hda/hdac_controller.c create mode 100644 sound/hda/hdac_stream.c -- 1.7.9.5

1 5

[alsa-devel] [PATCH] ASoC: rcar: mark device data as constant
by Uwe Kleine-König 01 Apr '15

01 Apr '15

A driver's device data should and can be const. This is a follow-up on commit 33187fb4a203 (ASoC: rsnd: constify of_device_id array) which marked the of_device_id as const. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- This is also patch 4 of my series[1] which degraded to this hunk when rebasing to next because all the other changes were already addressed by Fabian. Best regards Uwe [1] http://thread.gmane.org/gmane.linux.alsa.devel/136241/ sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 3d8e4d60c83d..9f48d75fa992 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -99,11 +99,11 @@ #define RSND_RATES SNDRV_PCM_RATE_8000_96000 #define RSND_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -static struct rsnd_of_data rsnd_of_data_gen1 = { +static const struct rsnd_of_data rsnd_of_data_gen1 = { .flags = RSND_GEN1, }; -static struct rsnd_of_data rsnd_of_data_gen2 = { +static const struct rsnd_of_data rsnd_of_data_gen2 = { .flags = RSND_GEN2, }; -- 2.1.4 -- Pengutronix e.K. | Uwe Kleine-König | Industrial Linux Solutions | http://www.pengutronix.de/ |

2 1

[alsa-devel] [PATCH v2 0/4] ASoC: constification and random cleanups
by Uwe Kleine-König 01 Apr '15

01 Apr '15

Hello, this is the series sent on Sunday rebased on next. There was a conflict with commit c660c0a80586 (ASoC: fsi: constify of_device_id array) which did a part of my 2nd patch but already conflicted with the first one. Mark said he already took patch 3, but that should not be a problem. Patch 4 applied just fine to next, so this is just a resend. Best regards Uwe Uwe Kleine-König (4): ASoC: fsi: reorder code to make a forward declaration superfluous ASoC: fsi: mark several data structures as const ASoC: fsi: fix license specification ASoC: mark of_device_id and device data as constant sound/soc/sh/fsi.c | 71 ++++++++++++++++++++++++------------------------ sound/soc/sh/rcar/core.c | 4 +-- 2 files changed, 37 insertions(+), 38 deletions(-) -- 2.1.4

2 10

[alsa-devel] [PATCH 0/5] Intel SST driver restructure
by Jie Yang 01 Apr '15

01 Apr '15

In order to prepare for the upstreaming of Skylake drivers and to allow reuse of code between different Intel audio DSP products, this patch series restructures the Intel directory as follows: 1. Create "common" directory. Move common files into this directory. 2. Create "boards" directory. Move all machine drivers into this directory. 3. Create "haswell" directory. Move all remaining hsw/bdw code into this directory. 4. Create "baytrail" directory. Move remaining baytrail driver to this directory. 5. Create "atom" directory. Move all MCG platform drivers(including sst/ folder) to this directory. Todo next step: 1. Change and move atom related codes(in atom/sst/) to corresponding and try share as many as possible codes(e.g. sst_acpi.c). 2. Share more common codes about IPC for different platforms, e.g. The methods for queueing and sending/reciving Tx/Rx messages, The methods for mailbox IO, The IPC error return codes, etc. Jie Yang (5): ASoC: Intel: create common folder and move common files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create atom folder and move atom platform files in sound/soc/intel/Makefile | 42 +++------------------- sound/soc/intel/atom/Makefile | 7 ++++ sound/soc/intel/{ => atom}/sst-atom-controls.c | 0 sound/soc/intel/{ => atom}/sst-atom-controls.h | 0 sound/soc/intel/{ => atom}/sst-mfld-dsp.h | 0 .../intel/{ => atom}/sst-mfld-platform-compress.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c | 0 sound/soc/intel/{ => atom}/sst-mfld-platform.h | 0 sound/soc/intel/{ => atom}/sst/Makefile | 0 sound/soc/intel/{ => atom}/sst/sst.c | 0 sound/soc/intel/{ => atom}/sst/sst.h | 0 sound/soc/intel/{ => atom}/sst/sst_acpi.c | 0 sound/soc/intel/{ => atom}/sst/sst_drv_interface.c | 0 sound/soc/intel/{ => atom}/sst/sst_ipc.c | 0 sound/soc/intel/{ => atom}/sst/sst_loader.c | 0 sound/soc/intel/{ => atom}/sst/sst_pci.c | 0 sound/soc/intel/{ => atom}/sst/sst_pvt.c | 0 sound/soc/intel/{ => atom}/sst/sst_stream.c | 0 sound/soc/intel/baytrail/Makefile | 4 +++ sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c | 4 +-- sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h | 0 sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c | 4 +-- sound/soc/intel/boards/Makefile | 15 ++++++++ sound/soc/intel/{ => boards}/broadwell.c | 6 ++-- sound/soc/intel/{ => boards}/byt-max98090.c | 2 +- sound/soc/intel/{ => boards}/byt-rt5640.c | 4 +-- .../{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5645.c | 0 sound/soc/intel/{ => boards}/cht_bsw_rt5672.c | 0 sound/soc/intel/{ => boards}/haswell.c | 6 ++-- sound/soc/intel/{ => boards}/mfld_machine.c | 0 sound/soc/intel/common/Makefile | 6 ++++ sound/soc/intel/{ => common}/sst-acpi.c | 0 sound/soc/intel/{ => common}/sst-dsp-priv.h | 0 sound/soc/intel/{ => common}/sst-dsp.c | 0 sound/soc/intel/{ => common}/sst-dsp.h | 0 sound/soc/intel/{ => common}/sst-firmware.c | 0 sound/soc/intel/haswell/Makefile | 4 +++ sound/soc/intel/{ => haswell}/sst-haswell-dsp.c | 6 ++-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.c | 4 +-- sound/soc/intel/{ => haswell}/sst-haswell-ipc.h | 0 sound/soc/intel/{ => haswell}/sst-haswell-pcm.c | 6 ++-- 43 files changed, 64 insertions(+), 60 deletions(-) create mode 100644 sound/soc/intel/atom/Makefile rename sound/soc/intel/{ => atom}/sst-atom-controls.c (100%) rename sound/soc/intel/{ => atom}/sst-atom-controls.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-dsp.h (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-compress.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform-pcm.c (100%) rename sound/soc/intel/{ => atom}/sst-mfld-platform.h (100%) rename sound/soc/intel/{ => atom}/sst/Makefile (100%) rename sound/soc/intel/{ => atom}/sst/sst.c (100%) rename sound/soc/intel/{ => atom}/sst/sst.h (100%) rename sound/soc/intel/{ => atom}/sst/sst_acpi.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_drv_interface.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_ipc.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_loader.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pci.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_pvt.c (100%) rename sound/soc/intel/{ => atom}/sst/sst_stream.c (100%) create mode 100644 sound/soc/intel/baytrail/Makefile rename sound/soc/intel/{ => baytrail}/sst-baytrail-dsp.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.c (99%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-ipc.h (100%) rename sound/soc/intel/{ => baytrail}/sst-baytrail-pcm.c (99%) create mode 100644 sound/soc/intel/boards/Makefile rename sound/soc/intel/{ => boards}/broadwell.c (98%) rename sound/soc/intel/{ => boards}/byt-max98090.c (99%) rename sound/soc/intel/{ => boards}/byt-rt5640.c (98%) rename sound/soc/intel/{bytcr_dpcm_rt5640.c => boards/bytcr_rt5640.c} (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5645.c (100%) rename sound/soc/intel/{ => boards}/cht_bsw_rt5672.c (100%) rename sound/soc/intel/{ => boards}/haswell.c (98%) rename sound/soc/intel/{ => boards}/mfld_machine.c (100%) create mode 100644 sound/soc/intel/common/Makefile rename sound/soc/intel/{ => common}/sst-acpi.c (100%) rename sound/soc/intel/{ => common}/sst-dsp-priv.h (100%) rename sound/soc/intel/{ => common}/sst-dsp.c (100%) rename sound/soc/intel/{ => common}/sst-dsp.h (100%) rename sound/soc/intel/{ => common}/sst-firmware.c (100%) create mode 100644 sound/soc/intel/haswell/Makefile rename sound/soc/intel/{ => haswell}/sst-haswell-dsp.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.c (99%) rename sound/soc/intel/{ => haswell}/sst-haswell-ipc.h (100%) rename sound/soc/intel/{ => haswell}/sst-haswell-pcm.c (99%) -- 1.9.1

6 16

[alsa-devel] Rane SL2/3/4 driver?
by Be 01 Apr '15

01 Apr '15

Hi, I recently published the first version of Crossfade GNU/Linux, a distro for DJing from USB drives on any PC (see http://mixxx.org/forums/viewtopic.php?f=1&t=7031 for details). I just tried it on my friend's Macbook which he uses with a Rane SL4 USB sound card. I realized there is no Linux driver for the current line of sound cards sold for digital vinyl emulation with Serato DJ. These are the Rane SL2, SL3, and SL4 (http://serato.com/hardware/all/mixers-interfaces ; the Denon DS1 was also recently announced but not available yet: http://serato.com/latest/blog/17973/announcing-denon-ds1 ). They range from 4 in/4 out to 10 in/10 out and all have phono preamps built in. There appears to be support for the old and discontinued Rane SL1, which used USB 1.1, but the SL2/3/4 use USB 2.0. Has anyone thought about writing a driver for these?

1 1

[alsa-devel] [RESEND PATCH] MAINTAINERS: change the Atmel audio alsa driver entry
by Nicolas Ferre 31 Mar '15

31 Mar '15

I take over the the maintainship of Atmel alsa drivers from Voice. Thanks for your work! Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Acked-by: Bo Shen <voice.shen(a)atmel.com> Cc: Mark Brown <broonie(a)kernel.org> --- Hi Mark, Here is the MAINTAINERS' change that I sent to the wrong address. Can you please take it with your tree? Thanks, bye. MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6cb40f8736ee..a915bb124fe7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1762,7 +1762,7 @@ S: Supported F: drivers/tty/serial/atmel_serial.c ATMEL Audio ALSA driver -M: Bo Shen <voice.shen(a)atmel.com> +M: Nicolas Ferre <nicolas.ferre(a)atmel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/atmel -- 2.1.3

2 1

[alsa-devel] [PATCH 1/4] ASoC: fsi: reorder code to make a forward declaration superfluous
by Uwe Kleine-König 31 Mar '15

31 Mar '15

Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/sh/fsi.c | 69 +++++++++++++++++++++++++++--------------------------- 1 file changed, 34 insertions(+), 35 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b87b22e88e43..dab6aa661a22 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1876,7 +1876,40 @@ static void fsi_handler_init(struct fsi_priv *fsi, } } -static struct of_device_id fsi_of_match[]; +static struct fsi_core fsi1_core = { + .ver = 1, + + /* Interrupt */ + .int_st = INT_ST, + .iemsk = IEMSK, + .imsk = IMSK, +}; + +static struct fsi_core fsi2_core = { + .ver = 2, + + /* Interrupt */ + .int_st = CPU_INT_ST, + .iemsk = CPU_IEMSK, + .imsk = CPU_IMSK, + .a_mclk = A_MST_CTLR, + .b_mclk = B_MST_CTLR, +}; + +static struct of_device_id fsi_of_match[] = { + { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, + { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, + {}, +}; +MODULE_DEVICE_TABLE(of, fsi_of_match); + +static struct platform_device_id fsi_id_table[] = { + { "sh_fsi", (kernel_ulong_t)&fsi1_core }, + { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, + {}, +}; +MODULE_DEVICE_TABLE(platform, fsi_id_table); + static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; @@ -2072,40 +2105,6 @@ static struct dev_pm_ops fsi_pm_ops = { .resume = fsi_resume, }; -static struct fsi_core fsi1_core = { - .ver = 1, - - /* Interrupt */ - .int_st = INT_ST, - .iemsk = IEMSK, - .imsk = IMSK, -}; - -static struct fsi_core fsi2_core = { - .ver = 2, - - /* Interrupt */ - .int_st = CPU_INT_ST, - .iemsk = CPU_IEMSK, - .imsk = CPU_IMSK, - .a_mclk = A_MST_CTLR, - .b_mclk = B_MST_CTLR, -}; - -static struct of_device_id fsi_of_match[] = { - { .compatible = "renesas,sh_fsi", .data = &fsi1_core}, - { .compatible = "renesas,sh_fsi2", .data = &fsi2_core}, - {}, -}; -MODULE_DEVICE_TABLE(of, fsi_of_match); - -static struct platform_device_id fsi_id_table[] = { - { "sh_fsi", (kernel_ulong_t)&fsi1_core }, - { "sh_fsi2", (kernel_ulong_t)&fsi2_core }, - {}, -}; -MODULE_DEVICE_TABLE(platform, fsi_id_table); - static struct platform_driver fsi_driver = { .driver = { .name = "fsi-pcm-audio", -- 2.1.4

2 6

[alsa-devel] [PATCH] ASoC: pcm512x: Do not redefine DIV_ROUND_CLOSEST_ULL
by Fabio Estevam 31 Mar '15

31 Mar '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> DIV_ROUND_CLOSEST_ULL is already defined at <linux/kernel.h>, so use the definition from this header file rather than re-defining it locally. This fixes the following build warning: sound/soc/codecs/pcm512x.c:34:0: warning: "DIV_ROUND_CLOSEST_ULL" redefined [enabled by default] In file included from include/linux/list.h:8:0, from include/linux/module.h:9, from sound/soc/codecs/pcm512x.c:19: include/linux/kernel.h:110:0: note: this is the location of the previous definition Reported-by: Olof's autobuilder <build(a)lixom.net> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/pcm512x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5a30fdd..e12764d 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,6 +18,7 @@ #include <linux/init.h> #include <linux/module.h> #include <linux/clk.h> +#include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -31,8 +32,6 @@ #define DIV_ROUND_DOWN_ULL(ll, d) \ ({ unsigned long long _tmp = (ll); do_div(_tmp, d); _tmp; }) -#define DIV_ROUND_CLOSEST_ULL(ll, d) \ - ({ unsigned long long _tmp = (ll)+(d)/2; do_div(_tmp, d); _tmp; }) #define PCM512x_NUM_SUPPLIES 3 static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { -- 1.9.1

2 1

Re: [alsa-devel] [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
by Mark Brown 31 Mar '15

31 Mar '15

On Mon, Mar 30, 2015 at 08:16:24PM -0700, Scott Branden wrote: The CC list for this patch is pretty wide - please look at who you're sending this to and try to send to only relevant people (for example I'm not sure the Raspberry Pi people need to review this). People working upstream get a lot of mail so it's helpful to avoid filling their inboxes with random irrelevant stuff. > sound/soc/bcm/Kconfig | 11 + > sound/soc/bcm/Makefile | 5 +- > sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++++++++++++++++ > sound/soc/bcm/cygnus-pcm.h | 45 ++ > sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++++++++++++++++++++++++++ > sound/soc/bcm/cygnus-ssp.h | 84 +++ > 6 files changed, 2675 insertions(+), 1 deletion(-) Send the DMA and DAI drivers as separate patches please, it makes review easier. > +config SND_SOC_CYGNUS > + tristate "SoC platform audio for Broadcom Cygnus chips" > + depends on ARCH_BCM_CYGNUS || COMPILE_TEST > + default ARCH_BCM_CYGNUS Remove the default setting here - we don't do this for other drivers, we shouldn't do it here. > +/* > + * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick. > + * This number should be a multiple of 256 > + */ > +#define PERIOD_BYTES_MIN 0x100 This sounds like it's a setting rather than actually the minimum? > +static const struct snd_pcm_hardware cygnus_pcm_hw = { > + .info = SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_INTERLEAVED, The DMA controller is integrated into the IP? > +static int enable_count; This looks bogus - why is this a global variable not part of the device struct and if it does need to be global why does it need no locking? > + if (aio->portnum == 0) > + *p_rbuf = RINGBUF_REG_PLAYBACK(0); > + else if (aio->portnum == 1) > + *p_rbuf = RINGBUF_REG_PLAYBACK(2); > + else if (aio->portnum == 2) > + *p_rbuf = RINGBUF_REG_PLAYBACK(4); > + else if (aio->portnum == 3) > + *p_rbuf = RINGBUF_REG_PLAYBACK(6); /*SPDIF */ > + else > + status = -EINVAL; This looks like a switch statement, there are many places in the code where you're writing switch statements as chains of ifs. > +static void ringbuf_inc(void __iomem *audio_io, bool is_playback, > + const struct ringbuf_regs *p_rbuf) > +{ > + u32 regval, endval, active_ptr; > + > + if (is_playback) > + active_ptr = p_rbuf->wraddr; > + else > + active_ptr = p_rbuf->rdaddr; > + > + endval = readl(audio_io + p_rbuf->endaddr); > + regval = readl(audio_io + active_ptr); > + regval = regval + p_rbuf->period_bytes; > + if (regval > endval) > + regval -= p_rbuf->buf_size; > + > + writel(regval, audio_io + active_ptr); > +} So it looks like we're getting an interrupt per period and we have to manually advance to the next one? > +static irqreturn_t cygnus_dma_irq(int irq, void *data) > +{ > + u32 r5_status; > + struct cygnus_audio *cygaud; > + > + cygaud = (struct cygnus_audio *)data; If you need to cast away from void something is very wrong. > + /* > + * R5 status bits Description > + * 0 ESR0 (playback FIFO interrupt) > + * 1 ESR1 (playback rbuf interrupt) > + * 2 ESR2 (capture rbuf interrupt) > + * 3 ESR3 (Freemark play. interrupt) > + * 4 ESR4 (Fullmark capt. interrupt) > + */ > + r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET); > + > + /* If playback interrupt happened */ > + if (ANY_PLAYBACK_IRQ & r5_status) > + handle_playback_irq(cygaud); > + > + /* If capture interrupt happened */ > + if (ANY_CAPTURE_IRQ & r5_status) > + handle_capture_irq(cygaud); > + > + /* > + * clear r5 interrupts after servicing them to avoid overwriting > + * esr_status > + */ > + writel(r5_status, cygaud->audio + INTH_R5F_CLEAR_OFFSET); This feels racy especially given that we seem to need every interrupt delivering. What if another period happens during the servicing? I don't understand what "overwriting esr_status" means here. > + return IRQ_HANDLED; If neither playback nor capture interrupts were flagged we should return IRQ_NONE. > +/* > + * This code is identical to what is done by the framework, when we do not > + * supply a 'copy' function. Having our own copy hook in place allows for > + * us to easily add some diagnotics when needed. > + */ > +int cygnus_pcm_copy(struct snd_pcm_substream *substream, int channel, > + snd_pcm_uframes_t pos, > + void __user *buf, snd_pcm_uframes_t count) This is obviously not suitable for mainline - "let's just cut'n'paste the shared code in case we want to add diagnostics in future" does not scale, you can always add local diagnostics in the core code. > +}; > +/* Blank line between these two please. Missing blanks between bits of code seem to be a common issue in this driver. > +static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable) > +{ > + u32 value; > + > + if (enable) { > + } else { Why not just write two functions? > +static int audio_ssp_source_bitres(struct cygnus_aio_port *aio, unsigned bits) > +{ > + u32 mask = 0x1f; > + u32 value = 0; > + > + if ((bits == 8) || (bits == 16) || (bits == 32)) { > + value = readl(aio->audio + aio->regs.bf_sourcech_cfg); > + value &= ~(mask << BF_SRC_CFGX_BIT_RES); > + > + /* 32 bit mode is coded as 0 */ > + if ((bits == 8) || (bits == 16)) > + value |= (bits << BF_SRC_CFGX_BIT_RES); > + > + writel(value, aio->audio + aio->regs.bf_sourcech_cfg); > + return 0; > + } > + > + pr_err("Bit resolution not supported %d\n", bits); > + return -EINVAL; dev_err() (this applies throughout the driver). It's not clear why this is a function and not just written as a single statement in the one place that it's used (there are multiple calls but they're all together and could just be inlined). > + if (!aio->bitrate) { > + pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__); > + return -EINVAL; > + } This seems bogus - why are we enforcing the use of set_clkdiv()? Can't we figure out something esneible? > + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > + /* Set the SSP up as slave */ > + case SND_SOC_DAIFMT_CBM_CFM: The comments here look odd due to the indentation and really aren't adding much anyway. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_RIGHT_J: > + case SND_SOC_DAIFMT_LEFT_J: > + return -EINVAL; These are just eaten by the default case. > + case SND_SOC_DAIFMT_DSP_A: > + case SND_SOC_DAIFMT_DSP_B: > + ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE); > + > + /* DSP_A = data after FS, DSP_B = data during FS */ > + if (SND_SOC_DAIFMT_DSP_A) > + ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT); That if statement isn't doing what was intended... > + } else { > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + audio_ssp_in_enable(aio, 1); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + audio_ssp_in_enable(aio, 0); > + break; > + } We can just ignore other triggers? It's not clear why this is different to the playback case. > +int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai) > +{ > + struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai); > + > + return aio->mode; > +} > +EXPORT_SYMBOL(cygnus_ssp_get_mode); What is this for, it's setting off alarm bells? Note also that ASoC is _GPL() only. > +static const struct snd_kcontrol_new pll_tweak_controls[] = { > + SOC_SINGLE_EXT("PLL Tweak", 0, 0, PLL_NDIV_FRACT_MAX, 0, > + pll_tweak_get, pll_tweak_put), > +}; Whatever this control is doing it should be a separate patch (as I think we discussed in person a ELC?), it's clearly something that's unusual and is likely to block the rest of the code as a result. At a high level this needs at least some documentation. > +int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + > + return snd_soc_add_dai_controls(cpu_dai, > + pll_tweak_controls, > + ARRAY_SIZE(pll_tweak_controls)); > +} > +EXPORT_SYMBOL(cygnus_ssp_add_pll_tweak_controls); Again, why is this being exported and why is it not _GPL? If the driver is adding controls I'd expect it to just add its controls itself. > +static int cygnus_ssp_probe(struct platform_device *pdev) > +{ > + struct device *dev = &pdev->dev; > + struct device_node *child_node; > + struct resource *res = pdev->resource; > + struct cygnus_audio *cygaud; > + int err = -EINVAL; > + int node_count; > + int active_port_count; > + > + if (!of_match_device(cygnus_ssp_of_match, dev)) { > + dev_err(dev, "Failed to find ssp controller\n"); > + return -ENODEV; > + } This error message is misleading - you mean to say that the driver got loaded for a device it doesn't understand. > + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + cygaud->audio = devm_ioremap_resource(dev, res); > + if (IS_ERR(cygaud->audio)) { > + dev_err(dev, "audio_io ioremap failed\n"); > + return PTR_ERR(cygaud->audio); devm_ioremap_resource() will complain for you, and in general you should be printing error codes. > + node_count = 0; This doesn't seem to be needed? > + node_count = of_get_child_count(pdev->dev.of_node); > + if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) { > + dev_err(dev, "Incorrct number of child nodes\n"); > + return -EINVAL; Spelling mistake there and it would be helpful to the user to tell them what we parsed.

1 0

Re: [alsa-devel] [pulseaudio-discuss] Can't get Pulseadio to playback stream on Raspbian with HifiBerry DAC+
by Raymond Yau 31 Mar '15

31 Mar '15

2015-3-30 上午12:22 於 "David Martinez" <david(a)sojournmusic.net> 寫道： > > Label says PiFi DAC+. v2.0 > > I'll need to look up how to apply the patch. I'm not familiar with how to do it. This look like driver bug since capabilities should be pvolume instead of volume if your sound card are half duplex Simple mixer control 'PCM',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 215 Front Left: 172 [80%] Front Right: 172 [80%] Simple mixer control 'Playback',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [-6.00dB] Front Right: 0 [0%] [-6.00dB] Simple mixer control 'Playback Boost',0 Capabilities: volume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 1 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] > >> >> > >> > I'll check HiFiBerry's site and see if I can get it. >> > >> >> https://www.hifiberry.com/dacplus/ >> >> What is the label on the black DAC chip ? >> >> Aplay/Arecord output >> !!-------------------- >> >> APLAY >> >> **** List of PLAYBACK Hardware Devices **** >> card 0: sndrpihifiberry [snd_rpi_hifiberry_dacplus], device 0: HiFiBerry DAC HiFi pcm512x-hifi-0 [] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> >> Seem half duplex >> >>

1 1

[alsa-devel] [RFC v2][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 31 Mar '15

31 Mar '15

This patchset is revised version of my previous set. [alsa-devel] [RFC][PATCH 00/11] digi00x: new driver for Digidesign 002/003 family http://mailman.alsa-project.org/pipermail/alsa-devel/2015-March/089141.html Improvements from the previous set: * improve double-oh-three (DOT) application * extend hwdep interface for asynchronous messaging * limit sampling rate for PCM substream when clock source is external or any PCM substreams are running * support s16 samples for PCM playback In this time, I test PCM functionality with my test programs. As a result, I can confirm that S/PDIF and ADAT output from my Digi 002 Rack includes expected PCM samples as my programs generates. I think we can judge the implementation of DOT is proper for Digi 002 rack. Currently, several issues remains: * The port for MIDI machine control message is not tested yet, because 002/003 console model are required. * When allocates 2 or more channel numbers for the device, after 15 to 20 seconds from playbacking, any PCM samples causes noisy sound. Then, all of LED on the front panel light. The streaming still continues correctly. * The actual effects of external clock source is not clear. When set the clock source is somewhat external, even if stopping the clock source, the device continues to sound PCM samples against my expectation. * The meaning of asynchronous messages is unknown. This patchset adds a functionality to receive it in userspace. You can test it with updated libhinawa sample script. https://github.com/takaswie/libhinawa Well, I'll stop my work for this driver in this developing period (for Linux 4.0), because this work is not in my original plan. I hope any users test this driver and report the result till Linux 4.2 merge window. If not, this driver will not be merged into upstream, naturally. I prepare for DKMS package to test this patchset. Please use it if needed. https://github.com/takaswie/snd-firewire-improve Takashi Sakamoto (11): ALSA: digi00x: add skeleton for Digi 002/003 device driver ALSA: digi00x: add streaming functionality ALSA: digi00x: add proc node for clock status ALSA: digi00x: add PCM functionality ALSA: digi00x: add MIDI functionality ALSA: digi00x: add hwdep interface ALSA: digi00x: support unknown asynchronous message ALSA: digi00x: support MIDI ports for device control ALSA: firewire-lib: allows to implement external MIDI callback function ALSA: digi00x: improve MIDI capture/playback ALSA: digi00x: apply double-oh-three algorithm to multiplex PCM samples include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 10 + sound/firewire/Makefile | 1 + sound/firewire/amdtp.c | 30 ++- sound/firewire/amdtp.h | 5 + sound/firewire/digi00x/Makefile | 3 + sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++++ sound/firewire/digi00x/digi00x-midi.c | 199 +++++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 344 +++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 86 +++++++ sound/firewire/digi00x/digi00x-protocol.c | 406 ++++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-stream.c | 393 +++++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x.c | 179 +++++++++++++ sound/firewire/digi00x/digi00x.h | 167 ++++++++++++ 15 files changed, 2026 insertions(+), 8 deletions(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-protocol.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.0

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Re: [alsa-devel] [PATCH 00/25] treewide: Use bool function return values of true/false not 1/0
by Joe Perches 31 Mar '15

31 Mar '15

On Mon, 2015-03-30 at 17:07 -0700, Casey Schaufler wrote: > On 3/30/2015 4:45 PM, Joe Perches wrote: > > Joe Perches (25): > > arm: Use bool function return values of true/false not 1/0 [etc...] > Why, and why these in particular? bool functions are probably better returning bool values instead of 1 and 0. Especially when the functions intermix returning returning 1/0 and true/false. (there are only a couple of those though) These are all the remaining instances in the kernel tree.

1 0

Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 31 Mar '15

31 Mar '15

> > > Well, I was just trying different modes. The last one was ASUS-MODE1. Before this I tried the "laptop" mode and the "generic" one, both of which did not work. > > What do you suggest? You don't need any model since the pin defaults of your Intel DG41CN are correct unless you want to use CD audio in connector !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x411111f0 0x16 0x411111f0 0x18 0x01a19840 0x19 0x02a19c50 0x1a 0x0181304f 0x1b 0x02214c20 0x1c 0x593301f0 0x1d 0x4004c601 0x1e 0x01441130 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN http://www.intel.com/support/motherboards/desktop/dg41cn/sb/CS-031343.htm You should check the front audio panel of your computer chassis if jack state of headphone is alway plugged http://www.intel.com/support/motherboards/desktop/sb/cs-015851.htm http://fedoraproject.org/wiki/How_to_debug_PulseAudio_problems You have to provide pulseaudio verbose log Default setting support multi channels but NO headphone playback volume control http://www.intel.com/support/motherboards/desktop/sb/CS-034198.htm You can disable multi channels by using early patching and hint multi_io=0 to get back the headphone playback volume control - multi_io (bool): try probing multi-I/O config (e.g. shared line-in/surround, mic/clfe jacks) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >> >> > I have recently installed Fedora 21 Mate on my desktop which runs on >> Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I >> was running Linux Mint. It was working all fine then, I may have fixed >> it when I had installed Mint two years ago but I do not remember what >> was done then. I also run Windows 7 on the same machine and there is >> no problem with the audio. >> >> > So the problem I have is that there is no sound from my speakers. A >> few times, there is a little "choppy" sound but thats it. When I put >> the headphones on, the sound is there. >> >> > I must also mention that I also installed Ubuntu Mate on it and faced >> the same problem. >> >> > I have run the Alsa diagnostic tool and the output is uploaded here: >> http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >> >> >I will really appreciate any help. I am stuck bad here.. >> >> Any reason to use asus-mode1 ? >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex >> >> Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono >> >> + [ALC662_FIXUP_ASUS_MODE1] = { >> + .type = ALC_FIXUP_PINS, >> + .v.pins = (const struct alc_pincfg[]) { >> + { 0x14, 0x99130110 }, /* speaker */ >> + { 0x18, 0x01a19c20 }, /* mic */ >> + { 0x19, 0x99a3092f }, /* int-mic */ >> + { 0x21, 0x0121401f }, /* HP out */ >> + { } >> + }, >> + .chained = true, >> + .chain_id = ALC662_FIXUP_SKU_IGNORE >> + }, >> >> Line out is muted when headphone jack is plugged >> >> control.30 { >> iface CARD >> name 'Front Headphone Jack' >> value true >> comment { >> access read >> type BOOLEAN >> count 1 >> } >> } >> >> Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Control: name="Speaker Playback Switch", index=0, device=0 >> ControlAmp: chs=3, dir=Out, idx=0, ofs=0 >> Control: name="Speaker Phantom Jack", index=0, device=0 >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0001003c: IN OUT HP EAPD Detect >> EAPD 0x2: EAPD >> Pin Default 0x01014010: [Jack] Line Out at Ext Rear >> Conn = 1/8, Color = Green >> DefAssociation = 0x1, Sequence = 0x0 >> Pin-ctls: 0x00: >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0c > >

1 0

Re: [alsa-devel] Realtek ALC662 issues - Intel DG41CN
by Raymond Yau 30 Mar '15

30 Mar '15

> I have recently installed Fedora 21 Mate on my desktop which runs on Intel DG41CN motherboard with Realtek ALC662 Audio. Prior to this I was running Linux Mint. It was working all fine then, I may have fixed it when I had installed Mint two years ago but I do not remember what was done then. I also run Windows 7 on the same machine and there is no problem with the audio. > So the problem I have is that there is no sound from my speakers. A few times, there is a little "choppy" sound but thats it. When I put the headphones on, the sound is there. > I must also mention that I also installed Ubuntu Mate on it and faced the same problem. > I have run the Alsa diagnostic tool and the output is uploaded here: http://www.alsa-project.org/db/?f=1f6dd98e26f68e890258fa2fe5eb08ee343c5c13 >I will really appreciate any help. I am stuck bad here.. Any reason to use asus-mode1 ? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem asus-mode1 is a broken pin fixup since node 0x21 is not any pin complex Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono + [ALC662_FIXUP_ASUS_MODE1] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x01a19c20 }, /* mic */ + { 0x19, 0x99a3092f }, /* int-mic */ + { 0x21, 0x0121401f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_SKU_IGNORE + }, Line out is muted when headphone jack is plugged control.30 { iface CARD name 'Front Headphone Jack' value true comment { access read type BOOLEAN count 1 } } Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c

1 0

Re: [alsa-devel] baytrail-pcm-audio: audio broken on Lenovo Thinkpad Tablet 10
by Sébastien Bourdeauducq 30 Mar '15

30 Mar '15

Hi, On 02/23/2015 08:58 PM, Jarkko Nikula wrote: >> Is there a way I can try sound/soc/codecs/rt5670.c instead? I edited it >> so that it attaches to my ACPI ID (and rt5640 doesn't), but that did not >> work. I noticed that the kernel has some Baytrail-specific modules for >> RT564x, and I guess those would need to be changed as well - but it's >> not straightforward. >> > Basically three changes are needed: > > 1. Hack add "10EC5640" to rt5670.c as you have done > 2. Add a new machine driver sound/soc/intel/sst-acpi.c/byt-rt5670 > I guess cht_bsw_rt5672.c can be used as a starting point for audio > routes and byt-rt5640.c for dai link definitions.mess > 3. Hack change sound/soc/intel/sst-acpi.c so that "10EC5640" loads > the new machine driver instead of byt-rt5640. Thanks. I did that and I get the following in the kernel log: [ 6.831422] byt-rt5670 byt-rt5670: ASoC: CPU DAI baytrail-pcm-audio not regis tered [ 6.831443] byt-rt5670 byt-rt5670: snd_soc_register_card failed -517 [ 6.831453] platform byt-rt5670: Driver byt-rt5670 requests probe deferral [ 6.884013] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 6.884023] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 Note: it seems to me that all Baytrail machines will have this "error: invalid DMA engine" message. AFAICT it is emitted from sst-firmware.c because sst->pdata->dma_engine is 0 and in sst-acpi.c the function sst_acpi_probe does not set it for Baytrail because resindex_dma_base is -1 in sst_acpi_baytrail_desc. [ 6.924094] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 6.924106] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 6.924109] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 6.931503] byt-rt5670 byt-rt5670: rt5670-aif1 <-> baytrail-pcm-audio mapping ok [ 6.931624] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Tx [ 6.931629] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Tx -> direct -> AIF1 Playback [ 6.931636] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out0 [ 6.931640] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out0 -> direct -> ssp2 Tx [ 6.931647] byt-rt5670 byt-rt5670: ASoC: no source widget found for codec_out1 [ 6.931650] byt-rt5670 byt-rt5670: ASoC: Failed to add route codec_out1 -> direct -> ssp2 Tx [ 6.931656] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931660] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in0 [ 6.931666] byt-rt5670 byt-rt5670: ASoC: no source widget found for ssp2 Rx [ 6.931670] byt-rt5670 byt-rt5670: ASoC: Failed to add route ssp2 Rx -> direct -> codec_in1 [ 6.931677] byt-rt5670 byt-rt5670: ASoC: no sink widget found for ssp2 Rx [ 6.931680] byt-rt5670 byt-rt5670: ASoC: Failed to add route AIF1 Capture -> direct -> ssp2 Rx My byt-rt5670 is there: https://gist.github.com/sbourdeauducq/8c5da8ac0da13e21ee31 I have to say that I am unfamiliar with Intel SST and I'm don't quite know what I am doing there. I wish sound devices were straightforward DMA-to-DAC devices instead of this complicated DSP architecture which AFAICT boils down to a glorified memcpy/resampler under Linux (this seems to be a pattern with this SoC though; I had a look at how to get the MIPI cameras to work on this machine and it's an even worse mess). > However, above changes may not make audio working since there has been > couple other new Baytrail based machines with correct codec but audio > seems to not move between the DSP and codec chip. Despite the error messages, the audio device is created and userspace applications can send data into it, but no sound is coming out. I also have to say that the settings in alsamixer are a terrible mess which seems not to originate from my hacks. > One assumption is if they use another SSP port than SSP2. Unfortunately > it's difficult to test since Linux DSP FW is hard-coded for SSP2. If only there were no DSP in the first place... Sébastien

1 0

[alsa-devel] [PATCH 0/5 v2] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 30 Mar '15

30 Mar '15

Hi Mark These are v2 of remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 35 ++++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 31 +++--- sound/soc/sh/rcar/gen.c | 6 -- sound/soc/sh/rcar/rsnd.h | 19 +++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 93 +++++++++++------- sound/soc/sh/rcar/ssi.c | 36 ++++--- 12 files changed, 729 insertions(+), 86 deletions(-) _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

3 7

[alsa-devel] Atomic operations not multi-thread aware
by Joël Krähemann 30 Mar '15

30 Mar '15

Hi, please take a a look at gsequencer.org especially at: libags-audio * http://gsequencer.org/api/ags/ags-AgsPort.html gsequencer * http://gsequencer.org/api/ags/ags-AgsLineMember.html I can crash my application while modifying values of the LADSPA plugin assigned dial widget what uses atomic operations and a task engine ... Where can me read about state of Atomic Operations of linux or glibc? Me doing further investigation. bests Joël

2 2

[alsa-devel] [PATCH v2 0/2] ALSA: jack: Refactoring for jack kctls
by Jie Yang 30 Mar '15

30 Mar '15

Currently only hda will create kctls for hda jacks, for asoc, people may need create jack kctls in specific driver. Here we are introducing kctls for each jack, by creating kctls and add them to jack controls list (considering exist of combo jack). At the same time, we will report events for each control in the list. With this implement, we are no longer need hda jack specific kctls, so here also remove jack kctls for hda. Any comments are welcome. Jie Yang (2): ALSA: jack: create jack kcontrols for every jack input device ALSA: hda - Remove jack kctls include/sound/jack.h | 8 ++++ sound/core/Kconfig | 3 -- sound/core/Makefile | 3 +- sound/core/jack.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++- sound/pci/hda/Kconfig | 10 +---- sound/pci/hda/hda_jack.c | 45 ++-------------------- sound/pci/hda/hda_jack.h | 3 -- 7 files changed, 110 insertions(+), 61 deletions(-) -- 1.9.1

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Re: [alsa-devel] getting the sound to work on my asus m6s00v
by Raymond Yau 28 Mar '15

28 Mar '15

2015-3-28 下午5:52 於 "Christian Gleerup" <cromozon(a)swn.nu> 寫道： > > Trying again with my previous kernel since the new kernel seems broken regarding loading the sound module > > nothin plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = Yes > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > > Headphone plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = No > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = No > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0xffffffff 0xffffffff = 0x80000000 + 0x7ffffff Try the modified version of hda_jack_sense_test.py which need hda_analyzer 's codec parser and alsa python https://bugzilla.kernel.org/show_bug.cgi?id=66621#c39 it need time for the impedance measurement to complete and become stable, You may need to increase time.sleep(0.02) if you still get 0xffffffff https://bugzilla.kernel.org/show_bug.cgi?id=66621#c41 # the hda_codec.py file comes from Jaroslav Kysela's hda_analyzer program. from hda_codec import * import time def parseoptions(): from optparse import OptionParser parser = OptionParser() parser.add_option("-c", "--card", dest="cardindex", default=0, metavar="CARD", help="card index (as can be seen in /proc/asound/cards)") parser.add_option("-i", "--codec", dest="codecindex", default=0, metavar="CODEC", help="codec index (as can be seen in /proc/asound/cardX/codecY)") (options, args) = parser.parse_args() return int(options.cardindex), int(options.codecindex) cardindex, codecindex = parseoptions() codec = HDACodec(cardindex, codecindex) codec.analyze() for nid in codec.nodes: node = codec.get_node(nid) if WIDGET_TYPE_IDS[node.wtype] == 'PIN': pd = " " impedance = " " if ((node.pincaps & 4) == 4): if (node.pincaps & 2): codec.rw(nid, VERBS['SET_PIN_SENSE'], 0) time.sleep(0.02) sense = codec.rw(nid, VERBS['GET_PIN_SENSE'], 0) if (node.pincaps & 1): impedance = "Imp=%d" % (sense & 0x7fffffff) if (sense & 0x80000000): pd = "PD=1" else: pd = "PD=0" print "Pin 0x%.2x [%s] %s %s at %s (%s) %s : %s %s" % (nid, node.jack_conn_name, node.jack_type_name, node.jack_location_name, node.jack_location2_name, node.jack_color_name, node.defcfg_misc , pd, impedance) > > Mic plugged in > ../hdajacksensetest/hdajacksensetest -a; for i in 0x14 0x15 0x15; do ./hda-verb /dev/snd/hwC0D0 $i SET_PIN_SENSE 0 nid ;./hda-verb /dev/snd/hwC0D0 $i GET_PIN_SENSE 0 nid; done > Pin 0x14 (Green Headphone, Rear side): present = Yes > Pin 0x15 (Not connected): present = No > Pin 0x16 (Not connected): present = No > Pin 0x17 (Not connected): present = No > Pin 0x18 (Pink Mic, Rear side): present = No > Pin 0x19 (Not connected): present = No > Pin 0x1a (Not connected): present = Yes > Pin 0x1b (Not connected): present = No > Pin 0x1c (Internal CD, Mobile-In): present = No > Pin 0x1d (Not connected): present = No > Pin 0x1e (Green SPDIF Out, Rear side): present = No > Pin 0x1f (Not connected): present = No > nid = 0x14, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x14, verb = 0xf09, param = 0x0 > value = 0xffffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > nid = 0x15, verb = 0x709, param = 0x0 > value = 0x0 > nid = 0x15, verb = 0xf09, param = 0x0 > value = 0x7fffffff > > > > > >

1 0

[alsa-devel] getting the sound to work on my asus m6s00v
by Christian Gleerup 28 Mar '15

28 Mar '15

I had some problems getting the sound to work on my asus m6s00v (I think that is the model, it is a bit worn so I can't really read it) the symptoms was * on the integrated speakers, no sound at all * on the headphone speakers, sound for a very very short period that faded into nothing. with help from debianuser on freenode#alsa we managed to get it working. the option that i had to load the module with is the following: sudo modprobe snd-hda-intel model=3stack-digout the status now (perfect)... * sound on integrated speaker. * sound in headphones (disables integrated speaker output which is fine) * mic works, both for direct playback and recording * when recording it is the 'digital' channel that controls the volume of the recording level, all others Notice: I don't have OSS or pulseaudio installed. I am thinking it would be really nice if the driver automatically could detect that this option is required. this is the output from the alsa-info script http://www.alsa-project.org/db/?f=28b98a3bd1efc6a80dbd5c54b52c4472d8ea5496 let me know if I can do anything to help with this. Regards.

3 18

[alsa-devel] [PATCH RFT] ASoC: max98925: Fix bit-width 24 settings in max98925_dai_hw_params
by Axel Lin 27 Mar '15

27 Mar '15

Trivial typo fix. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi Anish, This looks like a typo, can you confirm if this patch is correct. Thanks. sound/soc/codecs/max98925.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c index 0796dfa..9b5a17d 100644 --- a/sound/soc/codecs/max98925.c +++ b/sound/soc/codecs/max98925.c @@ -442,8 +442,8 @@ static int max98925_dai_hw_params(struct snd_pcm_substream *substream, case 24: regmap_update_bits(max98925->regmap, MAX98925_FORMAT, - M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); - max98925->ch_size = 32; + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_24); + max98925->ch_size = 24; break; case 32: regmap_update_bits(max98925->regmap, -- 1.9.1

2 1

Re: [alsa-devel] snd_usb_audio - siberia elite prism soundcard issues
by Raymond Yau 27 Mar '15

27 Mar '15

>> >> > >> >> > >> >> numid=11,iface=MIXER,name='Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=74,step=0 : values=54,54,54,54,54,54,54,54 | dBminmax-min=-74.00dB,max=0.00dB >> >> >> >> You need workaround since pulseaudio don't like volume control with more than two channels >> >> >> >> https://bugs.freedesktop.org/show_bug.cgi?id=84983#c35 >> >> >> >> Can you post output of your usb audio device ? >> >> >> >> lsusb -vvvv >> > >> > >> > >> > Certainly! >> > >> > Here it is: >> > >> > Bus 001 Device 003: ID 1038:1225 Ideazon, Inc. >> > Device Descriptor: >> > bLength 18 >> > bDescriptorType 1 >> > bcdUSB 2.00 >> > bDeviceClass 0 (Defined at Interface level) >> > bDeviceSubClass 0 >> > bDeviceProtocol 0 >> > bMaxPacketSize0 64 >> > idVendor 0x1038 Ideazon, Inc. >> > idProduct 0x1225 >> > bcdDevice 0.03 >> > iManufacturer 1 SteelSeries >> > iProduct 2 SteelSeries Siberia Elite >> > iSerial 3 000000000000 >> > bNumConfigurations 1 >> >> > >> > AudioControl Interface Descriptor: >> > bLength 13 >> > bDescriptorType 36 >> > bDescriptorSubtype 4 (MIXER_UNIT) >> > Warning: Descriptor too short >> >> https://github.com/gregkh/usbutils/blob/master/lsusb.c >> >> Seem no warning for descriptor too short for uac2 and can dump those uac2 specific variables >> >> > bUnitID 10 >> > bNrInPins 2 >> > baSourceID( 0) 9 >> > baSourceID( 1) 1 >> > bNrChannels 8 >> > wChannelConfig 0x063f >> > Left Front (L) >> > Right Front (R) >> > Center Front (C) >> > Low Freqency Enhancement (LFE) >> > Left Surround (LS) >> > Right Surround (RS) >> > Side Left (SL) >> > Side Right (SR) >> > iChannelNames 0 >> > bmControls 0x00 >> > bmControls 0x00 >> > iMixer 33 (error) >> > > > > Maybe it's the language barrier, but I am not sure if you wanted me to do something. I, out of curiosity, tried to compile that lsusb but I'm missing headers such as config.h. > > Should you need any more feedback from me about this device, please do not hesitate asking through mailing list or directly, just consider I'm not dev! :) > Do you mean there is no sound using alsa hw device since pulseaudio does not support the multichannel speaker playback volume comtrol ? speaker-test -c8 -t wav -D hw:CARD=Elite speaker-test -c2 -t wav -D hw:CARD=Elite You need to know how to compile alsa driver for you running kernel if you want to fix the problem >> >> > > > amixer: Control hw:2 element read error: Invalid argument If alsamixer or amixer cannot set the speaker playback volume Try skip the problematic PCM Volume controls by adding if (state->chip->usb_id == USB_ID(0x1038,0x1225)) if(unitid == 10) ich_has_control = 0; If the driver really need this mixer unit for the playback path, you need to fix those controls bUnitID 10 bNrInPins 2 baSourceID( 0) 9 microphone baSourceID( 1) 1 usb streaming This look like the first four problematic controls related to microphone and the last control related to streaming numid=5,iface=MIXER,name='PCM Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=6,iface=MIXER,name='PCM Volume',index=1 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=7,iface=MIXER,name='PCM Volume',index=2 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=8,iface=MIXER,name='PCM Volume',index=3 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument numid=9,iface=MIXER,name='PCM Volume',index=4 ; type=INTEGER,access=rw------,values=1,min=0,max=1,step=0 amixer: Control hw:2 element read error: Invalid argument

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

2 4

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Redefine format config for rt5650
by Bard Liao 27 Mar '15

27 Mar '15

rt5650 and rt5645 use different register bits for format configuration. This patch modifies rt5645_hw_params and rt5645_set_dai_fmt to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Mark, I am think if we should use different codec driver for rt5645 and rt5650. In fact, they are different Die. What do you think? --- --- sound/soc/codecs/rt5645.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4c384a1..0133c8c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2049,7 +2049,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk; + unsigned int val_len = 0, val_clk, mask_clk, dl_sft; int pre_div, bclk_ms, frame_size; rt5645->lrck[dai->id] = params_rate(params); @@ -2063,6 +2063,16 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); return -EINVAL; } + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + dl_sft = 4; + break; + default: + dl_sft = 2; + break; + } + bclk_ms = frame_size > 32; rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); @@ -2075,13 +2085,13 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, case 16: break; case 20: - val_len |= RT5645_I2S_DL_20; + val_len = 0x1; break; case 24: - val_len |= RT5645_I2S_DL_24; + val_len = 0x2; break; case 8: - val_len |= RT5645_I2S_DL_8; + val_len = 0x3; break; default: return -EINVAL; @@ -2093,7 +2103,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | pre_div << RT5645_I2S_PD1_SFT; snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; case RT5645_AIF2: @@ -2101,7 +2111,7 @@ static int rt5645_hw_params(struct snd_pcm_substream *substream, val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | pre_div << RT5645_I2S_PD2_SFT; snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_DL_MASK, val_len); + (0x3 << dl_sft), (val_len << dl_sft)); snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); break; default: @@ -2116,7 +2126,16 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0; + unsigned int reg_val = 0, pol_sft; + + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + pol_sft = 8; + break; + default: + pol_sft = 7; + break; + } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -2134,7 +2153,7 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_NF: - reg_val |= RT5645_I2S_BP_INV; + reg_val |= (1 << pol_sft); break; default: return -EINVAL; @@ -2158,12 +2177,12 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (dai->id) { case RT5645_AIF1: snd_soc_update_bits(codec, RT5645_I2S1_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; case RT5645_AIF2: snd_soc_update_bits(codec, RT5645_I2S2_SDP, - RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_MS_MASK | (1 << pol_sft) | RT5645_I2S_DF_MASK, reg_val); break; default: -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH v4 0/15] Those patches is used for dw_hdmi audio support
by Yakir Yang 27 Mar '15

27 Mar '15

We found Designware hdmi driver only support audio clock config, we can not play sound through it. To add Designware HDMI Audio support, we make those patch set: 1): fixed dw_hdmi irq bug, add irq control to suspend/resume interfaces. 2): add suspend/resume callback for dw_hdmi rockchip driver. 3): Warp irq control in functions. 4): Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() 5): Adjust n/cts config order. 6): Set ncts_atomic_write & cts_manual, according to dw_hdmi document. 7): Add audio support for more display resolutions(eg. 800x600). 8): Add audio support for No-CEA display resolutions. 9): Add Audio Sample Channel Status config interfaces to dw_hdmi driver. 10): add audio clock control interfaces to dw_hdmi driver. 11): creat "dw_hdmi-audio" platform device in dw_hdmi driver. 12): add codec driver for hdmi audio, callback dw_hdmi audio config functions. 13): add sound driver for hdmi audio, creat hdmi audio sound card. 14): add dt-bings file and add hdmi_audio node to corresponding dt file. Changes in v4: -Correct phy_type assignment bug - Combine CTS3 registers setting, reduce register operate times - Combine N3 registers setting - Add hdmi audio support when monitor support audio - Give HDMI_FC_AUD_SCHNL8 an readable value - Rename "hdmi_audio_*" to "dw_hdmi_audio_*" - Replace delaywork with irq thread, and add suspend/resume interfaces, Replace "dw-hdmi-audio" with consecutive strings. - Add ".pm = &snd_soc_pm_ops," Changes in v3: - Clear Hotplug interrupts before dw_hdmi_fb_register - Wrap irq control in fucntions - Setting the .pm member instead of suspend/resume - Add ID registers parse and record - Combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() - Only adjust the n/cts setting order - Set ncts_atomic_write & cts_manual - Determine whether sample channel should set by desig_id. - Delete hdmi_audio_config interface and modify audio clock control functions - Remove audio_config & get_connect_status callback functions and add write/read/mod register callback functions - Keep audio format config function in dw-hdmi-audio driver and remove audio_config & get_connect_status functions, move jack control to dw-hdmi-audio completely. - Delete the operation of jack in rockchip-hdmi-audio driver, get ready to switch to simple-audio-card driver. - modify cpu-of-node to i2s-controller Changes in v2: - Add irq control to suspend/resume interfaces - Add suspend/resume support for dw_hdmi_rockchip driver - add more n/cts combinations for more display resolutions - Enable audio support for No-CEA display mode - Add audio sample channel status setting - Add audio config interfaces to dw_hdmi driver - Update the audio control interfaces - Update dw_hdmi audio control interfaces, and adjust jack report process - give "codec-name" & "codec-dai-name" an const name - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt Daniel Kurtz (3): drm: bridge/dw_hdmi: adjust n/cts setting order drm: bridge/dw_hdmi: set ncts_atomic_write & cts_manual drm: bridge/dw_hdmi: add audio sample channel status setting Yakir Yang (12): drm: bridge/dw_hdmi: add irq control to suspend/resume drm: bridge/dw_hdmi: wrap irq control in fucntions drm: rockchip/dw_hdmi_rockchip: add resume/suspend support drm: bridge/dw_hdmi: add identification registers parse and record drm: bridge/dw_hdmi: combine hdmi_set_clock_regenerator_n() and hdmi_regenerate_cts() drm: bridge/dw_hdmi: add audio support for more display resolutions drm: bridge/dw_hdmi: enable audio support for No-CEA display resolutions drm: bridge/dw_hdmi: add enable/disable to dw_hdmi_audio callbacks drm: bridge/dw_hdmi: creat dw-hdmi-audio platform device ASoC: codec/dw-hdmi-audio: add codec driver for dw hdmi audio ASoC: rockchip/rockchip-hdmi-audio: add sound driver for hdmi audio dt-bindings: Add documentation for Rockchip dw-hdmi-audio .../sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 + drivers/gpu/drm/bridge/dw_hdmi.c | 385 ++++++++++++++++++--- drivers/gpu/drm/bridge/dw_hdmi.h | 53 ++- drivers/gpu/drm/rockchip/dw_hdmi-rockchip.c | 16 + include/drm/bridge/dw_hdmi.h | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 379 ++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 74 ++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 169 +++++++++ 12 files changed, 1076 insertions(+), 46 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c -- 2.1.2

4 11

[alsa-devel] [PATCH] ASoC: Add support for NAU8824 codec to ASoC
by Wan Zongshun 27 Mar '15

27 Mar '15

Signed-off-by: Wan Zongshun <mcuos.com(a)gmail.com> Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8824.c | 770 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8824.h | 379 ++++++++++++++++++++++ 4 files changed, 1156 insertions(+) create mode 100644 sound/soc/codecs/nau8824.c create mode 100644 sound/soc/codecs/nau8824.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c1..862b7bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8824 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -463,6 +464,10 @@ config SND_SOC_MAX98357A config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8824 + tristate "Nuvoton NAU8824 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666..acb7bda 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max98357a-objs := max98357a.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8824-objs := nau8824.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -250,6 +251,7 @@ obj-$(CONFIG_SND_SOC_MAX98357A) += snd-soc-max98357a.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c new file mode 100644 index 0000000..80c2283 --- /dev/null +++ b/sound/soc/codecs/nau8824.c @@ -0,0 +1,770 @@ +/* + * linux/sound/soc/codecs/nau8824.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> + +#include "nau8824.h" + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt); +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); +static int nau8824_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); +static const DECLARE_TLV_DB_SCALE(out_spk_vol_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -3000, 100, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -12800, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_left_vol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_right_vol_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new nau8824_snd_controls[] = { + /* SP Class-D mute control */ + SOC_DOUBLE("SP Playback Switch", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, NAU8824_CLASS_D_SFT, 1, 0), + /* SP Class-D driver output stage gain control */ + SOC_SINGLE_TLV("SP Left Volume", NAU8824_CLASS_D_GAIN2, + NAU8824_CLASS_D_GAIN_L_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, out_spk_vol_tlv), + SOC_SINGLE_TLV("SP Right Volume", NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_GAIN_R_SFT, + NAU8824_SPK_VOL_RSCL_RANGE, 0, + out_spk_vol_tlv), + /* SP Class-D mute control */ + SOC_DOUBLE("HP Playback Switch", NAU8824_HP_MUTE, NAU8824_L_MUTE_SFT, + NAU8824_R_MUTE_SFT, 1, 1), + SOC_SINGLE_TLV("HP Left Volume", NAU8824_HP_VOL, NAU8824_L_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + SOC_SINGLE_TLV("HP Right Volume", NAU8824_HP_VOL, NAU8824_R_VOL_SFT, + NAU8824_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), + /* DMIC control */ + SOC_DOUBLE("DMIC L R Switch", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + NAU8824_DMIC_CH1_SFT, 1, 0), + SOC_SINGLE("DMIC Enable", NAU8824_BIAS_ADJ, NAU8824_DMIC1_SFT, 1, 0), + SOC_SINGLE("VMID Switch", NAU8824_BIAS_ADJ, NAU8824_VMID_SFT, 1, 0), + SOC_SINGLE("BIAS Switch", NAU8824_BOOST, NAU8824_G_BIAS_SFT, 1, 0), + SOC_DOUBLE_R_TLV("MIC L R Gain", NAU8824_ADC_CH0_DGAIN_CTRL, + NAU8824_ADC_CH1_DGAIN_CTRL, 0, + NAU8824_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), +}; + +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, NAU8824_CLK_DIVIDER, + NAU8824_CLK_DMIC_SRC_MASK, + NAU8824_CLK_DMIC_DIV_4); + break; + default: + break; + } + return 0; +} + +static const struct snd_kcontrol_new nau8824_rec_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_rec_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_spo_mix[] = { + SOC_DAPM_SINGLE("SP L Switch", SND_SOC_NOPM, + 0, 0, 0), + SOC_DAPM_SINGLE("SP R Switch", SND_SOC_NOPM, + 0, 0, 0), +}; + +static const struct snd_kcontrol_new nau8824_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_L_HM_SFT, + 1, 0), + SOC_DAPM_SINGLE("HP R Switch", + NAU8824_HPO_MIXER, NAU8824_M_HPVOL_R_HM_SFT, + 1, 0), +}; + +static const char * const nau8824_stereo_adc1_src[] = { "ADC", "DMIC" }; + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_l1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH0_SFT, nau8824_stereo_adc1_src); + +static const SOC_ENUM_SINGLE_DECL(nau8824_stereo_adc_r1_enum, NAU8824_ENA_CTRL, + NAU8824_DMIC_CH1_SFT, nau8824_stereo_adc1_src); + +static const struct snd_kcontrol_new nau8824_sto_adc_l1_mux = + SOC_DAPM_ENUM("Stereo ADC L1 source", + nau8824_stereo_adc_l1_enum); + +static const struct snd_kcontrol_new nau8824_sto_adc_r1_mux = + SOC_DAPM_ENUM("Stereo ADC R1 source", + nau8824_stereo_adc_r1_enum); + +static int nau8824_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, NAU8824_ANALOG_CTRL2, + NAU8824_CLASS_D_CLAMP_MSK, + NAU8824_CLASS_D_CLAMP_BETTER); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_MASK, NAU8824_CLASS_D_DIS); + default: + break; + } + return 0; +} + +static int nau8824_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int nau8824_dac1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + return 0; +} + +static int micbias_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BOOST, NAU8824_G_BIAS_MASK, + NAU8824_G_BIAS_DIS); + return 0; +} + +static int vmid_power_on_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + if (SND_SOC_DAPM_EVENT_ON(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + else if (SND_SOC_DAPM_EVENT_OFF(event)) + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_DIS); + return 0; +} + +static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { + + SND_SOC_DAPM_INPUT("DMIC1"), + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_SUPPLY("micbias", SND_SOC_NOPM, 0, 0, + micbias_power_on_event, SND_SOC_DAPM_PRE_PMU + | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("vmid", SND_SOC_NOPM, 0, 0, vmid_power_on_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_PGA("DMIC L1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH0_SFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("DMIC R1", NAU8824_ENA_CTRL, NAU8824_DMIC_CH1_SFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, + SND_SOC_DAPM_PRE_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", SND_SOC_NOPM, 0, 0, nau8824_rec_l_mix, + ARRAY_SIZE(nau8824_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", SND_SOC_NOPM, 0, 0, nau8824_rec_r_mix, + ARRAY_SIZE(nau8824_rec_r_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &nau8824_sto_adc_r1_mux), + /* ADC IF1 */ + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC L1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_L_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC R1", NULL, NAU8824_DAC_CTRL, + NAU8824_DAC_R_SFT, 0, nau8824_dac1_event, + SND_SOC_DAPM_PRE_PMD), + + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPO MIX", SND_SOC_NOPM, 0, 0, nau8824_spo_mix, + ARRAY_SIZE(nau8824_spo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8824_hpo_mix, + ARRAY_SIZE(nau8824_hpo_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, NAU8824_CLASS_G_CTRL, + NAU8824_CLASS_G_SHIFT, 0, nau8824_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 1, NAU8824_CLASS_D_GAIN1, + NAU8824_CLASS_D_SFT, 0, nau8824_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8824_dapm_routes[] = { + + {"DMIC L1", NULL, "DMIC1"}, + {"DMIC R1", NULL, "DMIC1"}, + + {"BST1", NULL, "IN1P"}, + {"BST1", NULL, "IN1N"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"IN2P", NULL, "micbias"}, + {"IN2P", NULL, "vmid"}, + + {"RECMIXL", "BST1 Switch", "BST1"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC R", NULL, "RECMIXR"}, + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC CLK"}, + + {"Stereo ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo ADC L1 Mux", "DMIC", "DMIC L1"}, + {"Stereo ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo ADC R1 Mux", "DMIC", "DMIC R1"}, + {"AIF1TX", NULL, "IF1 ADC"}, + + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"SPO MIX", "SP L Switch", "DAC L1"}, + {"SPO MIX", "SP R Switch", "DAC R1"}, + {"SPK amp", NULL, "SPO MIX"}, + {"SPK", NULL, "SPK amp"}, + + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HP amp", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8824_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int val_len = 0; + + dev_dbg(codec->dev, "%s\n", __func__); + dev_dbg(codec->dev, "##- Data length: %d\n", params_format(params)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= NAU8824_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= NAU8824_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + case SNDRV_PCM_FORMAT_S32_LE: + val_len |= NAU8824_I2S_DL_32; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, NAU8824_I2S_DL_MASK, + val_len); + return 0; +} + +static int nau8824_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_EN); + else + snd_soc_update_bits(codec, NAU8824_DAC_MUTE_CTRL, + NAU8824_SOFT_MUTE_MASK, NAU8824_SOFT_MUTE_DIS); + return 0; +} + +static int nau8824_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int reg_val = 0, reg_val_2 = 0; + + dev_dbg(codec->dev, "%s: Entered\n", __func__); + dev_dbg(codec->dev, "###nau8824_set_dai_fmt %x\n", fmt); + dev_dbg(codec->dev, "##+ nau8824_set_dai_fmt (%x)\n", fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val_2 |= NAU8824_I2S_MS_M; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8824_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8824_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8824_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8824_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8824_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8824_I2S_DF_PCM_B; + reg_val |= NAU8824_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_1, + NAU8824_I2S_DL_MASK | NAU8824_I2S_DF_MASK + | NAU8824_I2S_BP_MASK | NAU8824_I2S_PCMB_MASK, + reg_val); + snd_soc_update_bits(codec, NAU8824_I2S_PCM_CTRL_2, NAU8824_I2S_MS_MASK, + reg_val_2); + + dev_dbg(codec->dev, "##-nau8824_set_dai_fmt Master\n"); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_FLL_freerun_mode(struct snd_soc_codec *codec, int IsEnable) +{ + if (IsEnable == true) { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, 0x03, 0x8853); + } else { + snd_soc_write(codec, 0x03, 0x0853); + snd_soc_write(codec, 0x09, 0x6000); + snd_soc_write(codec, 0x03, 0x8853); + } + return 0; +} + +void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + pr_debug("%s sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8824_INTERNALCLOCK: + nau8824_FLL_freerun_mode(codec, true); + break; + + case NAU8824_MCLK: + default: + nau8824_FLL_freerun_mode(codec, false); + break; + } +} + +static int nau8824_set_dai_pll(struct snd_soc_dai *dai, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "In nau8824: dai_set_pll\n"); + return 0; +} + +static int nau8824_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + int divider = 1; + + if (clk_id == NAU8824_MCLK) { + set_sys_clk(codec, NAU8824_MCLK); + dev_dbg(codec->dev, "%s: input freq = %d divider = %d", + __func__, freq, divider); + + } else if (clk_id == NAU8824_INTERNALCLOCK) { + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + } else { + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8824_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + if (level == codec->dapm.bias_level) { + dev_dbg(codec->dev, "##set_bias_level: level returning...\r\n"); + return -EINVAL; + } + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_ON\n"); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, + NAU8824_VMID_MASK, NAU8824_VMID_EN); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_EN); + break; + + case SND_SOC_BIAS_PREPARE: + dev_dbg(codec->dev, "###nau8824_set_bias_level BIAS_PREPARE\n"); + break; + + case SND_SOC_BIAS_STANDBY: + dev_dbg(codec->dev, "###nau8824_set_bias_level STANDBY\n"); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8824_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8824_INTERNALCLOCK); + snd_soc_update_bits(codec, NAU8824_BIAS_ADJ, NAU8824_VMID_MASK, + NAU8824_VMID_DIS); + snd_soc_update_bits(codec, NAU8824_BOOST, + NAU8824_G_BIAS_MASK, NAU8824_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8824_set_bias_level %d\n", level); + + return 0; +} + +static int nau8824_suspend(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +static int nau8824_resume(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "%s: Entered\n", __func__); + nau8824_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dev_dbg(codec->dev, "%s: Exiting\n", __func__); + return 0; +} + +struct nau8824_init_reg { + u8 reg; + u16 val; +}; + +static struct nau8824_init_reg init_list[] = { + {0x0000, 0x0001}, + {0x0000, 0x0000}, + {0x0066, 0x0060}, + {0x0076, 0x3000}, + {0x0004, 0x0002}, + {0x0005, 0xDD44}, + {0x0006, 0x0007}, + {0x0007, 0x0810}, + {0x0008, 0xC000}, + {0x0074, 0x1106}, + {0x0073, 0x330C}, + {0x0080, 0x0000}, + {0x006D, 0x0100}, + {0x006E, 0x0001}, + {0x006A, 0x0008}, + {0x006D, 0x0180}, + {0x006D, 0x1480}, + {0x006E, 0x0014}, + {0x0050, 0x2007}, + {0x0080, 0x3020}, + {0x006B, 0xC000}, + {0x007B, 0x1E1E}, + {0x006B, 0xC005}, + {0x0068, 0x8000}, + {0x0076, 0x300F}, + {0x007F, 0x000F}, + {0x0080, 0x0020}, + {0x006B, 0x0005}, + {0x007B, 0x0000}, + {0x0002, 0x0800}, + {0x0025, 0x1000}, + {0x0024, 0x0002}, + {0x0002, 0x0AA5}, + {0x0020, 0x0000}, + {0x0025, 0x1082}, + {0x0031, 0x0800}, + {0x0032, 0x0100}, + {0x0033, 0x0300}, + {0x0034, 0x0000}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0001, 0x0030}, + {0x0002, 0x3AA5}, + {0x0067, 0x4040}, + {0x007F, 0x300F}, + {0x0077, 0x0010}, + {0x0078, 0x3C00}, + {0x0076, 0x380F}, + {0x0076, 0x300F}, + {0x007A, 0x0202}, + {0x0067, 0x0000}, + {0x0072, 0x40A0}, + {0x0079, 0x0300}, + {0x0023, 0x1010}, + {0x0024, 0x0002}, + {0x0038, 0x1486}, + {0x003C, 0x1486}, + {0x0066, 0x0064}, + {0x0001, 0x40F0}, + {0x0001, 0x48F0}, + {0x0025, 0x3082}, + {0x0020, 0x0003}, + {0x0024, 0x8102}, + {0x002D, 0x0108}, + {0x002E, 0x0308}, + {0x002F, 0x0500}, + {0x0030, 0x0700}, + {0x001C, 0x000A}, + {0x001F, 0x2000}, + {0x0002, 0x3BA5}, + {0x0001, 0x48F3}, + {0x0002, 0xFBA5}, + {0x0014, 0x2210}, + {0x007c, 0x1e1e}, +}; + +#define NAU8824_INIT_REG_LEN ARRAY_SIZE(init_list) +static int nau8824_reg_init(struct snd_soc_codec *codec) +{ + int i; + + mdelay(1); + for (i = 0; i < NAU8824_INIT_REG_LEN; i++) { + if (init_list[i].reg == 0xFFFF) + mdelay(1); + else + snd_soc_write(codec, init_list[i].reg, + init_list[i].val); + } + return 0; +} + +static int nau8824_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct nau8824_priv *nau8824; + + nau8824 = snd_soc_codec_get_drvdata(codec); + codec->control_data = nau8824->regmap; + mutex_init(&nau8824->mutex); + nau8824_reg_init(codec); + + /* Dynamic Headset detection enabled */ + snd_soc_update_bits(codec, 0x01, 0x400, 0x400); + snd_soc_update_bits(codec, 0x02, 0x0008, 0x0008); + snd_soc_update_bits(codec, 0x0f, 0x0300, 0x0100); + snd_soc_write(codec, 0x09, 0xE000); + snd_soc_write(codec, NAU8824_IRQ_SETTING, 0x1006); + snd_soc_write(codec, 0x13, 0x1615); + snd_soc_write(codec, 0x15, 0x0414); + snd_soc_update_bits(codec, 0x16, 0xFF00, 0x5900); + snd_soc_update_bits(codec, 0x66, 0x0070, 0x0060); + + /* Program codec to use internal clock */ + nau8824_FLL_freerun_mode(codec, true); + return ret; +} + +static int nau8824_codec_remove(struct snd_soc_codec *codec) +{ + nau8824_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_nau8824 = { + .probe = nau8824_codec_probe, + .remove = nau8824_codec_remove, + .suspend = nau8824_suspend, + .resume = nau8824_resume, + .set_bias_level = nau8824_set_bias_level, + .controls = nau8824_snd_controls, + .num_controls = ARRAY_SIZE(nau8824_snd_controls), + .dapm_widgets = nau8824_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8824_dapm_widgets), + .dapm_routes = nau8824_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8824_dapm_routes), + .reg_word_size = sizeof(u16), + .reg_cache_default = init_list, + .set_sysclk = nau8824_set_sysclk, +}; + +static struct snd_soc_dai_ops nau8824_dai_ops = { + .hw_params = nau8824_hw_params, + .set_pll = nau8824_set_dai_pll, + .set_fmt = nau8824_set_dai_fmt, + .digital_mute = nau8824_dac_mute, +}; + +static struct snd_soc_dai_driver nau8824_dai_driver[] = { +{ + .name = "nau8824-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8824_RATES, + .formats = NAU8824_FORMATS, + }, + .ops = &nau8824_dai_ops, +} +}; + +static int nau8824_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct nau8824_priv *nau8824; + int ret; + + pr_debug("%s enter", __func__); + nau8824 = kzalloc(sizeof(*nau8824), GFP_KERNEL); + if (nau8824 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8824); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8824, + nau8824_dai_driver, + ARRAY_SIZE(nau8824_dai_driver)); + if (ret < 0) + kfree(nau8824); + + return ret; +} + +static int nau8824_i2c_remove(struct i2c_client *i2c) +{ + struct nau8824_priv *nau8824 = dev_get_drvdata(&i2c->dev); + + snd_soc_unregister_codec(nau8824->dev); + + return 0; +} + +static const struct i2c_device_id nau8824_i2c_id[] = { + { "nau8824", 0}, + {"10508824:00", 0}, + {"10508824", 0}, + { } +}; + +MODULE_DEVICE_TABLE(i2c, nau8824_i2c_id); +static struct i2c_driver nau8824_i2c_driver = { + .driver = { + .name = "nau8824", + .owner = THIS_MODULE, + }, + .probe = nau8824_i2c_probe, + .remove = (nau8824_i2c_remove), + .id_table = nau8824_i2c_id, +}; +module_i2c_driver(nau8824_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8824 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h new file mode 100644 index 0000000..1745dac --- /dev/null +++ b/sound/soc/codecs/nau8824.h @@ -0,0 +1,379 @@ +/* + * linux/sound/soc/codecs/nau8824.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8824_H +#define _NAU8824_H + +#define NAU8824_RESET 0x00 +#define NAU8824_ENA_CTRL 0x01 +#define NAU8824_CLK_EN 0x02 +#define NAU8824_CLK_DIVIDER 0x03 +#define NAU8824_FL_1 0x04 +#define NAU8824_FL_2 0x05 +#define NAU8824_FL_3 0x06 +#define NAU8824_FL_4 0x07 +#define NAU8824_FL_5 0x08 +#define NAU8824_FL_6 0x09 +#define NAU8824_IRQ_STATUS 0x10 +#define NAU8824_IRQ_CLEAR 0x11 +#define NAU8824_IRQ_SETTING 0x12 +#define NAU8824_SAR_ADC 0x13 +#define NAU8824_VDET_COEFFICIENT 0x14 +#define NAU8824_I2S_PCM_CTRL_1 0x1C +#define NAU8824_I2S_PCM_CTRL_2 0x1D +#define NAU8824_ADC_CH0_DGAIN_CTRL 0x2D +#define NAU8824_ADC_CH1_DGAIN_CTRL 0x2E +#define NAU8824_DAC_MUTE_CTRL 0x31 +#define NAU8824_DAC0_DIG_VOL 0x32 +#define NAU8824_DAC1_DIG_VOL 0x33 +#define NAU8824_CLASS_G_CTRL 0x50 +#define NAU8824_SAR_ADC_OUTPUT 0x59 +#define NAU8824_BIAS_ADJ 0x66 +#define NAU8824_ANALOG_CTRL2 0x6A +#define NAU8824_HPO_MIXER 0x6B +#define NAU8824_CLASS_D_GAIN1 0x6D +#define NAU8824_CLASS_D_GAIN2 0x6E +#define NAU8824_DAC_CTRL 0x73 +#define NAU8824_MIC_BIAS 0x74 +#define NAU8824_HP_MUTE 0x75 +#define NAU8824_BOOST 0x76 +#define NAU8824_HP_VOL 0x7B + +/* reg. NAU8824_ENA_CTRL (0x01) */ +#define NAU8824_DMIC_EDGE_CH23_MASK (0x01 << 15) +#define NAU8824_DMIC_EDGE_CH23_SFT 15 +#define NAU8824_DMIC_EDGE_CH01_MASK (0x01 << 14) +#define NAU8824_DMIC_EDGE_CH01_SFT 14 +#define NAU8824_ADC_OP_EN_MASK (0x01 << 11) +#define NAU8824_ADC_OP_EN_SFT 11 +#define NAU8824_ADC_OP_EN (0x01 << 11) +#define NAU8824_ADC_OP_DIS (0x00 << 11) +#define NAU8824_INT_SLEEP_MASK (0x01 << 10) +#define NAU8824_INT_SLEEP_SFT 10 +#define NAU8824_INT_SLEEP_ON (0x01 << 10) +#define NAU8824_INT_SLEEP_OFF (0x00 << 10) +#define NAU8824_DMIC_CH3_MASK (0x01 << 9) +#define NAU8824_DMIC_CH3_SFT 9 +#define NAU8824_DMIC_CH3_ON (0x01 << 9) +#define NAU8824_DMIC_CH3_OFF (0x00 << 9) +#define NAU8824_DMIC_CH2_MASK (0x01 << 8) +#define NAU8824_DMIC_CH2_SFT 8 +#define NAU8824_DMIC_CH2_ON (0x01 << 8) +#define NAU8824_DMIC_CH2_OFF (0x00 << 8) +#define NAU8824_DMIC_CH1_MASK (0x01 << 7) +#define NAU8824_DMIC_CH1_SFT 7 +#define NAU8824_DMIC_CH1_ON (0x01 << 7) +#define NAU8824_DMIC_CH1_OFF (0x00 << 7) +#define NAU8824_DMIC_CH0_MASK (0x01 << 6) +#define NAU8824_DMIC_CH0_SFT 6 +#define NAU8824_DMIC_CH0_ON (0x01 << 6) +#define NAU8824_DMIC_CH0_OFF (0x00 << 6) +#define NAU8824_DAC_CH1_EN_MASK (0x01 << 5) +#define NAU8824_DAC_CH1_EN_SFT 5 +#define NAU8824_DAC_CH1_EN (0x01 << 5) +#define NAU8824_DAC_CH1_DIS (0x00 << 5) +#define NAU8824_DAC_CH0_EN_MASK (0x01 << 4) +#define NAU8824_DAC_CH0_EN_SFT 4 +#define NAU8824_DAC_CH0_EN (0x01 << 4) +#define NAU8824_DAC_CH0_DIS (0x00 << 4) +#define NAU8824_ADC_CH3_EN_MASK (0x01 << 3) +#define NAU8824_ADC_CH3_EN_SFT 3 +#define NAU8824_ADC_CH3_EN (0x01 << 3) +#define NAU8824_ADC_CH3_DIS (0x00 << 3) +#define NAU8824_ADC_CH2_EN_MASK (0x01 << 2) +#define NAU8824_ADC_CH2_EN_SFT 2 +#define NAU8824_ADC_CH2_EN (0x01 << 2) +#define NAU8824_ADC_CH2_DIS (0x00 << 2) +#define NAU8824_ADC_CH1_EN_MASK (0x01 << 1) +#define NAU8824_ADC_CH1_EN_SFT 1 +#define NAU8824_ADC_CH1_EN (0x01 << 1) +#define NAU8824_ADC_CH1_DIS (0x00 << 1) +#define NAU8824_ADC_CH0_EN_MASK (0x01 << 0) +#define NAU8824_ADC_CH0_EN_SFT 0 +#define NAU8824_ADC_CH0_EN (0x01 << 0) +#define NAU8824_ADC_CH0_DIS (0x00 << 0) + +/* reg. NAU8824_CLK_EN (0x02) */ +#define NAU8824_CLK_SAR_EN_MASK (0x1 << 3) +#define NAU8824_CLK_SAR_EN_SFT 3 +#define NAU8824_CLK_SAR_EN (0x1 << 3) +#define NAU8824_CLK_SAR_DIS (0x0 << 3) + +/* reg. NAU8824_CLK_DIVIDER (0x03) */ +#define NAU8824_SYSCLK_SRC_MASK (0x1 << 15) +#define NAU8824_SYSCLK_SRC_SFT 15 +#define NAU8824_SYSCLK_SRC_VCO (0x1 << 15) +#define NAU8824_SYSCLK_SRC_MCLK (0x0 << 15) +#define NAU8824_CLK_CODEC_SRC_MASK (0x1 << 13) +#define NAU8824_CLK_CODEC_SRC_SFT 13 +#define NAU8824_CODEC_SRC_SYSCLK (0x1 << 13) +#define NAU8824_CODEC_SRC_MCLK (0x0 << 13) +#define NAU8824_CLK_DMIC_SRC_MASK (0x7 << 10) +#define NAU8824_CLK_DMIC_SRC_SFT 10 +#define NAU8824_CLK_DMIC_DIV_4 (0x2 << 10) + +/* reg. NAU8824_IRQ_STATUS (0x10) */ +#define NAU8824_IRQ_KEY_RELEASE (0x1 << 5) +#define NAU8824_IRQ_LONGKEY (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED (0x1 << 0) + +/* reg. NAU8824_IRQ_CLEAR (0x11) */ +#define NAU8824_IRQ_KEY_RELEASE_CLR (0x1 << 5) +#define NAU8824_IRQ_LONGKEY_CLR (0x1 << 4) +#define NAU8824_IRQ_SHORTKEY_CLR (0x1 << 3) +#define NAU8824_IRQ_MIC_INSERTED_CLR (0x1 << 2) +#define NAU8824_IRQ_JACK_EJECTED_CLR (0x1 << 1) +#define NAU8824_IRQ_JACK_INSERTED_CLR (0x1 << 0) + +/* reg NAU8824_SAR_ADC (0x13) */ +#define NAU8824_SAR_EN_MASK (0x1 << 12) +#define NAU8824_SAR_EN_SFT 12 +#define NAU8824_SAR_EN (0x1 << 12) +#define NAU8824_SAR_DIS (0x0 << 12) + +/* reg. NAU8824_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8824_I2S_BP_MASK (0x1 << 7) +#define NAU8824_I2S_BP_SFT 7 +#define NAU8824_I2S_BP_NOR (0x0 << 7) +#define NAU8824_I2S_BP_INV (0x1 << 7) +#define NAU8824_I2S_PCMB_MASK (0x1 << 6) +#define NAU8824_I2S_PCMB_SFT 6 +#define NAU8824_I2S_PCMB_EN (0x1 << 6) +#define NAU8824_I2S_PCMB_DIS (0x0 << 6) +#define NAU8824_I2S_DL_MASK (0x3 << 2) +#define NAU8824_I2S_DL_SFT 2 +#define NAU8824_I2S_DF_MASK (0x3) +#define NAU8824_I2S_DF_SFT 0 +#define NAU8824_I2S_DF_RIGHT (0x0) +#define NAU8824_I2S_DF_LEFT (0x1) +#define NAU8824_I2S_DF_I2S (0x2) +#define NAU8824_I2S_DF_PCM_A (0x3) +#define NAU8824_I2S_DF_PCM_B (0x3) + +/* reg. NAU8824_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8824_I2S_BCLKDIV_MASK 7 +#define NAU8824_I2S_BCLKDIV_SFT 0 + +#define NAU8824_I2S_MS_MASK (0x1 << 3) +#define NAU8824_I2S_MS_SFT 3 +#define NAU8824_I2S_MS_M (0x1 << 3) +#define NAU8824_I2S_MS_S (0x0 << 3) + +/* reg. NAU8824_DAC_MUTE_CTRL (0x31) */ +#define NAU8824_SOFT_MUTE_MASK (0x1 << 13) +#define NAU8824_SOFT_MUTE_SFT 13 +#define NAU8824_SOFT_MUTE_EN (0x1 << 13) +#define NAU8824_SOFT_MUTE_DIS (0x0 << 13) + +/* reg NAU8824_DAC0_DIG_VOL (0x32) */ +#define NAU8824_DAC0_GAIN_MASK (0x1FF << 0) + +/* reg NAU8824_DAC1_DIG_VOL (0x33) */ +#define NAU8824_DAC1_GAIN_MASK (0x1FF << 0) + +/* reg. NAU8824_CLASS_G_CTRL (0x50) */ +#define NAU8824_CLASS_G_CLK_SRC_MSK (0x3 << 14) +#define NAU8824_CLASS_G_CLK_DIS (0x3 << 14) +#define NAU8824_CLASS_G_CLK_SRC_MCLK (0x2 << 14) +#define NAU8824_CLASS_G_CLK_SRC_660K (0x1 << 14) +#define NAU8824_CLASS_G_CLK_SRC_2M (0x0 << 14) +#define NAU8824_CLASS_G_TIMER_MSK (0x3F << 8) +#define NAU8824_CLASS_G_TIMER_SFT 8 +#define NAU8824_CLASS_G_THRSLD_MSK (0x3 << 4) +#define NAU8824_CLASS_G_THRSLD_SFT 4 +#define NAU8824_CLASS_G_RIGHT_EN_MSK (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_EN (0x1 << 2) +#define NAU8824_CLASS_G_RIGHT_DIS (0x0 << 2) +#define NAU8824_CLASS_G_LEFT_EN_MSK (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_EN (0x1 << 1) +#define NAU8824_CLASS_G_LEFT_DIS (0x0 << 1) +#define NAU8824_CLASS_G_EN_MSK (0x1 << 0) +#define NAU8824_CLASS_G_SHIFT 0 +#define NAU8824_CLASS_G_EN (0x1 << 0) +#define NAU8824_CLASS_G_DIS (0x0 << 0) + +/* reg NAU8824_SAR_ADC_OUTPUT (0x59) */ +#define NAU8824_ADC_OUTPUT_MASK (0xFF << 0) +#define NAU8824_ADC_OUTPUT_SFT 0 + +/* reg. NAU8824_BIAS_ADJ_2 (0x66) */ +#define NAU8824_VMID_MASK (0x1 << 6) +#define NAU8824_VMID_SFT 6 +#define NAU8824_VMID_EN (0x1 << 6) +#define NAU8824_VMID_DIS (0x0 << 6) +#define NAU8824_VMID_OPTION_MASK (0x1 << 4) +#define NAU8824_VMID_OPTION_SFT 4 +#define NAU8824_DMIC2_MASK (0x1 << 3) +#define NAU8824_DMIC2_SFT 3 +#define NAU8824_DMIC2_EN (0x1 << 3) +#define NAU8824_DMIC2_DIS (0x0 << 3) +#define NAU8824_DMIC1_MASK (0x1 << 2) +#define NAU8824_DMIC1_SFT 2 +#define NAU8824_DMIC1_EN (0x1 << 2) +#define NAU8824_DMIC1_DIS (0x0 << 2) + +/* reg. NAU8824_ANALOG_CTRL2 (0x6A) */ +#define NAU8824_CLASS_D_CLAMP_MSK (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_SFT 3 +#define NAU8824_CLASS_D_CLAMP_BETTER (0x1 << 3) +#define NAU8824_CLASS_D_CLAMP_DEFAULT (0x0 << 3) + +/* reg. NAU8824_HPO_MIXER (0x6B) */ +#define NAU8824_M_HPVOL_R_HM (0x1 << 2) +#define NAU8824_M_HPVOL_R_HM_SFT 2 +#define NAU8824_M_HPVOL_L_HM (0x1 << 0) +#define NAU8824_M_HPVOL_L_HM_SFT 0 + +/* reg. NAU8824_CLASS_D_GAIN1 (0x6D) */ +#define NAU8824_CLASS_D_GAIN_R_MSK (0x1F << 8) +#define NAU8824_CLASS_D_GAIN_R_SFT 8 +#define NAU8824_CLASS_D_MASK (0x1 << 7) +#define NAU8824_CLASS_D_SFT 7 +#define NAU8824_CLASS_D_EN_MSK (0x1 << 7) +#define NAU8824_CLASS_D_EN (0x1 << 7) +#define NAU8824_CLASS_D_DIS (0x0 << 7) + +/* reg. NAU8824_CLASS_D_GAIN2 (0x6E) */ +#define NAU8824_CLASS_D_GAIN_L_MSK (0x1F << 0) +#define NAU8824_CLASS_D_GAIN_L_SFT 0 + +/* reg. NAU8824_DAC_CTRL (0x73) */ +#define NAU8824_DAC_R_MASK (0x1 << 13) +#define NAU8824_DAC_R_SFT 13 +#define NAU8824_DAC_R_EN (0x1 << 13) +#define NAU8824_DAC_R_DIS (0x0 << 13) +#define NAU8824_DAC_L_MASK (0x1 << 12) +#define NAU8824_DAC_L_SFT 12 +#define NAU8824_DAC_L_EN (0x1 << 12) +#define NAU8824_DAC_L_DIS (0x0 << 12) + +/* reg. NAU8824_MIC_BIAS (0x74) */ +#define NAU8824_BIAS_POWER_MASK (0x1 << 8) +#define NAU8824_BIAS_POWER_SFT 8 +#define NAU8824_BIAS_POWER_ON (0x1 << 8) +#define NAU8824_BIAS_POWER_DOWN (0x0 << 8) +#define NAU8824_BIAS_LEVEL_MASK (0x7 << 0) +#define NAU8824_BIAS_LEVEL_SFT 0 +#define NAU8824_BIAS_LEVEL_VDDA (0x0 << 0) + +/* reg. NAU8824_HP_MUTE (0x75) */ +#define NAU8824_R_MUTE (0x1 << 14) +#define NAU8824_R_MUTE_SFT 14 +#define NAU8824_L_MUTE (0x1 << 6) +#define NAU8824_L_MUTE_SFT 6 + +/* reg. NAU8824_BOOST (0x76) */ +#define NAU8824_VMID_PRECHARGE_MASK (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_SFT 13 +#define NAU8824_VMID_PRECHARGE_DIS (0x1 << 13) +#define NAU8824_VMID_PRECHARGE_EN (0x0 << 13) +#define NAU8824_G_BIAS_MASK (0x1 << 12) +#define NAU8824_G_BIAS_SFT 12 +#define NAU8824_G_BIAS_EN (0x1 << 12) +#define NAU8824_G_BIAS_DIS (0x0 << 12) + +/* reg. NAU8824_HP_VOL (0x7B) */ +#define NAU8824_R_VOL_MASK (0x1f << 8) +#define NAU8824_R_VOL_SFT 8 +#define NAU8824_SPK_L_VOL_MASK (0x1f) +#define NAU8824_SPK_R_VOL_MASK (0x1f << 8) +#define NAU8824_L_VOL_MASK (0x1f) +#define NAU8824_L_VOL_SFT 0 + +/* Volume Rescale */ +#define NAU8824_VOL_RSCL_MAX 0x1E +#define NAU8824_VOL_RSCL_RANGE 0x1E +#define NAU8824_SPK_VOL_RSCL_MAX 0x19 +#define NAU8824_SPK_VOL_RSCL_RANGE 0x19 +#define NAU8824_DAC_VOL_RSCL_RANGE 0x164 +#define NAU8824_ADC_VOL_RSCL_RANGE 0x164 + +/* Data Format */ +#define NAU8824_I2S_DL_32 (0x3 << 2) +#define NAU8824_I2S_DL_24 (0x2 << 2) +#define NAU8824_I2S_DL_20 (0x1 << 2) +#define NAU8824_I2S_DL_16 (0x0 << 2) + +#define NAU8824_FREQ_25000000 25000000 +#define NAU8824_PLL_CLKIN_MCLK 0 +#define NAU8824_BUTTONPRESS_MASK 0x20 +#define NAU8824_HSPLUG_MASK 0x10 + +enum { + NAU8824_INTERNALCLOCK = 0, + NAU8824_MCLK, +}; + +enum { + NAU8824_J_IN_EVENT, + NAU8824_J_OUT_EVENT, + NAU8824_BP_EVENT, + NAU8824_BR_EVENT, + NAU8824_UN_EVENT, +}; + +enum { + S_NAU8824_J_OUT, + S_NAU8824_J_IN_TEMP, + S_NAU8824_J_IN, + S_NAU8824_BUTTON_DECT, + S_NAU8824_NO_BUTTON_DECT, +}; + +#define NAU8824_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8824_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +struct nau8824_jack_data { + struct snd_soc_jack *jack; + int report; +}; + +struct nau8824_gpio_setup { + unsigned int reg; + u8 value; +}; + +struct nau8824_pdata { + unsigned int audio_mclk1; + unsigned int gpio_irq; + int naudint_irq; + int headset_detect; + int button_press_detect; +}; + +struct nau8824_priv { + struct nau8824_jack_data hs_jack; + struct workqueue_struct *workqueue; + struct delayed_work delayed_work; + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct mutex mutex; + struct nau8824_pdata pdata; + unsigned int irq; + bool jd_status; + bool bp_status; + int jack_type; +}; + +int nau8824_device_init(struct nau8824_priv *nau8824); +void nau8824_device_exit(struct nau8824_priv *nau8824); +void nau8824_enable_mic_bias(struct snd_soc_codec *codec, int enable); +int nau8824_query_jack_status(struct snd_soc_codec *codec); +int nau8824_query_btn_press(struct snd_soc_codec *codec); +void nau8824_btn_press_intr_enable(struct snd_soc_codec *codec, int enable); + +#endif /* _NAU8824_H */ -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: atmel_pcm: use endian agnostic IO accessors
by Ben Dooks 26 Mar '15

26 Mar '15

The atmel-pcm.h is using __raw_readl and __raw_writel for IO accessors which do not work if the processor is in big endian mode (the peripheral will still expect data in little endian format). Note, I do not think this is being used in the AVR32 which would require a different fix (possible after atmel_io.h is merged). This has not been run-tested as I do not have a board with an audio code on. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> -- CC: Bo Shen <voice.shen(a)atmel.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Mark Brown <broonie(a)kernel.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: alsa-devel(a)alsa-project.org --- sound/soc/atmel/atmel-pcm.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 12ae814..8f5f551 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -80,8 +80,8 @@ struct atmel_pcm_dma_params { /* * SSC register access (since ssc_writel() / ssc_readl() require literal name) */ -#define ssc_readx(base, reg) (__raw_readl((base) + (reg))) -#define ssc_writex(base, reg, value) __raw_writel((value), (base) + (reg)) +#define ssc_readx(base, reg) (readl_relaxed((base) + (reg))) +#define ssc_writex(base, reg, value) writel_relaxed((value), (base) + (reg)) int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd); void atmel_pcm_free(struct snd_pcm *pcm); -- 2.1.4

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[alsa-devel] [PATCH] ASoC: davinci-evm: drop un-necessary remove function
by Jyri Sarha 26 Mar '15

26 Mar '15

From: Manish Badarkhe <manishvb(a)ti.com> As davinci card gets registered using 'devm_' api there is no need to unregister the card in 'remove' function. Hence drop the 'remove' function. Signed-off-by: Manish Badarkhe <manishvb(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-evm.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b6bb594..8c2b9be 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -425,18 +425,8 @@ static int davinci_evm_probe(struct platform_device *pdev) return ret; } -static int davinci_evm_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver davinci_evm_driver = { .probe = davinci_evm_probe, - .remove = davinci_evm_remove, .driver = { .name = "davinci_evm", .pm = &snd_soc_pm_ops, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: snd_soc_tas5086: Reinit register values on probe
by Pascal Huerst 26 Mar '15

26 Mar '15

If the machine driver has been un/reloaded, the register values of the codec driver have to be reinitialized in order to run properly. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- sound/soc/codecs/tas5086.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..cd19171 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -851,10 +851,16 @@ static int tas5086_probe(struct snd_soc_codec *codec) } tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + ret = tas5086_init(codec->dev, priv); if (ret < 0) goto exit_disable_regulators; + ret = regcache_sync(priv->regmap); + if (ret < 0) + goto exit_disable_regulators; + /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) -- 1.9.3

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[alsa-devel] [PATCH] ALSA: hda - Add one more node in the EAPD supporting candidate list
by Hui Wang 26 Mar '15

26 Mar '15

We have a HP machine which use the codec node 0x17 connecting the internal speaker, and from the node capability, we saw the EAPD, if we don't set the EAPD on for this node, the internal speaker can't output any sound. Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1436745 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- picked some content from the alsa-info.txt of this machine: Codec: Realtek ALC221 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0221 Subsystem Id: 0x103c8105 Revision Id: 0x100103 No Modem Function Group found ... ... ... Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Speaker Phantom Jack", index=0, device=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00] Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 1 0x0f sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8775389..75880fe 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) { /* We currently only handle front, HP */ static hda_nid_t pins[] = { - 0x0f, 0x10, 0x14, 0x15, 0 + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 }; hda_nid_t *p; for (p = pins; *p; p++) -- 1.9.1

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[alsa-devel] Add AZX_DCAPS_I915_POWEREWELL on SKL is OK?
by Yang, Libin 26 Mar '15

26 Mar '15

Hi Takashi, Is it OK to add AZX_DCAPS_I915_POWEREWELL for SKL in hda_intel.c? SKL HDA controller is not in powerwell domain. But its HDMI codec is in the domain. My understanding is if the flag is added here, it means the controller is in the domain. But I can't find a good place to add a flag to indicate the HDMI codec need the power domain. Regards, Libin

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[alsa-devel] Fix a small typo in manpage cspctl.1 of alsa-tools/sb16_csp
by saulery＠legtux.org 26 Mar '15

26 Mar '15

Hello, I found an error in man pages cspctl.1. Could you please integrate this patch ? Regards,

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[alsa-devel] [PATCH] ALSA: hda_intel: apply the Seperate stream_tag for Sunrise Point
by libin.yang＠intel.com 26 Mar '15

26 Mar '15

From: Libin Yang <libin.yang(a)intel.com> The total stream number of Sunrise Point's input and output stream exceeds 15, which will cause some streams do not work because of the overflow on SDxCTL.STRM field if using the legacy stream tag allocation method. This patch uses the new stream tag allocation method by add the flag AZX_DCAPS_SEPARATE_STREAM_TAG for Skylake platform. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index feebc1d..15a8299 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1963,7 +1963,7 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, /* Sunrise Point */ { PCI_DEVICE(0x8086, 0xa170), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: rsnd: DPCM base sampling rate convert support
by Kuninori Morimoto 26 Mar '15

26 Mar '15

Hi Mark These are remains of my previous patch set. Basically, these doesn't need soc-core's new feature, but it is assuming DPCM base sampling rate convert has both playback/capture in same time. 1 - 2: new Renesas sound card for DPCM 3 : bug fix patch. it can be applied in linus/master branch 4 - 5: new feature, but based on 3) patch Kuninori Morimoto (5): 1) ASoC: rsrc-card: add Renesas sampling rate convert sound card support 2) ASoC: rsrc-card: add .be_hw_params_fixup support for convert rate 3) ASoC: rsnd: call clk_prepare_enable/unprepare() in probe/remove 4) ASoC: rsnd: remove useless debug message 5) ASoC: rsnd: add DPCM based sampling rate convert Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 67 +++++++++++++ sound/soc/sh/Kconfig | 5 + sound/soc/sh/rcar/Makefile | 5 +- sound/soc/sh/rcar/adg.c | 2 - sound/soc/sh/rcar/core.c | 25 ++++- sound/soc/sh/rcar/dma.c | 4 - sound/soc/sh/rcar/dvc.c | 28 +----- sound/soc/sh/rcar/gen.c | 4 - sound/soc/sh/rcar/rsnd.h | 12 ++- sound/soc/sh/rcar/rsrc-card.c | 512 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/rcar/src.c | 84 +++++++++------- sound/soc/sh/rcar/ssi.c | 31 ++---- 12 files changed, 683 insertions(+), 96 deletions(-)

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[alsa-devel] RawMIDI behaviour with MidiFace 4x4
by Kalvas, Taneli 25 Mar '15

25 Mar '15

Dear All, I am developing a patch librarian for a synthesizer. Because the communication happens with Midi sysex and timing is not relevant I chose RawMIDI interface and not the alsa sequencer. The problem I have encountered is that writing to the midi port with snd_rawmidi_write() seems to broken. The function accepts any size of data block and claims to have sent everything. In reality only a part of the message is sent. For a large block size I can see the activity light blinking for fraction of a second. This happens both in blocking and non-blocking mode. I would expect the snd_rawmidi_write() to accept only a part of the message. My hardware is Miditech Midiface 4x4, a USB midi interface: $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xe0610000 irq 45 1 [MIDI4x4 ]: USB-Audio - MIDI4x4 MIDIPLUS MIDI4x4 at usb-0000:00:14.0-1, full speed usbid: 1acc:1a0b My alsa is quite recent (aplay shows version 1.0.27.2), I am running Ubuntu with 3.13.0-39-generic kernel. I have tested the system by connecting the output port to an input port on the hardware and having another program output the received bytes on stdout. My observations: - As long as the buffer size is 276 bytes or less it gets sent ok - Larger buffer sizes cause corruption and roughly 300 bytes of received. - Alsa reports midi ring buffer size of 4096. - snd_rawmidi_status_get_xruns() on both ends reports 0 errors. - If I write N=128 byte blocks and usleep for 352*N after each write, everything works ok. Is this due to hardware incompatibility with alsa driver or what is going on? Is there a way to connect two programs via a virtual RawMIDI port so that I could test without using the hardware? Any other diagnostics that I could do? My test programs: write.cpp: int main( void ) { int intstatus; snd_rawmidi_t *midiout; if( (intstatus = snd_rawmidi_open(NULL, &midiout, "hw:1,0,1", 0)) ) { std::cerr << "unable to open MIDI out port: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } int N = 3000; // 276 bytes gets through ok. unsigned char msg[N]; for( int a = 0; a < N; a++ ) { msg[a] = a % 0x80; } msg[0] = 0xF0; msg[N-1] = 0xF7; ssize_t wrstatus; if( (wrstatus = snd_rawmidi_write(midiout, msg, N)) < 0 ) { std::cerr << "unable to write MIDI port: " << snd_strerror(wrstatus) << "\n"; return( -1 ); } std::cout << "write status = " << wrstatus << "\n"; if( (intstatus = snd_rawmidi_drain( midiout )) ) { std::cerr << "unable to drain: " << snd_strerror(intstatus) << "\n"; exit( 1 ); } // Sleep for the amount of time needed for transfer + 10 % extra + 100 ms usleep( 352*N + 100000 ); snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiout, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "xruns = " << err << "\n"; snd_rawmidi_close( midiout ); return( 0 ); } -------------------- read.cpp: bool quit = false; void signal_int( int ) { quit = true; } int main( void ) { int mode = SND_RAWMIDI_NONBLOCK; int intstat; std::string portname = "hw:1,0,0"; //std::string portname = "virtual"; snd_rawmidi_t* midiin; if( (intstat = snd_rawmidi_open(&midiin, NULL, portname.c_str(), mode)) < 0 ) { std::cerr << "unable to open MIDI port " << portname << ": " << snd_strerror(intstat) << "\n"; exit( 1 ); } int count = 0; unsigned char buf[1]; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " "; signal( SIGINT, signal_int ); while( !quit ) { int read = snd_rawmidi_read( midiin, buf, 1 ); if( read == 0 || read == -EBUSY || read == -EAGAIN ) { usleep( 320 ); continue; } else if( read < 0 ) { std::cerr << "unable to read MIDI port" << portname << ": " << snd_strerror(read) << "\n"; exit( 1 ); } std::cout << std::hex << std::setfill('0') << "0x" << std::setw(2) << (int)buf[0] << " " << std::flush; count++; if( count % 8 == 0 ) { std::cout << "\n"; std::cout << std::dec << std::setfill(' ') << std::setw(6) << count << " " << std::flush; } } snd_rawmidi_status_t *stat; snd_rawmidi_status_malloc( &stat ); snd_rawmidi_status( midiin, stat ); int err = snd_rawmidi_status_get_xruns( stat ); std::cout << "\nxruns = " << err << "\n"; snd_rawmidi_close( midiin ); return( 0 ); } Thanks. Taneli -- Taneli Kalvas Ph.D., Researcher Department of Physics, room FL114 P.O. Box 35 (YFL) 40014 University of Jyväskylä, Finland Mobile: +358-44-314-1602 Fax: +358-14-617-411 Email: taneli.kalvas(a)jyu.fi

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