May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] conf/ucm: Fix build error for UCM Dell WD15 Dock USB-audio
by Xiuli Pan 03 May '18

03 May '18

From: Pan Xiuli <xiuli.pan(a)linux.intel.com> Miss oneline in configure.ac in provious patch conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> --- configure.ac | 1 + 1 file changed, 1 insertion(+) diff --git a/configure.ac b/configure.ac index 3ee989e..693b5d5 100644 --- a/configure.ac +++ b/configure.ac @@ -725,6 +725,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ src/conf/ucm/DAISY-I2S/Makefile \ src/conf/ucm/DB410c/Makefile \ + src/conf/ucm/Dell-WD15-Dock/Makefile \ src/conf/ucm/GoogleNyan/Makefile \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt565: fix missing assignment to ret_val
by Colin King 03 May '18

03 May '18

From: Colin Ian King <colin.king(a)canonical.com> Currently, the check that ret_val is not -ENOENT is always true and the quirk bit BYT_RY5651_MCLK_EN is never being cleared because ret_val is always zero at this point from a previous assignment earlier on. I believe that ret_val should actually be assigned to the return from devm_clk_get() as this can return -ENOENT (from a deeper call to clk_get_sys) and that was the original intention to check this. Detected by CoverityScan, CID#1460228 ("Logically dead code") Fixes: 02c0a3b3047f ("ASoC: Intel: bytcr_rt5651: add MCLK, quirks and cleanups") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 3c7d93520c52..469e2695d121 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -856,9 +856,10 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); /* * Fall back to bit clock usage for -ENOENT (clock not * available likely due to missing dependencies), bail -- 2.17.0

3 2

[alsa-devel] [PATCH v2 1/4] ASoC: fsl_ssi: Switch to SPDX identifier
by Fabio Estevam 03 May '18

03 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Retain the copyright statements in fsl_ssi.h (Timur) sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 6 ++--- sound/soc/fsl/fsl_ssi_dbg.c | 18 ++++++-------- 3 files changed, 35 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1544166..0a64822 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1,34 +1,29 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * - * Some notes why imx-pcm-fiq is used instead of DMA on some boards: - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver +// +// Author: Timur Tabi <timur(a)freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// Some notes why imx-pcm-fiq is used instead of DMA on some boards: +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/init.h> #include <linux/io.h> diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 18f8dd5..0bdda60 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -1,12 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC * * Author: Timur Tabi <timur(a)freescale.com> * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. + * Copyright 2007-2008 Freescale Semiconductor, Inc. */ #ifndef _MPC8610_I2S_H diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e..1bacfa2 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -1,14 +1,10 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions - * - * Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix - * - * Splitted from fsl_ssi.c - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions +// +// Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix +// +// Split from fsl_ssi.c #include <linux/debugfs.h> #include <linux/device.h> -- 2.7.4

2 4

[alsa-devel] [PATCH 1/4] ASoC: fsl_ssi: Switch to SPDX identifier
by Fabio Estevam 03 May '18

03 May '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 57 +++++++++++++++++++++------------------------ sound/soc/fsl/fsl_ssi.h | 6 +---- sound/soc/fsl/fsl_ssi_dbg.c | 18 ++++++-------- 3 files changed, 34 insertions(+), 47 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1544166..0a64822 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1,34 +1,29 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver - * - * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2010 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * - * Some notes why imx-pcm-fiq is used instead of DMA on some boards: - * - * The i.MX SSI core has some nasty limitations in AC97 mode. While most - * sane processor vendors have a FIFO per AC97 slot, the i.MX has only - * one FIFO which combines all valid receive slots. We cannot even select - * which slots we want to receive. The WM9712 with which this driver - * was developed with always sends GPIO status data in slot 12 which - * we receive in our (PCM-) data stream. The only chance we have is to - * manually skip this data in the FIQ handler. With sampling rates different - * from 48000Hz not every frame has valid receive data, so the ratio - * between pcm data and GPIO status data changes. Our FIQ handler is not - * able to handle this, hence this driver only works with 48000Hz sampling - * rate. - * Reading and writing AC97 registers is another challenge. The core - * provides us status bits when the read register is updated with *another* - * value. When we read the same register two times (and the register still - * contains the same value) these status bits are not set. We work - * around this by not polling these bits but only wait a fixed delay. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) driver +// +// Author: Timur Tabi <timur(a)freescale.com> +// +// Copyright 2007-2010 Freescale Semiconductor, Inc. +// +// Some notes why imx-pcm-fiq is used instead of DMA on some boards: +// +// The i.MX SSI core has some nasty limitations in AC97 mode. While most +// sane processor vendors have a FIFO per AC97 slot, the i.MX has only +// one FIFO which combines all valid receive slots. We cannot even select +// which slots we want to receive. The WM9712 with which this driver +// was developed with always sends GPIO status data in slot 12 which +// we receive in our (PCM-) data stream. The only chance we have is to +// manually skip this data in the FIQ handler. With sampling rates different +// from 48000Hz not every frame has valid receive data, so the ratio +// between pcm data and GPIO status data changes. Our FIQ handler is not +// able to handle this, hence this driver only works with 48000Hz sampling +// rate. +// Reading and writing AC97 registers is another challenge. The core +// provides us status bits when the read register is updated with *another* +// value. When we read the same register two times (and the register still +// contains the same value) these status bits are not set. We work +// around this by not polling these bits but only wait a fixed delay. #include <linux/init.h> #include <linux/io.h> diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index 18f8dd5..146981f 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -1,12 +1,8 @@ +/* SPDX-License-Identifier: GPL-2.0 */ /* * fsl_ssi.h - ALSA SSI interface for the Freescale MPC8610 and i.MX SoC * * Author: Timur Tabi <timur(a)freescale.com> - * - * Copyright 2007-2008 Freescale Semiconductor, Inc. This file is licensed - * under the terms of the GNU General Public License version 2. This - * program is licensed "as is" without any warranty of any kind, whether - * express or implied. */ #ifndef _MPC8610_I2S_H diff --git a/sound/soc/fsl/fsl_ssi_dbg.c b/sound/soc/fsl/fsl_ssi_dbg.c index 7aac63e..ddf2f64 100644 --- a/sound/soc/fsl/fsl_ssi_dbg.c +++ b/sound/soc/fsl/fsl_ssi_dbg.c @@ -1,14 +1,10 @@ -/* - * Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions - * - * Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix - * - * Splitted from fsl_ssi.c - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale SSI ALSA SoC Digital Audio Interface (DAI) debugging functions +// +// Copyright 2014 Markus Pargmann <mpa(a)pengutronix.de>, Pengutronix +// +// Split from fsl_ssi.c #include <linux/debugfs.h> #include <linux/device.h> -- 2.7.4

3 9

[alsa-devel] [PATCH 0/6] usb-audio: quirks for Dell WD15 dock and some extensions
by Takashi Iwai 02 May '18

02 May '18

Hi, here is a patch set for adding quirks and some new extensions for Dell WD15 dock. The first patch is the resend of the previous post, and the rest are new. Basically it adds a new "Keep Interface" control for the card to skip resetting the USB interface per stream manipulation. It helps for reducing the long pause before each start of stream on Dell WD15 dock. OTOH, this was confirmed to be harmless on some dock early firmware version, so we can't set it statically in the driver. So, instead, the feature is exposed as a control element so that user can flip dynamically. This feature may be useful for other USB audio devices as well. Another extension is to allow the quirk entry to override the card longname field. This is needed to give a proper UCM profile. WD15 dock provides two streams, one for the headphone and another for the line out. It's a good example for using a UCM profile. Takashi === Takashi Iwai (6): ALSA: usb-audio: Initialize Dell Dock playback volumes ALSA: usb-audio: Avoid superfluous usb_set_interface() calls ALSA: usb-audio: Add keep_iface flag ALSA: usb-audio: Add "Keep Interface" control ALSA: usb-audio: Allow to override the longname string ALSA: usb-audio: Give proper vendor/product name for Dell WD15 Dock sound/usb/card.c | 146 ++++++++++++++++++++++++++++------------------- sound/usb/mixer.c | 47 +++++++++++++++ sound/usb/mixer_quirks.c | 34 +++++++++++ sound/usb/mixer_quirks.h | 4 ++ sound/usb/pcm.c | 28 ++++----- sound/usb/quirks-table.h | 10 ++++ sound/usb/usbaudio.h | 4 ++ 7 files changed, 200 insertions(+), 73 deletions(-) -- 2.16.3

1 6

[alsa-devel] [PATCH v3 00/13] ALSA: dice: use cache of stream formats to generate PCM rules/constraints
by Takashi Sakamoto 02 May '18

02 May '18

Hi, This patchset updates my previous one: [alsa-devel] [PATCH v2 00/11] ALSA: dice: use cache of stream formats to generate PCM rules/constraints http://mailman.alsa-project.org/pipermail/alsa-devel/2018-April/135330.html Once, ALSA dice driver cached stream formats to generate PCM constraints/rules. The cache is created by changing current sampling transmission frequency step by step directly at unit probing process. This feature was dropped because it changes state of clock on target device thus it's against users' expectation[1]. However this is not necessarily convenient because reconfiguration of sampling rate is not available. TC Applied Technologies (TCAT) has extended protocol for DICE interface, a.k.a EAP (Extended Application Protocol). In this extension, drivers can retrieve information of stream formats without changing state of target unit. Fortunately, this protocol is supported widely by many models. This patchset regain the cache to generate PCM constraints/rules by using this protocol. Userspace applications can reconfigure sampling rate via ALSA PCM interface. For units which doesn't support the extension, available set of sampling rate is still restricted at current sampling transmission frequency. This is also inconvenient and this patchset introduces a solution. Structure for unit entry (struct ieee1394_device_id) has a member for unit-specific data. This patchset uses it for callback function to handle hard-coded stream format. In this time, I adopt this way for some known models produced by TC Electronic and Alesis. If users find their models still have restrictions in a point of available sampling transmission frequency, it's preferable to work for this way. Changes from v2: - use robust representation to calcurate mode of sampling transmission frequency. [1] [alsa-devel] [PATCH 0/6 v2] ALSA: dice: constrain PCM substreams to current sampling transfer frequency http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104065.h… Takashi Sakamoto (13): ALSA: dice: add cache of stream formats ALSA: dice: add 'firewire' directory for proc nodes ALSA: dice: add proc node for stream formation ALSA: dice: cache stream formats at current mode of sampling transmission frequency ALSA: dice: add parameters of stream formats for models produced by TC Electronic ALSA: dice: add parameters of stream formats for models produced by Alesis ALSA: dice: use extended protocol to detect available stream formats ALSA: dice: use cache of stream format to check running stream ALSA: dice: add a helper function to restart all of available streams ALSA: dice: enable to change current sampling transmission frequency ALSA: dice: use stream formats to add MIDI substreams ALSA: dice: use cache for PCM constraints and rules ALSA: dice: remove local frag of force_two_pcms sound/firewire/dice/Makefile | 3 +- sound/firewire/dice/dice-alesis.c | 52 ++++++ sound/firewire/dice/dice-extension.c | 172 +++++++++++++++++++ sound/firewire/dice/dice-midi.c | 23 +-- sound/firewire/dice/dice-pcm.c | 233 ++++++++++++++++++-------- sound/firewire/dice/dice-proc.c | 70 +++++++- sound/firewire/dice/dice-stream.c | 283 ++++++++++++++++++++++++-------- sound/firewire/dice/dice-tcelectronic.c | 100 +++++++++++ sound/firewire/dice/dice.c | 139 ++++++++++------ sound/firewire/dice/dice.h | 24 ++- 10 files changed, 891 insertions(+), 208 deletions(-) create mode 100644 sound/firewire/dice/dice-alesis.c create mode 100644 sound/firewire/dice/dice-extension.c create mode 100644 sound/firewire/dice/dice-tcelectronic.c -- 2.14.1

2 14

[alsa-devel] [PATCH] ALSA: sc6000: fix spelling mistake: "iomaped" -> "iomapped"
by Colin King 02 May '18

02 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in KERN_ERR error messages Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/sc6000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index c09d9b914efe..a985e9183be9 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -592,7 +592,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) *vport = devm_ioport_map(devptr, port[dev], 0x10); if (*vport == NULL) { snd_printk(KERN_ERR PFX - "I/O port cannot be iomaped.\n"); + "I/O port cannot be iomapped.\n"); err = -EBUSY; goto err_unmap1; } @@ -607,7 +607,7 @@ static int snd_sc6000_probe(struct device *devptr, unsigned int dev) vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); if (!vmss_port) { snd_printk(KERN_ERR PFX - "MSS port I/O cannot be iomaped.\n"); + "MSS port I/O cannot be iomapped.\n"); err = -EBUSY; goto err_unmap2; } -- 2.17.0

2 1

[alsa-devel] [PATCH] ALSA: hda/ca0132: fix build failure when a local macro is defined
by Takashi Sakamoto 02 May '18

02 May '18

DECLARE_TLV_DB_SCALE (alias of SNDRV_CTL_TLVD_DECLARE_DB_SCALE) is used but tlv.h is not included. This causes build failure when local macro is defined by comment-out. This commit fixes the bug. At the same time, the alias macro is replaced with a destination macro added at a commit 46e860f76804 ("ALSA: rename TLV-related macros so that they're friendly to user applications") Reported-by: Connor McAdams <conmanx360(a)gmail.com> Fixes: 44f0c9782cc6 ('ALSA: hda/ca0132: Add tuning controls') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/pci/hda/patch_ca0132.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 768ea8651993..84261ef02c93 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -39,6 +39,10 @@ /* Enable this to see controls for tuning purpose. */ /*#define ENABLE_TUNING_CONTROLS*/ +#ifdef ENABLE_TUNING_CONTROLS +#include <sound/tlv.h> +#endif + #define FLOAT_ZERO 0x00000000 #define FLOAT_ONE 0x3f800000 #define FLOAT_TWO 0x40000000 @@ -3068,8 +3072,8 @@ static int equalizer_ctl_put(struct snd_kcontrol *kcontrol, return 1; } -static const DECLARE_TLV_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); -static const DECLARE_TLV_DB_SCALE(eq_db_scale, -2400, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0); static int add_tuning_control(struct hda_codec *codec, hda_nid_t pnid, hda_nid_t nid, -- 2.14.1

2 1

[alsa-devel] linux-sound: timestamps for audio frames
by Muni Sekhar 02 May '18

02 May '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, >From sound driver, I am wondering if we could get the timestamps when It receives for example the first and last audio frames. Is there any way we can extract or expose that information to user mode? -- Thanks, Sekhar

1 0

[alsa-devel] [PATCH v3 0/3] ASoC: Add Multi CPU DAI support
by Vinod Koul 02 May '18

02 May '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Changes in v3: - Rebased on asoc for-next Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add Multi CPU DAI support for PCM ops ASoC: Add Multi CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 483 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 565 insertions(+), 278 deletions(-) -- 2.7.4

4 9