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

[alsa-devel] [PATCH] ASoC: max98095: Add an explicit of_match_table
by Tushar Behera 23 Apr '14

23 Apr '14

Create an explicit of_match_table entry for MAX98095 codec. Also add a binding Documentation for this compatible string. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- .../devicetree/bindings/sound/max98095.txt | 16 ++++++++++++++++ sound/soc/codecs/max98095.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98095.txt diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt new file mode 100644 index 0000000..bacbeaa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98095.txt @@ -0,0 +1,16 @@ +MAX98095 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98095". + +- reg : The I2C address of the device. + +Example: + +max98095: codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; +}; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 03f0536..8e548af 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2399,10 +2399,17 @@ static const struct i2c_device_id max98095_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98095_i2c_id); +static const struct of_device_id max98095_of_match[] = { + { .compatible = "maxim,max98095", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98095_of_match); + static struct i2c_driver max98095_i2c_driver = { .driver = { .name = "max98095", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98095_of_match), }, .probe = max98095_i2c_probe, .remove = max98095_i2c_remove, -- 1.7.9.5

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[alsa-devel] How to show hda trace events in a self-built kernel?
by Lin, Mengdong 23 Apr '14

23 Apr '14

Hi, How to show hda trace events in a self-built kernel? With a self-built kernel under Ubuntu 13.04 or 13.10, I enabled hda trace events by #cd /sys/kernel/debug/tracing/ #echo 1 > events/hda/enable #echo 1 > events/had_intel/enable Then I check the codec proc entry and run "cat trace". But I got no output of the hda trace events. But I switch to the original kernel in Ubuntu 13.04 or 13.10 and repeat the above steps, I can get the output of hda trace events. Thanks Mengdong

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Add sound card driver for Snow board
by Tushar Behera 23 Apr '14

23 Apr '14

Added machine driver to instantiate I2S based sound card on Snow board. It has MAX98095 audio codec on board. There are some other variants for Snow board which have MAX98090 audio codec. Hence support for MAX98090 is also added to this driver. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Documentation/devicetree/bindings/sound/snow.txt | 17 ++ sound/soc/samsung/Kconfig | 11 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/snow.c | 190 ++++++++++++++++++++++ 4 files changed, 220 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/snow.txt create mode 100644 sound/soc/samsung/snow.c diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt new file mode 100644 index 0000000..6f3fd02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -0,0 +1,17 @@ +Audio Binding for Snow boards + +Required properties: +- compatible : Can be one of the following, + "google,snow-audio-max98090" or + "google,snow-audio-max98095" +- samsung,i2s-controller: The phandle of the Samsung I2S0 controller +- samsung,audio-codec: The phandle of the audio codec + +Example: + +sound { + compatible = "google,snow-audio-max98095"; + + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98095>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f2e2891..d07b04f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -231,3 +231,13 @@ config SND_SOC_LITTLEMILL select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 + +config SND_SOC_SNOW + tristate "Audio support for Google Snow boards" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SOC_MAX98095 + select SND_SAMSUNG_I2S + help + Say Y if you want to add audio support for various Snow + boards based on Exynos5 series of SoCs. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 86715d8..6d0212b 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -34,6 +34,7 @@ snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm8994-objs := smdk_wm8994.o +snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o @@ -58,6 +59,7 @@ obj-$(CONFIG_SND_SOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o +obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c new file mode 100644 index 0000000..e8eedd8 --- /dev/null +++ b/sound/soc/samsung/snow.c @@ -0,0 +1,190 @@ +/* + * ASoC machine driver for Snow boards + * + * 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. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <linux/of_device.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include "i2s.h" + +#define FIN_PLL_RATE 24000000 + +static int snow_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int bfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U24: + case SNDRV_PCM_FORMAT_S24: + bfs = 48; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + FIN_PLL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, SAMSUNG_I2S_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops snow_ops = { + .hw_params = snow_hw_params, +}; + +static int snow_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link snow_dai[] = { + { /* Playback DAI i/f */ + .name = "Playback", + .stream_name = "Playback", + .codec_dai_name = "HiFi", + .init = snow_init, + .ops = &snow_ops, + }, { /* Capture DAI i/f */ + .name = "Capture", + .stream_name = "Capture", + .codec_dai_name = "HiFi", + .ops = &snow_ops, + }, +}; + +static struct snd_soc_card snow_snd = { + .name = "Snow-I2S", + .dai_link = snow_dai, + .num_links = ARRAY_SIZE(snow_dai), +}; + +static int snow_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snow_snd; + struct device_node *i2s_node, *codec_node; + const char *name; + int i, ret; + + if (!pdev->dev.platform_data && !pdev->dev.of_node) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + + name = of_get_property(pdev->dev.of_node, "card-name", NULL); + if (name) + card->name = name; + + i2s_node = of_parse_phandle(pdev->dev.of_node, + "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Property 'i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + codec_node = of_parse_phandle(pdev->dev.of_node, + "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(snow_dai); i++) { + snow_dai[i].codec_of_node = codec_node; + snow_dai[i].cpu_of_node = i2s_node; + snow_dai[i].platform_of_node = i2s_node; + } + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + return ret; +} + +static int snow_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id snow_of_match[] = { + { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98095", }, + {}, +}; + +static struct platform_driver snow_driver = { + .driver = { + .name = "snow-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = snow_of_match, + }, + .probe = snow_driver_probe, + .remove = snow_driver_remove, +}; + +module_platform_driver(snow_driver); + +MODULE_DESCRIPTION("ALSA SoC Audio machine driver for Snow"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Export devm_snd_soc_register_platform()
by Lars-Peter Clausen 22 Apr '14

22 Apr '14

devm_snd_soc_register_platform() is used in drivers which can be build as modules, so it needs to be exported to avoid linkers errors like: ERROR: "devm_snd_soc_register_platform" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "devm_snd_soc_register_platform" [sound/soc/davinci/snd-soc-davinci.ko] undefined! Fixes: 8931bf620 ("ASoC: Add resource managed snd_soc_register_platform()") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-devres.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index e94aa02..057e5ef 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -85,6 +85,7 @@ int devm_snd_soc_register_platform(struct device *dev, return ret; } +EXPORT_SYMBOL_GPL(devm_snd_soc_register_platform); static void devm_card_release(struct device *dev, void *res) { -- 1.8.0

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[alsa-devel] [PATCH 0/3] ASoC: davinci: devm_snd_soc_register_platform and mcasp cleanup
by Peter Ujfalusi 22 Apr '14

22 Apr '14

Hi, Now that we have resource managed version of snd_soc_register_platform() convert the davinci ASoC code to use it. Clean the mcasp driver's init code while we are here so we can remove the platform driver's remove function since we now have managed resources only. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-pcm: Convert to use devm_snd_soc_register_platform() ASoC: davinci-mcasp: Convert to use devm_snd_soc_register_component() ASoC: davinci-mcasp: Optimize pm_runtime usage and clean up the init code sound/soc/davinci/davinci-i2s.c | 1 - sound/soc/davinci/davinci-mcasp.c | 69 ++++++++++----------------------------- sound/soc/davinci/davinci-pcm.c | 8 +---- sound/soc/davinci/davinci-pcm.h | 4 --- sound/soc/davinci/davinci-vcif.c | 1 - 5 files changed, 19 insertions(+), 64 deletions(-) -- 1.9.2

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[alsa-devel] [PATCH] ASoC: Intel: Fix audio crash due to negative address offset
by Jarkko Nikula 22 Apr '14

22 Apr '14

From: Wenkai Du <wenkai.du(a)intel.com> There were occasional ADSP crash during reboot testing: [ 11.883364] BUG: unable to handle kernel paging request at ffffc90121700000 [ 11.883380] IP: [<ffffffffc024d8bc>] sst_module_insert_fixed_block+0x24f/0x26d [snd_soc_sst_dsp] [ 11.883397] PGD 7800b067 PUD 0 [ 11.883405] Oops: 0002 [#1] SMP [ 11.886418] gsmi: Log Shutdown Reason 0x03 The virtual address, ffffc90121700000, was out of range. The virtual address is calculated by adding LPE base address with an offset: sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); The offset is calculated in sst_byt_parse_module, by subtraction of two virtual addresses dsp->addr.fw_ext and dsp->addr.lpe: block_data.offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); These virtual addresses are assigned by kernel from ioremap: sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size); In current driver code, offset is defined as unsigned int32: struct sst_module_data { ... u32 offset; /* offset in FW file */ }; Most of the time kernel assigned virtual addresses with addr.fw_ext greater than addr.lpe. But sometimes it was the other way round. Fix the problem by declaring offset as signed int32_t. Signed-off-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index fe8e81aad646..30ca14a6a835 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -136,7 +136,7 @@ struct sst_module_data { enum sst_data_type data_type; /* type of module data */ u32 size; /* size in bytes */ - u32 offset; /* offset in FW file */ + int32_t offset; /* offset in FW file */ u32 data_offset; /* offset in ADSP memory space */ void *data; /* module data */ }; -- 1.9.2

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Re: [alsa-devel] For firewire drivers, reduce period time down to 2msec?
by Clemens Ladisch 22 Apr '14

22 Apr '14

Takashi Sakamoto wrote: > Currently I'm working for next patchsets and I'll post them in Friday. I should be healthy again by then. > The patchset also includes my idea to apply common PCM hw constraints to > dice/speakers(oxfw)/fireworks/bebob. Currently the constraints includes: > 1.Maximum bits for PCM sample is limited by 24. > 2.Minimum time for period is currently 2msec. > 3.In blocking mode, frames per period/buffer is aligned to SYT_INTERVAL > > I think you'll agree with 1 and 3. Yes. > So I want to discuss the second one beforehand. > > Currently dice/speakers driver apply 5msec for period time. This is just because of the fixed INTERRUPT_INTERVAL, plus lots of safety margin to reduce the relative jitter of the period interrupt. (If the interrupt interval were dynamic, it should be possible to reduce periods to about 0.25 ms.) > I want to reduce this to 2msec. The reason of this '2msec' is: > - Currently firewire-lib process 16 packets in one time. > - This equals to 2msec (= INTERRUPT_INTERVAL / 8000 * 1000) > - During this 2msec, when pcm_period_pointer passes through period > boundary, snd_pcm_period_elapsed() is scheduled with tasklet. > - But tasklet can execute one time even if the function is scheduled > several times before executing it. > - The packetizing process is enough fast so the function can be > scheduled before executing. > - As a result, snd_pcm_period_elapsed() can be called one time every 2msec. > > Actually my devices can work fine with period == 2msec or less than > 5msec, without any disadvantage. ALSA actually uses a fixed period _size_. When the period length is set to exactly 2 ms, but the 16 packets between two FireWire interrupts contain one sample less than needed, the ALSA period interupt is delayed until the next FireWire interrupt. In that case, Jack with two periods is likely to underrun. The minimum time should be increased so that at least one FireWire interrupt is guaranteed to happen for each period, even when there are rounding errors and clock differences. Regards, Clemens

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[alsa-devel] Disabling DAPM from ASoC framework
by Santhosh N M 22 Apr '14

22 Apr '14

Hi, I'm a newbie to ALSA/ASoC framework. Lets say there is a requirement, where, the application (user space) takes the responsibility to switch ON/OFF the power components of the codec components (using ALSA Mixer Interface). In this case, is there a standard way to disable DAPM from ASoC? One way I see (and found to be working) is setting num_dapm_widgets=0 and num_dapm_routes=0 before invoking snd_soc_register_codec() Thanks in advance for your help. Best Regards, San

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[alsa-devel] [PATCH] ASoC: omap-hdmi: Remove excess curly bracket
by Peter Ujfalusi 22 Apr '14

22 Apr '14

Fix the error added by commit: ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 32614b49653c..537a1ec8ad61 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -330,7 +330,6 @@ static int omap_hdmi_probe(struct platform_device *pdev) return omap_pcm_platform_register(&pdev->dev); } -} static int omap_hdmi_remove(struct platform_device *pdev) { -- 1.9.2

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[alsa-devel] [PATCH v3 0/5] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 22 Apr '14

22 Apr '14

The only change in v3 is the renaming of the component IO mutex from 'mutex' to 'io_mutex'. v2 cover letter: This series contains the remaining patches from the previous series that have not been applied yet. The series has been rebased ontop of asoc/for-next and depends on topic/sta350, topic/core, topic/dapm, topic/cache and topic/component. Some minor updates were made to the patches and three new patches: * One new user of snd_kcontrol_chip() showed up, the patch converts it to snd_soc_kcontrol_codec() * Change return type of snd_soc_write() from unsigned int to int * Remove ASoC level tracing Original cover letter: This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. Lars-Peter Clausen (5): ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() ASoC: Remove ASoC level IO tracing include/sound/soc-dapm.h | 1 + include/sound/soc.h | 51 +++++-- include/trace/events/asoc.h | 92 ------------- sound/soc/omap/omap-hdmi.c | 1 - sound/soc/soc-core.c | 285 +++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 91 ++---------- sound/soc/soc-io.c | 327 ++++++++++++++++++++++++++------------------ 7 files changed, 447 insertions(+), 401 deletions(-) -- 1.8.0

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