April 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 00/14] ALSA: fireface: new driver for RME Fireface series
by Takashi Sakamoto 05 Apr '17

05 Apr '17

Hi, This patchset updates my previous RFCv3, and go for ALSA upstream. [alsa-devel] [RFC v3][PATCH 00/13] ALSA: fireface: new driver for RME Fireface series http://mailman.alsa-project.org/pipermail/alsa-devel/2017-February/117811.h… This patchset newly adds a driver into ALSA firewire stack, to support for Fireface 400 in RME FireWire series. Unfortunately, units on this series are completely against IEC 61883-1/6. This brings much commits and descriptions into this patchset. I'd ask you to review much amount of codes, as well as MOTU FireWire series, to finish my work for AMDTP variants[0]. As I noted to my former patchset, units of the series have unique feature; resampling between packet/data block processing layer and sample processing layer. This patchset relies on this layer, in short the driver doesn't handle sampling clock frequency, just handle sampling transmission frequency and expects units to generates audio from resampled data. Users should use proper application to configure sampling clock frequency when they want to match the frequency and sampling transmission frequency. Changes from RFCv3: * Add function pointer to AMDTP structure to reduce branching on packet processing. * Improve commit description. [0]: [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants Takashi Sakamoto (14): ALSA: fireface: add skeleton for RME Fireface series ALSA: fireface: postpone sound card registration ALSA: fireface: add model specific structure ALSA: fireface: add an abstraction layer for model-specific protocols ALSA: fireface: add transaction support ALSA: fireface: add support for MIDI functionality ALSA: fireface: add proc node to help debugging ALSA: firewire-lib: use the same prototype for functions to handle packet ALSA: firewire-lib: add no-header packet processing ALSA: fireface: add unique data processing layer ALSA: fireface: add stream management functionality ALSA: fireface: add support for PCM functionality ALSA: fireface: add hwdep interface ALSA: fireface: add support for Fireface 400 include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 2 +- sound/firewire/Kconfig | 8 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 107 ++++++-- sound/firewire/amdtp-stream.h | 6 + sound/firewire/fireface/Makefile | 3 + sound/firewire/fireface/amdtp-ff.c | 155 +++++++++++ sound/firewire/fireface/ff-hwdep.c | 191 +++++++++++++ sound/firewire/fireface/ff-midi.c | 131 +++++++++ sound/firewire/fireface/ff-pcm.c | 409 ++++++++++++++++++++++++++++ sound/firewire/fireface/ff-proc.c | 63 +++++ sound/firewire/fireface/ff-protocol-ff400.c | 371 +++++++++++++++++++++++++ sound/firewire/fireface/ff-stream.c | 282 +++++++++++++++++++ sound/firewire/fireface/ff-transaction.c | 295 ++++++++++++++++++++ sound/firewire/fireface/ff.c | 209 ++++++++++++++ sound/firewire/fireface/ff.h | 146 ++++++++++ 18 files changed, 2365 insertions(+), 23 deletions(-) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/amdtp-ff.c create mode 100644 sound/firewire/fireface/ff-hwdep.c create mode 100644 sound/firewire/fireface/ff-midi.c create mode 100644 sound/firewire/fireface/ff-pcm.c create mode 100644 sound/firewire/fireface/ff-proc.c create mode 100644 sound/firewire/fireface/ff-protocol-ff400.c create mode 100644 sound/firewire/fireface/ff-stream.c create mode 100644 sound/firewire/fireface/ff-transaction.c create mode 100644 sound/firewire/fireface/ff.c create mode 100644 sound/firewire/fireface/ff.h -- 2.9.3

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[alsa-devel] [PATCH v2 1/2] ASoC: fsl_ssi: Remove an outdated comment
by Fabio Estevam 05 Apr '17

05 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. Remove such outdated comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Only remove the outdated part of the comment sound/soc/fsl/fsl_ssi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..e0639e17 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -57,10 +57,8 @@ /** * FSLSSI_I2S_RATES: sample rates supported by the I2S * - * This driver currently only supports the SSI running in I2S slave mode, - * which means the codec determines the sample rate. Therefore, we tell - * ALSA that we support all rates and let the codec driver decide what rates - * are really supported. + * Tell ALSA that we support all rates and let the codec driver decide what + * rates are really supported. */ #define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS -- 2.7.4

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[alsa-devel] [PATCH] ASoC: twl6040: Add control for HS and HF mono to stereo selection
by Peter Ujfalusi 05 Apr '17

05 Apr '17

The new controls will give user the ability to route the left PDM channel data to the right headset/handsfree DAC. HS mono to stereo switch: PDM channel 1 (or mono) data to both HS DAC. HF mono to stereo switch: PDM channel 3 data to both HF DAC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 748036e851ea..2b6ad09e0886 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -606,6 +606,14 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { twl6040_headset_power_get_enum, twl6040_headset_power_put_enum), + /* Left HS PDM data routed to Right HSDAC */ + SOC_SINGLE("Headset Mono to Stereo Playback Switch", + TWL6040_REG_HSRCTL, 7, 1, 0), + + /* Left HF PDM data routed to Right HFDAC */ + SOC_SINGLE("Handsfree Mono to Stereo Playback Switch", + TWL6040_REG_HFRCTL, 5, 1, 0), + SOC_ENUM_EXT("PLL Selection", twl6040_power_mode_enum, twl6040_pll_get_enum, twl6040_pll_put_enum), }; -- 2.12.2

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[alsa-devel] [PATCH] ASoC: max9867: export OF device ID as module aliases
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

The I2C core always reports a MODALIAS of the form i2c:<foo> even if the device was registered via OF, this means that exporting the OF device ID table device aliases in the module is not needed. But in order to change how the core reports modaliases to user-space, it's better to export it. While there, move the MODULE_DEVICE_TABLE(i2c, max9867_i2c_id) just next to the I2C device table declaration, for consistency with other drivers. Before this patch: $ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias alias: i2c:max9867 After this patch: $ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias alias: i2c:max9867 alias: of:N*T*Cmaxim,max9867C* alias: of:N*T*Cmaxim,max9867 Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/max9867.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6cdf15ab46de..0247edc9c84e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -516,13 +516,13 @@ static const struct i2c_device_id max9867_i2c_id[] = { { "max9867", 0 }, { } }; +MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); static const struct of_device_id max9867_of_match[] = { { .compatible = "maxim,max9867", }, { } }; - -MODULE_DEVICE_TABLE(i2c, max9867_i2c_id); +MODULE_DEVICE_TABLE(of, max9867_of_match); static const struct dev_pm_ops max9867_pm_ops = { SET_SYSTEM_SLEEP_PM_OPS(max9867_suspend, max9867_resume) -- 2.9.3

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[alsa-devel] [PATCH] ASoC: cs53l30: Set .of_match_table to OF device ID table
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

The driver has an OF device ID table but the struct i2c_driver .of_match_table field is not set. Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index cb47fb595ff4..1e0d5973b758 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -1130,6 +1130,7 @@ MODULE_DEVICE_TABLE(i2c, cs53l30_id); static struct i2c_driver cs53l30_i2c_driver = { .driver = { .name = "cs53l30", + .of_match_table = cs53l30_of_match, .pm = &cs53l30_runtime_pm, }, .id_table = cs53l30_id, -- 2.9.3

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[alsa-devel] [PATCH 0/6] ASoC: Add OF device table to I2C drivers that are missing it
by Javier Martinez Canillas 05 Apr '17

05 Apr '17

Hello, This series add OF device ID tables to ASoC I2C drivers whose devices are either used in Device Tree source files or are listed in binding docs as a compatible string. That's done because the plan is to change the I2C core to report proper OF modaliases instead of always reporting a MODALIAS=i2c:<foo> regardless if a device was registered via DT or using the legacy platform data mechanism. So these patches will make sure that ASoC I2C drivers modules will continue to be autoloaded once the I2C core is changed to report proper OF modalias. Best regards, Javier Javier Martinez Canillas (6): ASoc: rt5645: Add OF device ID table ASoC: ssm4567: Add OF device ID table ASoC: sta529: Add OF device ID table ASoC: uda1380: Add OF device ID table ASoC: wm8978: Add OF device ID table ASoC: rt5677: Add OF device ID table sound/soc/codecs/rt5645.c | 10 ++++++++++ sound/soc/codecs/rt5677.c | 7 +++++++ sound/soc/codecs/ssm4567.c | 9 +++++++++ sound/soc/codecs/sta529.c | 7 +++++++ sound/soc/codecs/uda1380.c | 7 +++++++ sound/soc/codecs/wm8978.c | 7 +++++++ 6 files changed, 47 insertions(+) -- 2.9.3

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[alsa-devel] [PATCH 0/2] ASoC: codec: wm8960: Relax bit clock computation when using PLL
by Daniel Baluta 05 Apr '17

05 Apr '17

This is a follow up of commit 3c01b9ee2ab ("ASoC: codec: wm8960: Relax bit clock computation") where we relaxed bitclk when sysclk was derived from MCLK. Now, we do the same thing for sysclk derived using PLL. Daniel Baluta (2): ASoC: codec: wm9860: Refactor PLL out freq search ASoC: codec: wm8960: Relax bit clock computation when using PLL sound/soc/codecs/wm8960.c | 105 +++++++++++++++++++++++++++++++++------------- 1 file changed, 76 insertions(+), 29 deletions(-) -- 2.7.4

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[alsa-devel] [RFC][PATCH] ASoC: soc-core: verify Sound Card normality
by Kuninori Morimoto 05 Apr '17

05 Apr '17

Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform components. If system uses Kernel modules, we can disable these drivers by using rmmod command. In such case, we can't disable CPU/Codec/Platform driver without disabling Sound Card driver. But on the other hand, we can disable these drivers by using unbind command. In such case, we can disable these drivers randomly. In this case, we can create dirty Sound Card which is missing necessary components. (1) If user disabled Sound Card first, but did nothing to other drivers, user can't use Sound because Sound Card is no longer exists. (2) If user disabled CPU/Codec/Platform driver randomly, but did nothing to Sound Card, user still be able to use Sound Card, because dirty Sound Card still exists. In this case, Sound system will be crashed if user started sound playback/capture. But we can't block such random unbind now. To avoid Sound Card crash in (2) case, we need to unregister Sound Card whenever CPU/Codec/Platform component were unregistered. This patch solves this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 07e4eec..52760bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3236,6 +3236,11 @@ static void snd_soc_component_cleanup(struct snd_soc_component *component) static void snd_soc_component_del_unlocked(struct snd_soc_component *component) { + struct snd_soc_card *card = component->card; + + if (card) + snd_soc_unregister_card(card); + list_del(&component->list); } -- 1.9.1

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[alsa-devel] Create a general function for codec jack detection
by Bard Liao 05 Apr '17

05 Apr '17

Mark, There are many codecs with the capability of jack detection. Usually, we create a jack on machine driver and call an exported codec function, like rt5670_set_jack_detect, to use the jack and report the jack type. Could we create a general function for such exported functions? The benefit is we don't need to use different function names for each codecs. So we can have a general machine driver with jack detection function. Below is what I am thinking the patch will look like. diff --git a/include/sound/soc.h b/include/sound/soc.h index cdfb55f..e5a863a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -434,6 +434,8 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); @@ -913,6 +915,7 @@ struct snd_soc_codec_driver { int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); + int (*set_jack)(struct snd_soc_codec *codec, struct snd_soc_jack *jack); /* codec IO */ struct regmap *(*get_regmap)(struct device *); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fbaa1bb..ec49577 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,6 +22,23 @@ #include <trace/events/asoc.h> /** + * snd_soc_codec_set_jack - configure codec jack. + * @codec: CODEC + * @jack: structure to use for the jack + * + * Configures and enables jack detection function. + */ +int snd_soc_codec_set_jack(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + if (codec->driver->set_jack) + return codec->driver->set_jack(codec, jack); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_jack); + +/** * snd_soc_card_jack_new - Create a new jack * @card: ASoC card * @id: an identifying string for this jack -- Thanks. Bard Liao

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[alsa-devel] [PATCH 38/38] Annotate hardware config module parameters in sound/pci/
by David Howells 05 Apr '17

05 Apr '17

When the kernel is running in secure boot mode, we lock down the kernel to prevent userspace from modifying the running kernel image. Whilst this includes prohibiting access to things like /dev/mem, it must also prevent access by means of configuring driver modules in such a way as to cause a device to access or modify the kernel image. To this end, annotate module_param* statements that refer to hardware configuration and indicate for future reference what type of parameter they specify. The parameter parser in the core sees this information and can skip such parameters with an error message if the kernel is locked down. The module initialisation then runs as normal, but just sees whatever the default values for those parameters is. Note that we do still need to do the module initialisation because some drivers have viable defaults set in case parameters aren't specified and some drivers support automatic configuration (e.g. PNP or PCI) in addition to manually coded parameters. This patch annotates drivers in sound/pci/. Suggested-by: Alan Cox <gnomes(a)lxorguk.ukuu.org.uk> Signed-off-by: David Howells <dhowells(a)redhat.com> cc: Jaroslav Kysela <perex(a)perex.cz> cc: Takashi Iwai <tiwai(a)suse.com> cc: alsa-devel(a)alsa-project.org --- sound/pci/als4000.c | 2 +- sound/pci/cmipci.c | 6 +++--- sound/pci/ens1370.c | 2 +- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/ymfpci/ymfpci.c | 6 +++--- 7 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 92bc06d01288..7844a75d8ed9 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -102,7 +102,7 @@ MODULE_PARM_DESC(id, "ID string for ALS4000 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ALS4000 soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address for ALS4000 soundcard. (0 = disabled)"); #endif diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index aeedc270ed9b..430f064c64da 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -68,14 +68,14 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for C-Media PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable C-Media PCI soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port."); module_param_array(soft_ac3, bool, NULL, 0444); MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only)."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #endif diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 164adad91650..90376739c5e1 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -106,7 +106,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Ensoniq AudioPCI soundcard."); #ifdef SUPPORT_JOYSTICK #ifdef CHIP1371 -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); #else module_param_array(joystick, bool, NULL, 0444); diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 19c9df6b0f3d..f067c76d77f8 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -137,12 +137,12 @@ MODULE_PARM_DESC(id, "ID string for Riptide soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Riptide soundcard."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, int, NULL, 0444); +module_param_hw_array(joystick_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port # for Riptide soundcard."); #endif -module_param_array(mpu_port, int, NULL, 0444); +module_param_hw_array(mpu_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU401 port # for Riptide driver."); -module_param_array(opl3_port, int, NULL, 0444); +module_param_hw_array(opl3_port, int, ioport, NULL, 0444); MODULE_PARM_DESC(opl3_port, "OPL3 port # for Riptide driver."); /* diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index a6aa48c5b969..8e3d4ec39c35 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -66,7 +66,7 @@ module_param_array(reverb, bool, NULL, 0444); MODULE_PARM_DESC(reverb, "Enable reverb (SRAM is present) for S3 SonicVibes soundcard."); module_param_array(mge, bool, NULL, 0444); MODULE_PARM_DESC(mge, "MIC Gain Enable for S3 SonicVibes soundcard."); -module_param(dmaio, uint, 0444); +module_param_hw(dmaio, uint, ioport, 0444); MODULE_PARM_DESC(dmaio, "DDMA i/o base address for S3 SonicVibes soundcard."); /* diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 2d8c14e3f8d2..127834021175 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -92,7 +92,7 @@ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for VIA 82xx bridge."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for VIA 82xx bridge."); -module_param(mpu_port, long, 0444); +module_param_hw(mpu_port, long, ioport, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port. (VT82C686x only)"); #ifdef SUPPORT_JOYSTICK module_param(joystick, bool, 0444); diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 812e27a1bcbc..4faf3e1ed06a 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -55,12 +55,12 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for the Yamaha DS-1 PCI soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Yamaha DS-1 soundcard."); -module_param_array(mpu_port, long, NULL, 0444); +module_param_hw_array(mpu_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 Port."); -module_param_array(fm_port, long, NULL, 0444); +module_param_hw_array(fm_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM OPL-3 Port."); #ifdef SUPPORT_JOYSTICK -module_param_array(joystick_port, long, NULL, 0444); +module_param_hw_array(joystick_port, long, ioport, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address"); #endif module_param_array(rear_switch, bool, NULL, 0444);

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